diff options
Diffstat (limited to 'src/modules/alsa')
74 files changed, 23001 insertions, 0 deletions
diff --git a/src/modules/alsa/90-pulseaudio.rules b/src/modules/alsa/90-pulseaudio.rules new file mode 100644 index 0000000..7bfacda --- /dev/null +++ b/src/modules/alsa/90-pulseaudio.rules @@ -0,0 +1,170 @@ +# do not edit this file, it will be overwritten on update + +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# Lesser General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +SUBSYSTEM!="sound", GOTO="pulseaudio_end" +ACTION!="change", GOTO="pulseaudio_end" +KERNEL!="card*", GOTO="pulseaudio_end" +SUBSYSTEMS=="usb", GOTO="pulseaudio_check_usb" +SUBSYSTEMS=="pci", GOTO="pulseaudio_check_pci" +SUBSYSTEMS=="firewire", GOTO="pulseaudio_firewire_quirk" + +SUBSYSTEMS=="platform", DRIVERS=="thinkpad_acpi", ENV{PULSE_IGNORE}="1" + +# Force enable speaker and internal mic for some laptops +# This should only be necessary for kernels 3.3, 3.4 and 3.5 (as they are lacking the phantom jack kctls). +# Acer AOA150 +ATTRS{subsystem_vendor}=="0x1025", ATTRS{subsystem_device}=="0x015b", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Acer Aspire 4810TZ +ATTRS{subsystem_vendor}=="0x1025", ATTRS{subsystem_device}=="0x022a", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Packard bell dot m/a +ATTRS{subsystem_vendor}=="0x1025", ATTRS{subsystem_device}=="0x028c", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Acer Aspire 1810TZ +ATTRS{subsystem_vendor}=="0x1025", ATTRS{subsystem_device}=="0x029b", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Acer AOD260 and AO532h +ATTRS{subsystem_vendor}=="0x1025", ATTRS{subsystem_device}=="0x0349", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell MXC051 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x01b5", ENV{PULSE_PROFILE_SET}="force-speaker.conf" +# Dell Inspiron 6400 and E1505 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x01bd", ENV{PULSE_PROFILE_SET}="force-speaker.conf" +# Dell Latitude D620 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x01c2", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell Latitude D820 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x01cc", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell Latitude D520 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x01d4", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell Latitude D420 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x01d6", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell Inspiron 1525 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x022f", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell Inspiron 1011 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x02f4", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell XPS 14 (L401X) +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x0468", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell XPS 15 (L501X) +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x046e", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell XPS 15 (L502X) +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x050e", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Dell Inspiron 3420 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x0553", ENV{PULSE_PROFILE_SET}="force-speaker.conf" +# Dell Inspiron 3520 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x0555", ENV{PULSE_PROFILE_SET}="force-speaker.conf" +# Dell Vostro 2420 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x0556", ENV{PULSE_PROFILE_SET}="force-speaker.conf" +# Dell Vostro 2520 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x0558", ENV{PULSE_PROFILE_SET}="force-speaker.conf" +# Dell Inspiron One 2020 +ATTRS{subsystem_vendor}=="0x1028", ATTRS{subsystem_device}=="0x0579", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Asus 904HA (1000H) +ATTRS{subsystem_vendor}=="0x1043", ATTRS{subsystem_device}=="0x831a", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Asus T101MT +ATTRS{subsystem_vendor}=="0x1043", ATTRS{subsystem_device}=="0x83ce", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Sony Vaio VGN-SR21M +ATTRS{subsystem_vendor}=="0x104d", ATTRS{subsystem_device}=="0x9033", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Sony Vaio VPC-W115XG +ATTRS{subsystem_vendor}=="0x104d", ATTRS{subsystem_device}=="0x9064", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Fujitsu Lifebook S7110 +ATTRS{subsystem_vendor}=="0x10cf", ATTRS{subsystem_device}=="0x1397", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Fujitsu Lifebook A530 +ATTRS{subsystem_vendor}=="0x10cf", ATTRS{subsystem_device}=="0x1531", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Toshiba A200 +ATTRS{subsystem_vendor}=="0x1179", ATTRS{subsystem_device}=="0xff00", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# MSI X360 +ATTRS{subsystem_vendor}=="0x1462", ATTRS{subsystem_device}=="0x1053", ENV{PULSE_PROFILE_SET}="force-speaker-and-int-mic.conf" +# Lenovo 3000 Y410 +ATTRS{subsystem_vendor}=="0x17aa", ATTRS{subsystem_device}=="0x384e", ENV{PULSE_PROFILE_SET}="force-speaker.conf" + +GOTO="pulseaudio_end" + +LABEL="pulseaudio_check_usb" +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="1978", ENV{PULSE_PROFILE_SET}="native-instruments-audio8dj.conf" +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="0839", ENV{PULSE_PROFILE_SET}="native-instruments-audio4dj.conf" +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="baff", ENV{PULSE_PROFILE_SET}="native-instruments-traktorkontrol-s4.conf" +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="4711", ENV{PULSE_PROFILE_SET}="native-instruments-korecontroller.conf" + +# This ID 17cc:041c is verified for the older Audio 2 DJ model (pre-2014 ish). +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="041c", ENV{PULSE_PROFILE_SET}="native-instruments-traktor-audio2.conf" +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="041d", ENV{PULSE_PROFILE_SET}="native-instruments-traktor-audio2.conf" + +# There appear to be two IDs in use for Traktor Audio 6 (or maybe 17cc:1011 +# is just incorrect - 17cc:1010 has been verified to be correct at least +# for some hardware). +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="1010", ENV{PULSE_PROFILE_SET}="native-instruments-traktor-audio6.conf" +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="1011", ENV{PULSE_PROFILE_SET}="native-instruments-traktor-audio6.conf" + +ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="1021", ENV{PULSE_PROFILE_SET}="native-instruments-traktor-audio10.conf" +ATTRS{idVendor}=="0763", ATTRS{idProduct}=="2012", ENV{PULSE_PROFILE_SET}="maudio-fasttrack-pro.conf" +ATTRS{idVendor}=="045e", ATTRS{idProduct}=="02bb", ENV{PULSE_PROFILE_SET}="kinect-audio.conf" +ATTRS{idVendor}=="041e", ATTRS{idProduct}=="322c", ENV{PULSE_PROFILE_SET}="sb-omni-surround-5.1.conf" +ATTRS{idVendor}=="0bda", ATTRS{idProduct}=="4014", ENV{PULSE_PROFILE_SET}="dell-dock-tb16-usb-audio.conf" + +# ID 1038:12ad is for the 2018 refresh of the Arctis 7. +# ID 1038:1294 is for Arctis Pro Wireless (which works with the Arctis 7 configuration). +ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1260", ENV{PULSE_PROFILE_SET}="usb-gaming-headset.conf" +ATTRS{idVendor}=="1038", ATTRS{idProduct}=="12ad", ENV{PULSE_PROFILE_SET}="usb-gaming-headset.conf" +ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1294", ENV{PULSE_PROFILE_SET}="usb-gaming-headset.conf" +ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1730", ENV{PULSE_PROFILE_SET}="usb-gaming-headset.conf" +# Lucidsound LS31 +ATTRS{idVendor}=="2f12", ATTRS{idProduct}=="0109", ENV{PULSE_PROFILE_SET}="usb-gaming-headset.conf" +# ID 9886:002c is for the Astro A50 Gen4 +ATTRS{idVendor}=="9886", ATTRS{idProduct}=="002c", ENV{PULSE_PROFILE_SET}="usb-gaming-headset.conf" +# ID 1532:0520 is for the Razer Kraken Tournament Edition +ATTRS{idVendor}=="1532", ATTRS{idProduct}=="0520", ENV{PULSE_PROFILE_SET}="usb-gaming-headset.conf" + + +# ID 1038:1250 is for the Arctis 5 +# ID 1037:12aa is for the Arctis 5 2019 +# ID 1038:1252 is for the Arctis Pro 2019 edition +ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1250", ENV{PULSE_PROFILE_SET}="steelseries-arctis-common-usb-audio.conf" +ATTRS{idVendor}=="1038", ATTRS{idProduct}=="12aa", ENV{PULSE_PROFILE_SET}="steelseries-arctis-common-usb-audio.conf" +ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1252", ENV{PULSE_PROFILE_SET}="steelseries-arctis-common-usb-audio.conf" + +ATTRS{idVendor}=="147a", ATTRS{idProduct}=="e055", ENV{PULSE_PROFILE_SET}="cmedia-high-speed-true-hdaudio.conf" + +# HyperX Cloud Orbit S has three modes. Each mode has a separate product ID. +# ID_SERIAL for this device is the device name + mode repeated three times. +# ID_SERIAL is used for the ID_ID property, and the ID_ID property is used in +# the card name in PulseAudio. The resulting card name is too long for the name +# length limit, so we set a more sensible ID_ID here (the same as the default +# ID_ID, but without repetition in the serial part). +ATTRS{idVendor}=="0951", ATTRS{idProduct}=="16ff", ENV{ID_ID}="usb-HyperX_Cloud_Orbit_S_2Ch-$env{ID_USB_INTERFACE_NUM}" +ATTRS{idVendor}=="0951", ATTRS{idProduct}=="1702", ENV{ID_ID}="usb-HyperX_Cloud_Orbit_S_Hi-Res_2Ch-$env{ID_USB_INTERFACE_NUM}" +ATTRS{idVendor}=="0951", ATTRS{idProduct}=="1703", ENV{ID_ID}="usb-HyperX_Cloud_Orbit_S_3D_8Ch-$env{ID_USB_INTERFACE_NUM}" + +GOTO="pulseaudio_end" + +LABEL="pulseaudio_check_pci" + +# Creative SoundBlaster Audigy-based cards +# EMU10k2/CA0100/CA0102/CA10200 +ATTRS{vendor}=="0x1102", ATTRS{device}=="0x0004", ENV{PULSE_PROFILE_SET}="audigy.conf" +# CA0108/CA10300 +ATTRS{vendor}=="0x1102", ATTRS{device}=="0x0008", ENV{PULSE_PROFILE_SET}="audigy.conf" + +GOTO="pulseaudio_end" + +LABEL="pulseaudio_firewire_quirk" + +# Focusrite Saffire Pro 10/26 i/o has a quirk to disappear from IEEE 1394 bus when losing connections. +# https://bugzilla.kernel.org/show_bug.cgi?id=199365 +ENV{ID_VENDOR_ID}=="0x00130e", ENV{ID_MODEL_ID}=="0x000003", ENV{PULSE_IGNORE}="1" +# Both of Saffire Pro 10 i/o and Liquid Saffire 56 have the same ID_MODEL_ID +# (0x000006), but Liquid Saffire 56 doesn't suffer from the problem, so we +# can't use ID_MODEL_ID to identify the problematic card. ID_MODEL works +# better here. +ENV{ID_VENDOR_ID}=="0x00130e", ENV{ID_MODEL}=="Pro10IO" ENV{PULSE_IGNORE}="1" + +LABEL="pulseaudio_end" diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c new file mode 100644 index 0000000..e494a12 --- /dev/null +++ b/src/modules/alsa/alsa-mixer.c @@ -0,0 +1,5386 @@ +/*** + This file is part of PulseAudio. + + Copyright 2004-2009 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <sys/types.h> +#include <alsa/asoundlib.h> +#include <math.h> + +#ifdef HAVE_VALGRIND_MEMCHECK_H +#include <valgrind/memcheck.h> +#endif + +#include <pulse/mainloop-api.h> +#include <pulse/sample.h> +#include <pulse/timeval.h> +#include <pulse/util.h> +#include <pulse/volume.h> +#include <pulse/xmalloc.h> +#include <pulse/utf8.h> + +#include <pulsecore/i18n.h> +#include <pulsecore/log.h> +#include <pulsecore/macro.h> +#include <pulsecore/core-util.h> +#include <pulsecore/conf-parser.h> +#include <pulsecore/strbuf.h> + +#include "alsa-mixer.h" +#include "alsa-util.h" + +#ifdef HAVE_VALGRIND_MEMCHECK_H +/* These macros are workarounds for a bug in valgrind, which is not handling the + * ALSA TLV syscalls correctly. See + * http://valgrind.10908.n7.nabble.com/Missing-ioctl-for-SNDRV-CTL-IOCTL-TLV-READ-td42711.html */ + +static inline int vgfix_get_capture_dB(snd_mixer_elem_t *a, snd_mixer_selem_channel_id_t b, long *c) { + int r = snd_mixer_selem_get_capture_dB(a, b, c); + VALGRIND_MAKE_MEM_DEFINED(c, sizeof(*c)); + return r; +} + +static inline int vgfix_get_playback_dB(snd_mixer_elem_t *a, snd_mixer_selem_channel_id_t b, long *c) { + int r = snd_mixer_selem_get_playback_dB(a, b, c); + VALGRIND_MAKE_MEM_DEFINED(c, sizeof(*c)); + return r; +} + +static inline int vgfix_ask_capture_vol_dB(snd_mixer_elem_t *a, long b, long *c) { + int r = snd_mixer_selem_ask_capture_vol_dB(a, b, c); + VALGRIND_MAKE_MEM_DEFINED(c, sizeof(*c)); + return r; +} + +static inline int vgfix_ask_playback_vol_dB(snd_mixer_elem_t *a, long b, long *c) { + int r = snd_mixer_selem_ask_playback_vol_dB(a, b, c); + VALGRIND_MAKE_MEM_DEFINED(c, sizeof(*c)); + return r; +} + +static inline int vgfix_get_capture_dB_range(snd_mixer_elem_t *a, long *b, long *c) { + int r = snd_mixer_selem_get_capture_dB_range(a, b, c); + VALGRIND_MAKE_MEM_DEFINED(b, sizeof(*b)); + VALGRIND_MAKE_MEM_DEFINED(c, sizeof(*c)); + return r; +} + +static inline int vgfix_get_playback_dB_range(snd_mixer_elem_t *a, long *b, long *c) { + int r = snd_mixer_selem_get_playback_dB_range(a, b, c); + VALGRIND_MAKE_MEM_DEFINED(b, sizeof(*b)); + VALGRIND_MAKE_MEM_DEFINED(c, sizeof(*c)); + return r; +} + +#define snd_mixer_selem_get_capture_dB(a, b, c) vgfix_get_capture_dB(a, b, c) +#define snd_mixer_selem_get_playback_dB(a, b, c) vgfix_get_playback_dB(a, b, c) +#define snd_mixer_selem_ask_capture_vol_dB(a, b, c) vgfix_ask_capture_vol_dB(a, b, c) +#define snd_mixer_selem_ask_playback_vol_dB(a, b, c) vgfix_ask_playback_vol_dB(a, b, c) +#define snd_mixer_selem_get_capture_dB_range(a, b, c) vgfix_get_capture_dB_range(a, b, c) +#define snd_mixer_selem_get_playback_dB_range(a, b, c) vgfix_get_playback_dB_range(a, b, c) + +#endif + +static int setting_select(pa_alsa_setting *s, snd_mixer_t *m); + +struct description_map { + const char *key; + const char *description; +}; + +struct description2_map { + const char *key; + const char *description; + pa_device_port_type_t type; +}; + +char *pa_alsa_mixer_id_to_string(char *dst, size_t dst_len, pa_alsa_mixer_id *id) { + if (id->index > 0) { + snprintf(dst, dst_len, "'%s',%d", id->name, id->index); + } else { + snprintf(dst, dst_len, "'%s'", id->name); + } + return dst; +} + +static int alsa_id_decode(const char *src, char *name, int *index) { + char *idx, c; + int i; + + *index = 0; + c = src[0]; + /* Strip quotes in entries such as 'Speaker',1 or "Speaker",1 */ + if (c == '\'' || c == '"') { + strcpy(name, src + 1); + for (i = 0; name[i] != '\0' && name[i] != c; i++); + idx = NULL; + if (name[i]) { + name[i] = '\0'; + idx = strchr(name + i + 1, ','); + } + } else { + strcpy(name, src); + idx = strchr(name, ','); + } + if (idx == NULL) + return 0; + *idx = '\0'; + idx++; + if (*idx < '0' || *idx > '9') { + pa_log("Element %s: index value is invalid", src); + return 1; + } + *index = atoi(idx); + return 0; +} + +pa_alsa_jack *pa_alsa_jack_new(pa_alsa_path *path, const char *mixer_device_name, const char *name, int index) { + pa_alsa_jack *jack; + + pa_assert(name); + + jack = pa_xnew0(pa_alsa_jack, 1); + jack->path = path; + jack->mixer_device_name = pa_xstrdup(mixer_device_name); + jack->name = pa_xstrdup(name); + jack->alsa_id.name = pa_sprintf_malloc("%s Jack", name); + jack->alsa_id.index = index; + jack->state_unplugged = PA_AVAILABLE_NO; + jack->state_plugged = PA_AVAILABLE_YES; + jack->ucm_devices = pa_dynarray_new(NULL); + jack->ucm_hw_mute_devices = pa_dynarray_new(NULL); + + return jack; +} + +void pa_alsa_jack_free(pa_alsa_jack *jack) { + pa_assert(jack); + + pa_dynarray_free(jack->ucm_hw_mute_devices); + pa_dynarray_free(jack->ucm_devices); + + pa_xfree(jack->alsa_id.name); + pa_xfree(jack->name); + pa_xfree(jack->mixer_device_name); + pa_xfree(jack); +} + +void pa_alsa_jack_set_has_control(pa_alsa_jack *jack, bool has_control) { + pa_alsa_ucm_device *device; + unsigned idx; + + pa_assert(jack); + + if (has_control == jack->has_control) + return; + + jack->has_control = has_control; + + PA_DYNARRAY_FOREACH(device, jack->ucm_hw_mute_devices, idx) + pa_alsa_ucm_device_update_available(device); + + PA_DYNARRAY_FOREACH(device, jack->ucm_devices, idx) + pa_alsa_ucm_device_update_available(device); +} + +void pa_alsa_jack_set_plugged_in(pa_alsa_jack *jack, bool plugged_in) { + pa_alsa_ucm_device *device; + unsigned idx; + + pa_assert(jack); + + if (plugged_in == jack->plugged_in) + return; + + jack->plugged_in = plugged_in; + + /* XXX: If this is a headphone jack that mutes speakers when plugged in, + * and the headphones get unplugged, then the headphone device must be set + * to unavailable and the speaker device must be set to unknown. So far so + * good. But there's an ugly detail: we must first set the availability of + * the speakers and then the headphones. We shouldn't need to care about + * the order, but we have to, because module-switch-on-port-available gets + * separate events for the two devices, and the intermediate state between + * the two events is such that the second event doesn't trigger the desired + * port switch, if the event order is "wrong". + * + * These are the transitions when the event order is "right": + * + * speakers: 1) unavailable -> 2) unknown -> 3) unknown + * headphones: 1) available -> 2) available -> 3) unavailable + * + * In the 2 -> 3 transition, headphones become unavailable, and + * module-switch-on-port-available sees that speakers can be used, so the + * port gets changed as it should. + * + * These are the transitions when the event order is "wrong": + * + * speakers: 1) unavailable -> 2) unavailable -> 3) unknown + * headphones: 1) available -> 2) unavailable -> 3) unavailable + * + * In the 1 -> 2 transition, headphones become unavailable, and there are + * no available ports to use, so no port change happens. In the 2 -> 3 + * transition, speaker availability becomes unknown, but that's not + * a strong enough signal for module-switch-on-port-available, so it still + * doesn't do the port switch. + * + * We should somehow merge the two events so that + * module-switch-on-port-available would handle both transitions in one go. + * If module-switch-on-port-available used a defer event to delay + * the port availability processing, that would probably do the trick. */ + + PA_DYNARRAY_FOREACH(device, jack->ucm_hw_mute_devices, idx) + pa_alsa_ucm_device_update_available(device); + + PA_DYNARRAY_FOREACH(device, jack->ucm_devices, idx) + pa_alsa_ucm_device_update_available(device); +} + +void pa_alsa_jack_add_ucm_device(pa_alsa_jack *jack, pa_alsa_ucm_device *device) { + pa_alsa_ucm_device *idevice; + unsigned idx, prio, iprio; + + pa_assert(jack); + pa_assert(device); + + /* store the ucm device with the sequence of priority from low to high. this + * could guarantee when the jack state is changed, the device with highest + * priority will send to the module-switch-on-port-available last */ + prio = device->playback_priority ? device->playback_priority : device->capture_priority; + + PA_DYNARRAY_FOREACH(idevice, jack->ucm_devices, idx) { + iprio = idevice->playback_priority ? idevice->playback_priority : idevice->capture_priority; + if (iprio > prio) + break; + } + pa_dynarray_insert_by_index(jack->ucm_devices, device, idx); +} + +void pa_alsa_jack_add_ucm_hw_mute_device(pa_alsa_jack *jack, pa_alsa_ucm_device *device) { + pa_assert(jack); + pa_assert(device); + + pa_dynarray_append(jack->ucm_hw_mute_devices, device); +} + +static const char *lookup_description(const char *key, const struct description_map dm[], unsigned n) { + unsigned i; + + if (!key) + return NULL; + + for (i = 0; i < n; i++) + if (pa_streq(dm[i].key, key)) + return _(dm[i].description); + + return NULL; +} + +static const struct description2_map *lookup_description2(const char *key, const struct description2_map dm[], unsigned n) { + unsigned i; + + if (!key) + return NULL; + + for (i = 0; i < n; i++) + if (pa_streq(dm[i].key, key)) + return &dm[i]; + + return NULL; +} + +struct pa_alsa_fdlist { + unsigned num_fds; + struct pollfd *fds; + /* This is a temporary buffer used to avoid lots of mallocs */ + struct pollfd *work_fds; + + snd_mixer_t *mixer; + snd_hctl_t *hctl; + + pa_mainloop_api *m; + pa_defer_event *defer; + pa_io_event **ios; + + bool polled; + + void (*cb)(void *userdata); + void *userdata; +}; + +static void io_cb(pa_mainloop_api *a, pa_io_event *e, int fd, pa_io_event_flags_t events, void *userdata) { + + struct pa_alsa_fdlist *fdl = userdata; + int err; + unsigned i; + unsigned short revents; + + pa_assert(a); + pa_assert(fdl); + pa_assert(fdl->mixer || fdl->hctl); + pa_assert(fdl->fds); + pa_assert(fdl->work_fds); + + if (fdl->polled) + return; + + fdl->polled = true; + + memcpy(fdl->work_fds, fdl->fds, sizeof(struct pollfd) * fdl->num_fds); + + for (i = 0; i < fdl->num_fds; i++) { + if (e == fdl->ios[i]) { + if (events & PA_IO_EVENT_INPUT) + fdl->work_fds[i].revents |= POLLIN; + if (events & PA_IO_EVENT_OUTPUT) + fdl->work_fds[i].revents |= POLLOUT; + if (events & PA_IO_EVENT_ERROR) + fdl->work_fds[i].revents |= POLLERR; + if (events & PA_IO_EVENT_HANGUP) + fdl->work_fds[i].revents |= POLLHUP; + break; + } + } + + pa_assert(i != fdl->num_fds); + + if (fdl->hctl) + err = snd_hctl_poll_descriptors_revents(fdl->hctl, fdl->work_fds, fdl->num_fds, &revents); + else + err = snd_mixer_poll_descriptors_revents(fdl->mixer, fdl->work_fds, fdl->num_fds, &revents); + + if (err < 0) { + pa_log_error("Unable to get poll revent: %s", pa_alsa_strerror(err)); + return; + } + + a->defer_enable(fdl->defer, 1); + + if (revents) { + if (fdl->hctl) + snd_hctl_handle_events(fdl->hctl); + else + snd_mixer_handle_events(fdl->mixer); + } +} + +static void defer_cb(pa_mainloop_api *a, pa_defer_event *e, void *userdata) { + struct pa_alsa_fdlist *fdl = userdata; + unsigned num_fds, i; + int err, n; + struct pollfd *temp; + + pa_assert(a); + pa_assert(fdl); + pa_assert(fdl->mixer || fdl->hctl); + + a->defer_enable(fdl->defer, 0); + + if (fdl->hctl) + n = snd_hctl_poll_descriptors_count(fdl->hctl); + else + n = snd_mixer_poll_descriptors_count(fdl->mixer); + + if (n < 0) { + pa_log("snd_mixer_poll_descriptors_count() failed: %s", pa_alsa_strerror(n)); + return; + } + else if (n == 0) { + pa_log_warn("Mixer has no poll descriptors. Please control mixer from PulseAudio only."); + return; + } + num_fds = (unsigned) n; + + if (num_fds != fdl->num_fds) { + if (fdl->fds) + pa_xfree(fdl->fds); + if (fdl->work_fds) + pa_xfree(fdl->work_fds); + fdl->fds = pa_xnew0(struct pollfd, num_fds); + fdl->work_fds = pa_xnew(struct pollfd, num_fds); + } + + memset(fdl->work_fds, 0, sizeof(struct pollfd) * num_fds); + + if (fdl->hctl) + err = snd_hctl_poll_descriptors(fdl->hctl, fdl->work_fds, num_fds); + else + err = snd_mixer_poll_descriptors(fdl->mixer, fdl->work_fds, num_fds); + + if (err < 0) { + pa_log_error("Unable to get poll descriptors: %s", pa_alsa_strerror(err)); + return; + } + + fdl->polled = false; + + if (memcmp(fdl->fds, fdl->work_fds, sizeof(struct pollfd) * num_fds) == 0) + return; + + if (fdl->ios) { + for (i = 0; i < fdl->num_fds; i++) + a->io_free(fdl->ios[i]); + + if (num_fds != fdl->num_fds) { + pa_xfree(fdl->ios); + fdl->ios = NULL; + } + } + + if (!fdl->ios) + fdl->ios = pa_xnew(pa_io_event*, num_fds); + + /* Swap pointers */ + temp = fdl->work_fds; + fdl->work_fds = fdl->fds; + fdl->fds = temp; + + fdl->num_fds = num_fds; + + for (i = 0;i < num_fds;i++) + fdl->ios[i] = a->io_new(a, fdl->fds[i].fd, + ((fdl->fds[i].events & POLLIN) ? PA_IO_EVENT_INPUT : 0) | + ((fdl->fds[i].events & POLLOUT) ? PA_IO_EVENT_OUTPUT : 0), + io_cb, fdl); +} + +struct pa_alsa_fdlist *pa_alsa_fdlist_new(void) { + struct pa_alsa_fdlist *fdl; + + fdl = pa_xnew0(struct pa_alsa_fdlist, 1); + + return fdl; +} + +void pa_alsa_fdlist_free(struct pa_alsa_fdlist *fdl) { + pa_assert(fdl); + + if (fdl->defer) { + pa_assert(fdl->m); + fdl->m->defer_free(fdl->defer); + } + + if (fdl->ios) { + unsigned i; + pa_assert(fdl->m); + for (i = 0; i < fdl->num_fds; i++) + fdl->m->io_free(fdl->ios[i]); + pa_xfree(fdl->ios); + } + + if (fdl->fds) + pa_xfree(fdl->fds); + if (fdl->work_fds) + pa_xfree(fdl->work_fds); + + pa_xfree(fdl); +} + +/* We can listen to either a snd_hctl_t or a snd_mixer_t, but not both */ +int pa_alsa_fdlist_set_handle(struct pa_alsa_fdlist *fdl, snd_mixer_t *mixer_handle, snd_hctl_t *hctl_handle, pa_mainloop_api *m) { + pa_assert(fdl); + pa_assert(hctl_handle || mixer_handle); + pa_assert(!(hctl_handle && mixer_handle)); + pa_assert(m); + pa_assert(!fdl->m); + + fdl->hctl = hctl_handle; + fdl->mixer = mixer_handle; + fdl->m = m; + fdl->defer = m->defer_new(m, defer_cb, fdl); + + return 0; +} + +struct pa_alsa_mixer_pdata { + pa_rtpoll *rtpoll; + pa_rtpoll_item *poll_item; + snd_mixer_t *mixer; +}; + +struct pa_alsa_mixer_pdata *pa_alsa_mixer_pdata_new(void) { + struct pa_alsa_mixer_pdata *pd; + + pd = pa_xnew0(struct pa_alsa_mixer_pdata, 1); + + return pd; +} + +void pa_alsa_mixer_pdata_free(struct pa_alsa_mixer_pdata *pd) { + pa_assert(pd); + + if (pd->poll_item) { + pa_rtpoll_item_free(pd->poll_item); + } + + pa_xfree(pd); +} + +static int rtpoll_work_cb(pa_rtpoll_item *i) { + struct pa_alsa_mixer_pdata *pd; + struct pollfd *p; + unsigned n_fds; + unsigned short revents = 0; + int err, ret = 0; + + pd = pa_rtpoll_item_get_work_userdata(i); + pa_assert_fp(pd); + pa_assert_fp(i == pd->poll_item); + + p = pa_rtpoll_item_get_pollfd(i, &n_fds); + + if ((err = snd_mixer_poll_descriptors_revents(pd->mixer, p, n_fds, &revents)) < 0) { + pa_log_error("Unable to get poll revent: %s", pa_alsa_strerror(err)); + ret = -1; + goto fail; + } + + if (revents) { + if (revents & (POLLNVAL | POLLERR)) { + pa_log_debug("Device disconnected, stopping poll on mixer"); + goto fail; + } else if (revents & POLLERR) { + /* This shouldn't happen. */ + pa_log_error("Got a POLLERR (revents = %04x), stopping poll on mixer", revents); + goto fail; + } + + err = snd_mixer_handle_events(pd->mixer); + + if (PA_LIKELY(err >= 0)) { + pa_rtpoll_item_free(i); + pa_alsa_set_mixer_rtpoll(pd, pd->mixer, pd->rtpoll); + } else { + pa_log_error("Error handling mixer event: %s", pa_alsa_strerror(err)); + ret = -1; + goto fail; + } + } + + return ret; + +fail: + pa_rtpoll_item_free(i); + + pd->poll_item = NULL; + pd->rtpoll = NULL; + pd->mixer = NULL; + + return ret; +} + +int pa_alsa_set_mixer_rtpoll(struct pa_alsa_mixer_pdata *pd, snd_mixer_t *mixer, pa_rtpoll *rtp) { + pa_rtpoll_item *i; + struct pollfd *p; + int err, n; + + pa_assert(pd); + pa_assert(mixer); + pa_assert(rtp); + + if ((n = snd_mixer_poll_descriptors_count(mixer)) < 0) { + pa_log("snd_mixer_poll_descriptors_count() failed: %s", pa_alsa_strerror(n)); + return -1; + } + else if (n == 0) { + pa_log_warn("Mixer has no poll descriptors. Please control mixer from PulseAudio only."); + return 0; + } + + i = pa_rtpoll_item_new(rtp, PA_RTPOLL_LATE, (unsigned) n); + + p = pa_rtpoll_item_get_pollfd(i, NULL); + + memset(p, 0, sizeof(struct pollfd) * n); + + if ((err = snd_mixer_poll_descriptors(mixer, p, (unsigned) n)) < 0) { + pa_log_error("Unable to get poll descriptors: %s", pa_alsa_strerror(err)); + pa_rtpoll_item_free(i); + return -1; + } + + pd->rtpoll = rtp; + pd->poll_item = i; + pd->mixer = mixer; + + pa_rtpoll_item_set_work_callback(i, rtpoll_work_cb, pd); + + return 0; +} + +static const snd_mixer_selem_channel_id_t alsa_channel_ids[PA_CHANNEL_POSITION_MAX] = { + [PA_CHANNEL_POSITION_MONO] = SND_MIXER_SCHN_MONO, /* The ALSA name is just an alias! */ + + [PA_CHANNEL_POSITION_FRONT_CENTER] = SND_MIXER_SCHN_FRONT_CENTER, + [PA_CHANNEL_POSITION_FRONT_LEFT] = SND_MIXER_SCHN_FRONT_LEFT, + [PA_CHANNEL_POSITION_FRONT_RIGHT] = SND_MIXER_SCHN_FRONT_RIGHT, + + [PA_CHANNEL_POSITION_REAR_CENTER] = SND_MIXER_SCHN_REAR_CENTER, + [PA_CHANNEL_POSITION_REAR_LEFT] = SND_MIXER_SCHN_REAR_LEFT, + [PA_CHANNEL_POSITION_REAR_RIGHT] = SND_MIXER_SCHN_REAR_RIGHT, + + [PA_CHANNEL_POSITION_LFE] = SND_MIXER_SCHN_WOOFER, + + [PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER] = SND_MIXER_SCHN_UNKNOWN, + + [PA_CHANNEL_POSITION_SIDE_LEFT] = SND_MIXER_SCHN_SIDE_LEFT, + [PA_CHANNEL_POSITION_SIDE_RIGHT] = SND_MIXER_SCHN_SIDE_RIGHT, + + [PA_CHANNEL_POSITION_AUX0] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX1] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX2] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX3] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX4] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX5] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX6] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX7] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX8] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX9] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX10] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX11] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX12] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX13] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX14] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX15] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX16] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX17] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX18] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX19] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX20] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX21] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX22] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX23] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX24] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX25] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX26] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX27] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX28] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX29] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX30] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_AUX31] = SND_MIXER_SCHN_UNKNOWN, + + [PA_CHANNEL_POSITION_TOP_CENTER] = SND_MIXER_SCHN_UNKNOWN, + + [PA_CHANNEL_POSITION_TOP_FRONT_CENTER] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_TOP_FRONT_LEFT] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_TOP_FRONT_RIGHT] = SND_MIXER_SCHN_UNKNOWN, + + [PA_CHANNEL_POSITION_TOP_REAR_CENTER] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_TOP_REAR_LEFT] = SND_MIXER_SCHN_UNKNOWN, + [PA_CHANNEL_POSITION_TOP_REAR_RIGHT] = SND_MIXER_SCHN_UNKNOWN +}; + +static snd_mixer_selem_channel_id_t alsa_channel_positions[POSITION_MASK_CHANNELS] = { + SND_MIXER_SCHN_FRONT_LEFT, + SND_MIXER_SCHN_FRONT_RIGHT, + SND_MIXER_SCHN_REAR_LEFT, + SND_MIXER_SCHN_REAR_RIGHT, + SND_MIXER_SCHN_FRONT_CENTER, + SND_MIXER_SCHN_WOOFER, + SND_MIXER_SCHN_SIDE_LEFT, + SND_MIXER_SCHN_SIDE_RIGHT, +#if POSITION_MASK_CHANNELS > 8 +#error "Extend alsa_channel_positions[] array (9+)" +#endif +}; + +static void setting_free(pa_alsa_setting *s) { + pa_assert(s); + + if (s->options) + pa_idxset_free(s->options, NULL); + + pa_xfree(s->name); + pa_xfree(s->description); + pa_xfree(s); +} + +static void option_free(pa_alsa_option *o) { + pa_assert(o); + + pa_xfree(o->alsa_name); + pa_xfree(o->name); + pa_xfree(o->description); + pa_xfree(o); +} + +static void decibel_fix_free(pa_alsa_decibel_fix *db_fix) { + pa_assert(db_fix); + + pa_xfree(db_fix->name); + pa_xfree(db_fix->db_values); + + pa_xfree(db_fix->key); + pa_xfree(db_fix); +} + +static void element_free(pa_alsa_element *e) { + pa_alsa_option *o; + pa_assert(e); + + while ((o = e->options)) { + PA_LLIST_REMOVE(pa_alsa_option, e->options, o); + option_free(o); + } + + if (e->db_fix) + decibel_fix_free(e->db_fix); + + pa_xfree(e->alsa_id.name); + pa_xfree(e); +} + +void pa_alsa_path_free(pa_alsa_path *p) { + pa_alsa_jack *j; + pa_alsa_element *e; + pa_alsa_setting *s; + + pa_assert(p); + + while ((j = p->jacks)) { + PA_LLIST_REMOVE(pa_alsa_jack, p->jacks, j); + pa_alsa_jack_free(j); + } + + while ((e = p->elements)) { + PA_LLIST_REMOVE(pa_alsa_element, p->elements, e); + element_free(e); + } + + while ((s = p->settings)) { + PA_LLIST_REMOVE(pa_alsa_setting, p->settings, s); + setting_free(s); + } + + pa_proplist_free(p->proplist); + pa_xfree(p->availability_group); + pa_xfree(p->name); + pa_xfree(p->description); + pa_xfree(p->description_key); + pa_xfree(p); +} + +void pa_alsa_path_set_free(pa_alsa_path_set *ps) { + pa_assert(ps); + + if (ps->paths) + pa_hashmap_free(ps->paths); + + pa_xfree(ps); +} + +int pa_alsa_path_set_is_empty(pa_alsa_path_set *ps) { + if (ps && !pa_hashmap_isempty(ps->paths)) + return 0; + return 1; +} + +static long to_alsa_dB(pa_volume_t v) { + return lround(pa_sw_volume_to_dB(v) * 100.0); +} + +static pa_volume_t from_alsa_dB(long v) { + return pa_sw_volume_from_dB((double) v / 100.0); +} + +static long to_alsa_volume(pa_volume_t v, long min, long max) { + long w; + + w = (long) round(((double) v * (double) (max - min)) / PA_VOLUME_NORM) + min; + return PA_CLAMP_UNLIKELY(w, min, max); +} + +static pa_volume_t from_alsa_volume(long v, long min, long max) { + return (pa_volume_t) round(((double) (v - min) * PA_VOLUME_NORM) / (double) (max - min)); +} + +#define SELEM_INIT(sid, aid) \ + do { \ + snd_mixer_selem_id_alloca(&(sid)); \ + snd_mixer_selem_id_set_name((sid), (aid)->name); \ + snd_mixer_selem_id_set_index((sid), (aid)->index); \ + } while(false) + +static int element_get_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_channel_map *cm, pa_cvolume *v) { + snd_mixer_selem_id_t *sid; + snd_mixer_elem_t *me; + snd_mixer_selem_channel_id_t c; + pa_channel_position_mask_t mask = 0; + char buf[64]; + unsigned k; + + pa_assert(m); + pa_assert(e); + pa_assert(cm); + pa_assert(v); + + SELEM_INIT(sid, &e->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return -1; + } + + pa_cvolume_mute(v, cm->channels); + + /* We take the highest volume of all channels that match */ + + for (c = 0; c <= SND_MIXER_SCHN_LAST; c++) { + int r; + pa_volume_t f; + + if (e->has_dB) { + long value = 0; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + if (snd_mixer_selem_has_playback_channel(me, c)) { + if (e->db_fix) { + if ((r = snd_mixer_selem_get_playback_volume(me, c, &value)) >= 0) { + /* If the channel volume is outside the limits set + * by the dB fix, we clamp the hw volume to be + * within the limits. */ + if (value < e->db_fix->min_step) { + value = e->db_fix->min_step; + snd_mixer_selem_set_playback_volume(me, c, value); + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_debug("Playback volume for element %s channel %i was below the dB fix limit. " + "Volume reset to %0.2f dB.", buf, c, + e->db_fix->db_values[value - e->db_fix->min_step] / 100.0); + } else if (value > e->db_fix->max_step) { + value = e->db_fix->max_step; + snd_mixer_selem_set_playback_volume(me, c, value); + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_debug("Playback volume for element %s channel %i was over the dB fix limit. " + "Volume reset to %0.2f dB.", buf, c, + e->db_fix->db_values[value - e->db_fix->min_step] / 100.0); + } + + /* Volume step -> dB value conversion. */ + value = e->db_fix->db_values[value - e->db_fix->min_step]; + } + } else + r = snd_mixer_selem_get_playback_dB(me, c, &value); + } else + r = -1; + } else { + if (snd_mixer_selem_has_capture_channel(me, c)) { + if (e->db_fix) { + if ((r = snd_mixer_selem_get_capture_volume(me, c, &value)) >= 0) { + /* If the channel volume is outside the limits set + * by the dB fix, we clamp the hw volume to be + * within the limits. */ + if (value < e->db_fix->min_step) { + value = e->db_fix->min_step; + snd_mixer_selem_set_capture_volume(me, c, value); + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_debug("Capture volume for element %s channel %i was below the dB fix limit. " + "Volume reset to %0.2f dB.", buf, c, + e->db_fix->db_values[value - e->db_fix->min_step] / 100.0); + } else if (value > e->db_fix->max_step) { + value = e->db_fix->max_step; + snd_mixer_selem_set_capture_volume(me, c, value); + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_debug("Capture volume for element %s channel %i was over the dB fix limit. " + "Volume reset to %0.2f dB.", buf, c, + e->db_fix->db_values[value - e->db_fix->min_step] / 100.0); + } + + /* Volume step -> dB value conversion. */ + value = e->db_fix->db_values[value - e->db_fix->min_step]; + } + } else + r = snd_mixer_selem_get_capture_dB(me, c, &value); + } else + r = -1; + } + + if (r < 0) + continue; + +#ifdef HAVE_VALGRIND_MEMCHECK_H + VALGRIND_MAKE_MEM_DEFINED(&value, sizeof(value)); +#endif + + f = from_alsa_dB(value); + + } else { + long value = 0; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + if (snd_mixer_selem_has_playback_channel(me, c)) + r = snd_mixer_selem_get_playback_volume(me, c, &value); + else + r = -1; + } else { + if (snd_mixer_selem_has_capture_channel(me, c)) + r = snd_mixer_selem_get_capture_volume(me, c, &value); + else + r = -1; + } + + if (r < 0) + continue; + + f = from_alsa_volume(value, e->min_volume, e->max_volume); + } + + for (k = 0; k < cm->channels; k++) + if (e->masks[c][e->n_channels-1] & PA_CHANNEL_POSITION_MASK(cm->map[k])) + if (v->values[k] < f) + v->values[k] = f; + + mask |= e->masks[c][e->n_channels-1]; + } + + for (k = 0; k < cm->channels; k++) + if (!(mask & PA_CHANNEL_POSITION_MASK(cm->map[k]))) + v->values[k] = PA_VOLUME_NORM; + + return 0; +} + +int pa_alsa_path_get_volume(pa_alsa_path *p, snd_mixer_t *m, const pa_channel_map *cm, pa_cvolume *v) { + pa_alsa_element *e; + + pa_assert(m); + pa_assert(p); + pa_assert(cm); + pa_assert(v); + + if (!p->has_volume) + return -1; + + pa_cvolume_reset(v, cm->channels); + + PA_LLIST_FOREACH(e, p->elements) { + pa_cvolume ev; + + if (e->volume_use != PA_ALSA_VOLUME_MERGE) + continue; + + pa_assert(!p->has_dB || e->has_dB); + + if (element_get_volume(e, m, cm, &ev) < 0) + return -1; + + /* If we have no dB information all we can do is take the first element and leave */ + if (!p->has_dB) { + *v = ev; + return 0; + } + + pa_sw_cvolume_multiply(v, v, &ev); + } + + return 0; +} + +static int element_get_switch(pa_alsa_element *e, snd_mixer_t *m, bool *b) { + snd_mixer_selem_id_t *sid; + snd_mixer_elem_t *me; + snd_mixer_selem_channel_id_t c; + char buf[64]; + + pa_assert(m); + pa_assert(e); + pa_assert(b); + + SELEM_INIT(sid, &e->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return -1; + } + + /* We return muted if at least one channel is muted */ + + for (c = 0; c <= SND_MIXER_SCHN_LAST; c++) { + int r; + int value = 0; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + if (snd_mixer_selem_has_playback_channel(me, c)) + r = snd_mixer_selem_get_playback_switch(me, c, &value); + else + r = -1; + } else { + if (snd_mixer_selem_has_capture_channel(me, c)) + r = snd_mixer_selem_get_capture_switch(me, c, &value); + else + r = -1; + } + + if (r < 0) + continue; + + if (!value) { + *b = false; + return 0; + } + } + + *b = true; + return 0; +} + +int pa_alsa_path_get_mute(pa_alsa_path *p, snd_mixer_t *m, bool *muted) { + pa_alsa_element *e; + + pa_assert(m); + pa_assert(p); + pa_assert(muted); + + if (!p->has_mute) + return -1; + + PA_LLIST_FOREACH(e, p->elements) { + bool b; + + if (e->switch_use != PA_ALSA_SWITCH_MUTE) + continue; + + if (element_get_switch(e, m, &b) < 0) + return -1; + + if (!b) { + *muted = true; + return 0; + } + } + + *muted = false; + return 0; +} + +/* Finds the closest item in db_fix->db_values and returns the corresponding + * step. *db_value is replaced with the value from the db_values table. + * Rounding is done based on the rounding parameter: -1 means rounding down and + * +1 means rounding up. */ +static long decibel_fix_get_step(pa_alsa_decibel_fix *db_fix, long *db_value, int rounding) { + unsigned i = 0; + unsigned max_i = 0; + + pa_assert(db_fix); + pa_assert(db_value); + pa_assert(rounding != 0); + + max_i = db_fix->max_step - db_fix->min_step; + + if (rounding > 0) { + for (i = 0; i < max_i; i++) { + if (db_fix->db_values[i] >= *db_value) + break; + } + } else { + for (i = 0; i < max_i; i++) { + if (db_fix->db_values[i + 1] > *db_value) + break; + } + } + + *db_value = db_fix->db_values[i]; + + return i + db_fix->min_step; +} + +/* Alsa lib documentation says for snd_mixer_selem_set_playback_dB() direction argument, + * that "-1 = accurate or first below, 0 = accurate, 1 = accurate or first above". + * But even with accurate nearest dB volume step is not selected, so that is why we need + * this function. Returns 0 and nearest selectable volume in *value_dB on success or + * negative error code if fails. */ +static int element_get_nearest_alsa_dB(snd_mixer_elem_t *me, snd_mixer_selem_channel_id_t c, pa_alsa_direction_t d, long *value_dB) { + + long alsa_val; + long value_high; + long value_low; + int r = -1; + + pa_assert(me); + pa_assert(value_dB); + + if (d == PA_ALSA_DIRECTION_OUTPUT) { + if ((r = snd_mixer_selem_ask_playback_dB_vol(me, *value_dB, +1, &alsa_val)) >= 0) + r = snd_mixer_selem_ask_playback_vol_dB(me, alsa_val, &value_high); + + if (r < 0) + return r; + + if (value_high == *value_dB) + return r; + + if ((r = snd_mixer_selem_ask_playback_dB_vol(me, *value_dB, -1, &alsa_val)) >= 0) + r = snd_mixer_selem_ask_playback_vol_dB(me, alsa_val, &value_low); + } else { + if ((r = snd_mixer_selem_ask_capture_dB_vol(me, *value_dB, +1, &alsa_val)) >= 0) + r = snd_mixer_selem_ask_capture_vol_dB(me, alsa_val, &value_high); + + if (r < 0) + return r; + + if (value_high == *value_dB) + return r; + + if ((r = snd_mixer_selem_ask_capture_dB_vol(me, *value_dB, -1, &alsa_val)) >= 0) + r = snd_mixer_selem_ask_capture_vol_dB(me, alsa_val, &value_low); + } + + if (r < 0) + return r; + + if (labs(value_high - *value_dB) < labs(value_low - *value_dB)) + *value_dB = value_high; + else + *value_dB = value_low; + + return r; +} + +static int element_set_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_channel_map *cm, pa_cvolume *v, bool deferred_volume, bool write_to_hw) { + + snd_mixer_selem_id_t *sid; + pa_cvolume rv; + snd_mixer_elem_t *me; + snd_mixer_selem_channel_id_t c; + pa_channel_position_mask_t mask = 0; + char buf[64]; + unsigned k; + + pa_assert(m); + pa_assert(e); + pa_assert(cm); + pa_assert(v); + pa_assert(pa_cvolume_compatible_with_channel_map(v, cm)); + + SELEM_INIT(sid, &e->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return -1; + } + + pa_cvolume_mute(&rv, cm->channels); + + for (c = 0; c <= SND_MIXER_SCHN_LAST; c++) { + int r; + pa_volume_t f = PA_VOLUME_MUTED; + bool found = false; + + for (k = 0; k < cm->channels; k++) + if (e->masks[c][e->n_channels-1] & PA_CHANNEL_POSITION_MASK(cm->map[k])) { + found = true; + if (v->values[k] > f) + f = v->values[k]; + } + + if (!found) { + /* Hmm, so this channel does not exist in the volume + * struct, so let's bind it to the overall max of the + * volume. */ + f = pa_cvolume_max(v); + } + + if (e->has_dB) { + long value = to_alsa_dB(f); + int rounding; + + if (e->volume_limit >= 0 && value > (e->max_dB * 100)) + value = e->max_dB * 100; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + /* If we call set_playback_volume() without checking first + * if the channel is available, ALSA behaves very + * strangely and doesn't fail the call */ + if (snd_mixer_selem_has_playback_channel(me, c)) { + rounding = +1; + if (e->db_fix) { + if (write_to_hw) + r = snd_mixer_selem_set_playback_volume(me, c, decibel_fix_get_step(e->db_fix, &value, rounding)); + else { + decibel_fix_get_step(e->db_fix, &value, rounding); + r = 0; + } + + } else { + if (write_to_hw) { + if (deferred_volume) { + if ((r = element_get_nearest_alsa_dB(me, c, PA_ALSA_DIRECTION_OUTPUT, &value)) >= 0) + r = snd_mixer_selem_set_playback_dB(me, c, value, 0); + } else { + if ((r = snd_mixer_selem_set_playback_dB(me, c, value, rounding)) >= 0) + r = snd_mixer_selem_get_playback_dB(me, c, &value); + } + } else { + long alsa_val; + if ((r = snd_mixer_selem_ask_playback_dB_vol(me, value, rounding, &alsa_val)) >= 0) + r = snd_mixer_selem_ask_playback_vol_dB(me, alsa_val, &value); + } + } + } else + r = -1; + } else { + if (snd_mixer_selem_has_capture_channel(me, c)) { + rounding = -1; + if (e->db_fix) { + if (write_to_hw) + r = snd_mixer_selem_set_capture_volume(me, c, decibel_fix_get_step(e->db_fix, &value, rounding)); + else { + decibel_fix_get_step(e->db_fix, &value, rounding); + r = 0; + } + + } else { + if (write_to_hw) { + if (deferred_volume) { + if ((r = element_get_nearest_alsa_dB(me, c, PA_ALSA_DIRECTION_INPUT, &value)) >= 0) + r = snd_mixer_selem_set_capture_dB(me, c, value, 0); + } else { + if ((r = snd_mixer_selem_set_capture_dB(me, c, value, rounding)) >= 0) + r = snd_mixer_selem_get_capture_dB(me, c, &value); + } + } else { + long alsa_val; + if ((r = snd_mixer_selem_ask_capture_dB_vol(me, value, rounding, &alsa_val)) >= 0) + r = snd_mixer_selem_ask_capture_vol_dB(me, alsa_val, &value); + } + } + } else + r = -1; + } + + if (r < 0) + continue; + + f = from_alsa_dB(value); + + } else { + long value; + + value = to_alsa_volume(f, e->min_volume, e->max_volume); + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + if (snd_mixer_selem_has_playback_channel(me, c)) { + if ((r = snd_mixer_selem_set_playback_volume(me, c, value)) >= 0) + r = snd_mixer_selem_get_playback_volume(me, c, &value); + } else + r = -1; + } else { + if (snd_mixer_selem_has_capture_channel(me, c)) { + if ((r = snd_mixer_selem_set_capture_volume(me, c, value)) >= 0) + r = snd_mixer_selem_get_capture_volume(me, c, &value); + } else + r = -1; + } + + if (r < 0) + continue; + + f = from_alsa_volume(value, e->min_volume, e->max_volume); + } + + for (k = 0; k < cm->channels; k++) + if (e->masks[c][e->n_channels-1] & PA_CHANNEL_POSITION_MASK(cm->map[k])) + if (rv.values[k] < f) + rv.values[k] = f; + + mask |= e->masks[c][e->n_channels-1]; + } + + for (k = 0; k < cm->channels; k++) + if (!(mask & PA_CHANNEL_POSITION_MASK(cm->map[k]))) + rv.values[k] = PA_VOLUME_NORM; + + *v = rv; + return 0; +} + +int pa_alsa_path_set_volume(pa_alsa_path *p, snd_mixer_t *m, const pa_channel_map *cm, pa_cvolume *v, bool deferred_volume, bool write_to_hw) { + + pa_alsa_element *e; + pa_cvolume rv; + + pa_assert(m); + pa_assert(p); + pa_assert(cm); + pa_assert(v); + pa_assert(pa_cvolume_compatible_with_channel_map(v, cm)); + + if (!p->has_volume) + return -1; + + rv = *v; /* Remaining adjustment */ + pa_cvolume_reset(v, cm->channels); /* Adjustment done */ + + PA_LLIST_FOREACH(e, p->elements) { + pa_cvolume ev; + + if (e->volume_use != PA_ALSA_VOLUME_MERGE) + continue; + + pa_assert(!p->has_dB || e->has_dB); + + ev = rv; + if (element_set_volume(e, m, cm, &ev, deferred_volume, write_to_hw) < 0) + return -1; + + if (!p->has_dB) { + *v = ev; + return 0; + } + + pa_sw_cvolume_multiply(v, v, &ev); + pa_sw_cvolume_divide(&rv, &rv, &ev); + } + + return 0; +} + +static int element_set_switch(pa_alsa_element *e, snd_mixer_t *m, bool b) { + snd_mixer_elem_t *me; + snd_mixer_selem_id_t *sid; + char buf[64]; + int r; + + pa_assert(m); + pa_assert(e); + + SELEM_INIT(sid, &e->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return -1; + } + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + r = snd_mixer_selem_set_playback_switch_all(me, b); + else + r = snd_mixer_selem_set_capture_switch_all(me, b); + + if (r < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to set switch of %s: %s", buf, pa_alsa_strerror(errno)); + } + + return r; +} + +int pa_alsa_path_set_mute(pa_alsa_path *p, snd_mixer_t *m, bool muted) { + pa_alsa_element *e; + + pa_assert(m); + pa_assert(p); + + if (!p->has_mute) + return -1; + + PA_LLIST_FOREACH(e, p->elements) { + + if (e->switch_use != PA_ALSA_SWITCH_MUTE) + continue; + + if (element_set_switch(e, m, !muted) < 0) + return -1; + } + + return 0; +} + +/* Depending on whether e->volume_use is _OFF, _ZERO or _CONSTANT, this + * function sets all channels of the volume element to e->min_volume, 0 dB or + * e->constant_volume. */ +static int element_set_constant_volume(pa_alsa_element *e, snd_mixer_t *m) { + snd_mixer_elem_t *me = NULL; + snd_mixer_selem_id_t *sid = NULL; + int r = 0; + long volume = -1; + bool volume_set = false; + char buf[64]; + + pa_assert(m); + pa_assert(e); + + SELEM_INIT(sid, &e->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return -1; + } + + switch (e->volume_use) { + case PA_ALSA_VOLUME_OFF: + volume = e->min_volume; + volume_set = true; + break; + + case PA_ALSA_VOLUME_ZERO: + if (e->db_fix) { + long dB = 0; + + volume = decibel_fix_get_step(e->db_fix, &dB, (e->direction == PA_ALSA_DIRECTION_OUTPUT ? +1 : -1)); + volume_set = true; + } + break; + + case PA_ALSA_VOLUME_CONSTANT: + volume = e->constant_volume; + volume_set = true; + break; + + default: + pa_assert_not_reached(); + } + + if (volume_set) { + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + r = snd_mixer_selem_set_playback_volume_all(me, volume); + else + r = snd_mixer_selem_set_capture_volume_all(me, volume); + } else { + pa_assert(e->volume_use == PA_ALSA_VOLUME_ZERO); + pa_assert(!e->db_fix); + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + r = snd_mixer_selem_set_playback_dB_all(me, 0, +1); + else + r = snd_mixer_selem_set_capture_dB_all(me, 0, -1); + } + + if (r < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to set volume of %s: %s", buf, pa_alsa_strerror(errno)); + } + + return r; +} + +int pa_alsa_path_select(pa_alsa_path *p, pa_alsa_setting *s, snd_mixer_t *m, bool device_is_muted) { + pa_alsa_element *e; + int r = 0; + + pa_assert(m); + pa_assert(p); + + pa_log_debug("Activating path %s", p->name); + pa_alsa_path_dump(p); + + /* First turn on hw mute if available, to avoid noise + * when setting the mixer controls. */ + if (p->mute_during_activation) { + PA_LLIST_FOREACH(e, p->elements) { + if (e->switch_use == PA_ALSA_SWITCH_MUTE) + /* If the muting fails here, that's not a critical problem for + * selecting a path, so we ignore the return value. + * element_set_switch() will print a warning anyway, so this + * won't be a silent failure either. */ + (void) element_set_switch(e, m, false); + } + } + + PA_LLIST_FOREACH(e, p->elements) { + + switch (e->switch_use) { + case PA_ALSA_SWITCH_OFF: + r = element_set_switch(e, m, false); + break; + + case PA_ALSA_SWITCH_ON: + r = element_set_switch(e, m, true); + break; + + case PA_ALSA_SWITCH_MUTE: + case PA_ALSA_SWITCH_IGNORE: + case PA_ALSA_SWITCH_SELECT: + r = 0; + break; + } + + if (r < 0) + return -1; + + switch (e->volume_use) { + case PA_ALSA_VOLUME_OFF: + case PA_ALSA_VOLUME_ZERO: + case PA_ALSA_VOLUME_CONSTANT: + r = element_set_constant_volume(e, m); + break; + + case PA_ALSA_VOLUME_MERGE: + case PA_ALSA_VOLUME_IGNORE: + r = 0; + break; + } + + if (r < 0) + return -1; + } + + if (s) + setting_select(s, m); + + /* Finally restore hw mute to the device mute status. */ + if (p->mute_during_activation) { + PA_LLIST_FOREACH(e, p->elements) { + if (e->switch_use == PA_ALSA_SWITCH_MUTE) { + if (element_set_switch(e, m, !device_is_muted) < 0) + return -1; + } + } + } + + return 0; +} + +static int check_required(pa_alsa_element *e, snd_mixer_elem_t *me) { + bool has_switch; + bool has_enumeration; + bool has_volume; + + pa_assert(e); + pa_assert(me); + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + has_switch = + snd_mixer_selem_has_playback_switch(me) || + (e->direction_try_other && snd_mixer_selem_has_capture_switch(me)); + } else { + has_switch = + snd_mixer_selem_has_capture_switch(me) || + (e->direction_try_other && snd_mixer_selem_has_playback_switch(me)); + } + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + has_volume = + snd_mixer_selem_has_playback_volume(me) || + (e->direction_try_other && snd_mixer_selem_has_capture_volume(me)); + } else { + has_volume = + snd_mixer_selem_has_capture_volume(me) || + (e->direction_try_other && snd_mixer_selem_has_playback_volume(me)); + } + + has_enumeration = snd_mixer_selem_is_enumerated(me); + + if ((e->required == PA_ALSA_REQUIRED_SWITCH && !has_switch) || + (e->required == PA_ALSA_REQUIRED_VOLUME && !has_volume) || + (e->required == PA_ALSA_REQUIRED_ENUMERATION && !has_enumeration)) + return -1; + + if (e->required == PA_ALSA_REQUIRED_ANY && !(has_switch || has_volume || has_enumeration)) + return -1; + + if ((e->required_absent == PA_ALSA_REQUIRED_SWITCH && has_switch) || + (e->required_absent == PA_ALSA_REQUIRED_VOLUME && has_volume) || + (e->required_absent == PA_ALSA_REQUIRED_ENUMERATION && has_enumeration)) + return -1; + + if (e->required_absent == PA_ALSA_REQUIRED_ANY && (has_switch || has_volume || has_enumeration)) + return -1; + + if (e->required_any != PA_ALSA_REQUIRED_IGNORE) { + switch (e->required_any) { + case PA_ALSA_REQUIRED_VOLUME: + e->path->req_any_present |= (e->volume_use != PA_ALSA_VOLUME_IGNORE); + break; + case PA_ALSA_REQUIRED_SWITCH: + e->path->req_any_present |= (e->switch_use != PA_ALSA_SWITCH_IGNORE); + break; + case PA_ALSA_REQUIRED_ENUMERATION: + e->path->req_any_present |= (e->enumeration_use != PA_ALSA_ENUMERATION_IGNORE); + break; + case PA_ALSA_REQUIRED_ANY: + e->path->req_any_present |= + (e->volume_use != PA_ALSA_VOLUME_IGNORE) || + (e->switch_use != PA_ALSA_SWITCH_IGNORE) || + (e->enumeration_use != PA_ALSA_ENUMERATION_IGNORE); + break; + default: + pa_assert_not_reached(); + } + } + + if (e->enumeration_use == PA_ALSA_ENUMERATION_SELECT) { + pa_alsa_option *o; + PA_LLIST_FOREACH(o, e->options) { + e->path->req_any_present |= (o->required_any != PA_ALSA_REQUIRED_IGNORE) && + (o->alsa_idx >= 0); + if (o->required != PA_ALSA_REQUIRED_IGNORE && o->alsa_idx < 0) + return -1; + if (o->required_absent != PA_ALSA_REQUIRED_IGNORE && o->alsa_idx >= 0) + return -1; + } + } + + return 0; +} + +static int element_ask_vol_dB(snd_mixer_elem_t *me, pa_alsa_direction_t dir, long value, long *dBvalue) { + if (dir == PA_ALSA_DIRECTION_OUTPUT) + return snd_mixer_selem_ask_playback_vol_dB(me, value, dBvalue); + else + return snd_mixer_selem_ask_capture_vol_dB(me, value, dBvalue); +} + +static bool element_probe_volume(pa_alsa_element *e, snd_mixer_elem_t *me) { + + long min_dB = 0, max_dB = 0; + int r; + bool is_mono; + pa_channel_position_t p; + char buf[64]; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + if (!snd_mixer_selem_has_playback_volume(me)) { + if (e->direction_try_other && snd_mixer_selem_has_capture_volume(me)) + e->direction = PA_ALSA_DIRECTION_INPUT; + else + return false; + } + } else { + if (!snd_mixer_selem_has_capture_volume(me)) { + if (e->direction_try_other && snd_mixer_selem_has_playback_volume(me)) + e->direction = PA_ALSA_DIRECTION_OUTPUT; + else + return false; + } + } + + e->direction_try_other = false; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + r = snd_mixer_selem_get_playback_volume_range(me, &e->min_volume, &e->max_volume); + else + r = snd_mixer_selem_get_capture_volume_range(me, &e->min_volume, &e->max_volume); + + if (r < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to get volume range of %s: %s", buf, pa_alsa_strerror(r)); + return false; + } + + if (e->min_volume >= e->max_volume) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Your kernel driver is broken for element %s: it reports a volume range from %li to %li which makes no sense.", + buf, e->min_volume, e->max_volume); + return false; + } + if (e->volume_use == PA_ALSA_VOLUME_CONSTANT && (e->min_volume > e->constant_volume || e->max_volume < e->constant_volume)) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Constant volume %li configured for element %s, but the available range is from %li to %li.", + e->constant_volume, buf, e->min_volume, e->max_volume); + return false; + } + + + if (e->db_fix && ((e->min_volume > e->db_fix->min_step) || (e->max_volume < e->db_fix->max_step))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("The step range of the decibel fix for element %s (%li-%li) doesn't fit to the " + "real hardware range (%li-%li). Disabling the decibel fix.", buf, + e->db_fix->min_step, e->db_fix->max_step, e->min_volume, e->max_volume); + + decibel_fix_free(e->db_fix); + e->db_fix = NULL; + } + + if (e->db_fix) { + e->has_dB = true; + e->min_volume = e->db_fix->min_step; + e->max_volume = e->db_fix->max_step; + min_dB = e->db_fix->db_values[0]; + max_dB = e->db_fix->db_values[e->db_fix->max_step - e->db_fix->min_step]; + } else if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + e->has_dB = snd_mixer_selem_get_playback_dB_range(me, &min_dB, &max_dB) >= 0; + else + e->has_dB = snd_mixer_selem_get_capture_dB_range(me, &min_dB, &max_dB) >= 0; + + /* Check that the kernel driver returns consistent limits with + * both _get_*_dB_range() and _ask_*_vol_dB(). */ + if (e->has_dB && !e->db_fix) { + long min_dB_checked = 0; + long max_dB_checked = 0; + + if (element_ask_vol_dB(me, e->direction, e->min_volume, &min_dB_checked) < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to query the dB value for %s at volume level %li", buf, e->min_volume); + return false; + } + + if (element_ask_vol_dB(me, e->direction, e->max_volume, &max_dB_checked) < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to query the dB value for %s at volume level %li", buf, e->max_volume); + return false; + } + + if (min_dB != min_dB_checked || max_dB != max_dB_checked) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Your kernel driver is broken: the reported dB range for %s (from %0.2f dB to %0.2f dB) " + "doesn't match the dB values at minimum and maximum volume levels: %0.2f dB at level %li, " + "%0.2f dB at level %li.", buf, min_dB / 100.0, max_dB / 100.0, + min_dB_checked / 100.0, e->min_volume, max_dB_checked / 100.0, e->max_volume); + return false; + } + } + + if (e->has_dB) { + e->min_dB = ((double) min_dB) / 100.0; + e->max_dB = ((double) max_dB) / 100.0; + + if (min_dB >= max_dB) { + pa_assert(!e->db_fix); + pa_log_warn("Your kernel driver is broken: it reports a volume range from %0.2f dB to %0.2f dB which makes no sense.", + e->min_dB, e->max_dB); + e->has_dB = false; + } + } + + if (e->volume_limit >= 0) { + if (e->volume_limit <= e->min_volume || e->volume_limit > e->max_volume) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Volume limit for element %s of path %s is invalid: %li isn't within the valid range " + "%li-%li. The volume limit is ignored.", + buf, e->path->name, e->volume_limit, e->min_volume + 1, e->max_volume); + } else { + e->max_volume = e->volume_limit; + + if (e->has_dB) { + if (e->db_fix) { + e->db_fix->max_step = e->max_volume; + e->max_dB = ((double) e->db_fix->db_values[e->db_fix->max_step - e->db_fix->min_step]) / 100.0; + } else if (element_ask_vol_dB(me, e->direction, e->max_volume, &max_dB) < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to get dB value of %s: %s", buf, pa_alsa_strerror(r)); + e->has_dB = false; + } else + e->max_dB = ((double) max_dB) / 100.0; + } + } + } + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + is_mono = snd_mixer_selem_is_playback_mono(me) > 0; + else + is_mono = snd_mixer_selem_is_capture_mono(me) > 0; + + if (is_mono) { + e->n_channels = 1; + + if ((e->override_map & (1 << (e->n_channels-1))) && e->masks[SND_MIXER_SCHN_MONO][e->n_channels-1] == 0) { + pa_log_warn("Override map for mono element %s is invalid, ignoring override map", e->path->name); + e->override_map &= ~(1 << (e->n_channels-1)); + } + if (!(e->override_map & (1 << (e->n_channels-1)))) { + for (p = PA_CHANNEL_POSITION_FRONT_LEFT; p < PA_CHANNEL_POSITION_MAX; p++) { + if (alsa_channel_ids[p] == SND_MIXER_SCHN_UNKNOWN) + continue; + e->masks[alsa_channel_ids[p]][e->n_channels-1] = 0; + } + e->masks[SND_MIXER_SCHN_MONO][e->n_channels-1] = PA_CHANNEL_POSITION_MASK_ALL; + } + e->merged_mask = e->masks[SND_MIXER_SCHN_MONO][e->n_channels-1]; + return true; + } + + e->n_channels = 0; + for (p = PA_CHANNEL_POSITION_FRONT_LEFT; p < PA_CHANNEL_POSITION_MAX; p++) { + if (alsa_channel_ids[p] == SND_MIXER_SCHN_UNKNOWN) + continue; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + e->n_channels += snd_mixer_selem_has_playback_channel(me, alsa_channel_ids[p]) > 0; + else + e->n_channels += snd_mixer_selem_has_capture_channel(me, alsa_channel_ids[p]) > 0; + } + + if (e->n_channels <= 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Volume element %s with no channels?", buf); + return false; + } else if (e->n_channels > POSITION_MASK_CHANNELS) { + /* FIXME: In some places code like this is used: + * + * e->masks[alsa_channel_ids[p]][e->n_channels-1] + * + * The definition of e->masks is + * + * pa_channel_position_mask_t masks[SND_MIXER_SCHN_LAST + 1][POSITION_MASK_CHANNELS]; + * + * Since the array size is fixed at POSITION_MASK_CHANNELS, we obviously + * don't support elements with more than POSITION_MASK_CHANNELS + * channels... */ + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Volume element %s has %u channels. That's too much! I can't handle that!", buf, e->n_channels); + return false; + } + +retry: + if (!(e->override_map & (1 << (e->n_channels-1)))) { + for (p = PA_CHANNEL_POSITION_FRONT_LEFT; p < PA_CHANNEL_POSITION_MAX; p++) { + bool has_channel; + + if (alsa_channel_ids[p] == SND_MIXER_SCHN_UNKNOWN) + continue; + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + has_channel = snd_mixer_selem_has_playback_channel(me, alsa_channel_ids[p]) > 0; + else + has_channel = snd_mixer_selem_has_capture_channel(me, alsa_channel_ids[p]) > 0; + + e->masks[alsa_channel_ids[p]][e->n_channels-1] = has_channel ? PA_CHANNEL_POSITION_MASK(p) : 0; + } + } + + e->merged_mask = 0; + for (p = PA_CHANNEL_POSITION_FRONT_LEFT; p < PA_CHANNEL_POSITION_MAX; p++) { + if (alsa_channel_ids[p] == SND_MIXER_SCHN_UNKNOWN) + continue; + + e->merged_mask |= e->masks[alsa_channel_ids[p]][e->n_channels-1]; + } + + if (e->merged_mask == 0) { + if (!(e->override_map & (1 << (e->n_channels-1)))) { + pa_log_warn("Channel map for element %s is invalid", e->path->name); + return false; + } + pa_log_warn("Override map for element %s has empty result, ignoring override map", e->path->name); + e->override_map &= ~(1 << (e->n_channels-1)); + goto retry; + } + + return true; +} + +static int element_probe(pa_alsa_element *e, snd_mixer_t *m) { + snd_mixer_selem_id_t *sid; + snd_mixer_elem_t *me; + + pa_assert(m); + pa_assert(e); + pa_assert(e->path); + + SELEM_INIT(sid, &e->alsa_id); + + if (!(me = snd_mixer_find_selem(m, sid))) { + + if (e->required != PA_ALSA_REQUIRED_IGNORE) + return -1; + + e->switch_use = PA_ALSA_SWITCH_IGNORE; + e->volume_use = PA_ALSA_VOLUME_IGNORE; + e->enumeration_use = PA_ALSA_ENUMERATION_IGNORE; + + return 0; + } + + if (e->switch_use != PA_ALSA_SWITCH_IGNORE) { + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) { + + if (!snd_mixer_selem_has_playback_switch(me)) { + if (e->direction_try_other && snd_mixer_selem_has_capture_switch(me)) + e->direction = PA_ALSA_DIRECTION_INPUT; + else + e->switch_use = PA_ALSA_SWITCH_IGNORE; + } + + } else { + + if (!snd_mixer_selem_has_capture_switch(me)) { + if (e->direction_try_other && snd_mixer_selem_has_playback_switch(me)) + e->direction = PA_ALSA_DIRECTION_OUTPUT; + else + e->switch_use = PA_ALSA_SWITCH_IGNORE; + } + } + + if (e->switch_use != PA_ALSA_SWITCH_IGNORE) + e->direction_try_other = false; + } + + if (!element_probe_volume(e, me)) + e->volume_use = PA_ALSA_VOLUME_IGNORE; + + if (e->switch_use == PA_ALSA_SWITCH_SELECT) { + pa_alsa_option *o; + + PA_LLIST_FOREACH(o, e->options) + o->alsa_idx = pa_streq(o->alsa_name, "on") ? 1 : 0; + } else if (e->enumeration_use == PA_ALSA_ENUMERATION_SELECT) { + int n; + pa_alsa_option *o; + + if ((n = snd_mixer_selem_get_enum_items(me)) < 0) { + pa_log("snd_mixer_selem_get_enum_items() failed: %s", pa_alsa_strerror(n)); + return -1; + } + + PA_LLIST_FOREACH(o, e->options) { + int i; + + for (i = 0; i < n; i++) { + char buf[128]; + + if (snd_mixer_selem_get_enum_item_name(me, i, sizeof(buf), buf) < 0) + continue; + + if (!pa_streq(buf, o->alsa_name)) + continue; + + o->alsa_idx = i; + } + } + } + + if (check_required(e, me) < 0) + return -1; + + return 0; +} + +static int jack_probe(pa_alsa_jack *j, pa_alsa_mapping *mapping, snd_mixer_t *m) { + bool has_control; + + pa_assert(j); + pa_assert(j->path); + + if (j->append_pcm_to_name) { + char *new_name; + + if (!mapping) { + /* This could also be an assertion, because this should never + * happen. At the time of writing, mapping can only be NULL when + * module-alsa-sink/source synthesizes a path, and those + * synthesized paths never have any jacks, so jack_probe() should + * never be called with a NULL mapping. */ + pa_log("Jack %s: append_pcm_to_name is set, but mapping is NULL. Can't use this jack.", j->name); + return -1; + } + + new_name = pa_sprintf_malloc("%s,pcm=%i Jack", j->name, mapping->hw_device_index); + pa_xfree(j->alsa_id.name); + j->alsa_id.name = new_name; + j->append_pcm_to_name = false; + } + + has_control = pa_alsa_mixer_find_card(m, &j->alsa_id, 0) != NULL; + pa_alsa_jack_set_has_control(j, has_control); + + if (j->has_control) { + if (j->required_absent != PA_ALSA_REQUIRED_IGNORE) + return -1; + if (j->required_any != PA_ALSA_REQUIRED_IGNORE) + j->path->req_any_present = true; + } else { + if (j->required != PA_ALSA_REQUIRED_IGNORE) + return -1; + } + + return 0; +} + +pa_alsa_element * pa_alsa_element_get(pa_alsa_path *p, const char *section, bool prefixed) { + pa_alsa_element *e; + char *name; + int index; + + pa_assert(p); + pa_assert(section); + + if (prefixed) { + if (!pa_startswith(section, "Element ")) + return NULL; + + section += 8; + } + + /* This is not an element section, but an enum section? */ + if (strchr(section, ':')) + return NULL; + + name = alloca(strlen(section) + 1); + if (alsa_id_decode(section, name, &index)) + return NULL; + + if (p->last_element && pa_streq(p->last_element->alsa_id.name, name) && + p->last_element->alsa_id.index == index) + return p->last_element; + + PA_LLIST_FOREACH(e, p->elements) + if (pa_streq(e->alsa_id.name, name) && e->alsa_id.index == index) + goto finish; + + e = pa_xnew0(pa_alsa_element, 1); + e->path = p; + e->alsa_id.name = pa_xstrdup(name); + e->alsa_id.index = index; + e->direction = p->direction; + e->volume_limit = -1; + + PA_LLIST_INSERT_AFTER(pa_alsa_element, p->elements, p->last_element, e); + +finish: + p->last_element = e; + return e; +} + +static pa_alsa_jack* jack_get(pa_alsa_path *p, const char *section) { + pa_alsa_jack *j; + char *name; + int index; + + if (!pa_startswith(section, "Jack ")) + return NULL; + section += 5; + + name = alloca(strlen(section) + 1); + if (alsa_id_decode(section, name, &index)) + return NULL; + + if (p->last_jack && pa_streq(p->last_jack->name, name) && + p->last_jack->alsa_id.index == index) + return p->last_jack; + + PA_LLIST_FOREACH(j, p->jacks) + if (pa_streq(j->name, name) && j->alsa_id.index == index) + goto finish; + + j = pa_alsa_jack_new(p, NULL, name, index); + PA_LLIST_INSERT_AFTER(pa_alsa_jack, p->jacks, p->last_jack, j); + +finish: + p->last_jack = j; + return j; +} + +static pa_alsa_option* option_get(pa_alsa_path *p, const char *section) { + char *en, *name; + const char *on; + pa_alsa_option *o; + pa_alsa_element *e; + size_t len; + int index; + + if (!pa_startswith(section, "Option ")) + return NULL; + + section += 7; + + /* This is not an enum section, but an element section? */ + if (!(on = strchr(section, ':'))) + return NULL; + + len = on - section; + en = alloca(len + 1); + strncpy(en, section, len); + en[len] = '\0'; + + name = alloca(strlen(en) + 1); + if (alsa_id_decode(en, name, &index)) + return NULL; + + on++; + + if (p->last_option && + pa_streq(p->last_option->element->alsa_id.name, name) && + p->last_option->element->alsa_id.index == index && + pa_streq(p->last_option->alsa_name, on)) { + return p->last_option; + } + + pa_assert_se(e = pa_alsa_element_get(p, en, false)); + + PA_LLIST_FOREACH(o, e->options) + if (pa_streq(o->alsa_name, on)) + goto finish; + + o = pa_xnew0(pa_alsa_option, 1); + o->element = e; + o->alsa_name = pa_xstrdup(on); + o->alsa_idx = -1; + + if (p->last_option && p->last_option->element == e) + PA_LLIST_INSERT_AFTER(pa_alsa_option, e->options, p->last_option, o); + else + PA_LLIST_PREPEND(pa_alsa_option, e->options, o); + +finish: + p->last_option = o; + return o; +} + +static int element_parse_switch(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + + pa_assert(state); + + p = state->userdata; + + if (!(e = pa_alsa_element_get(p, state->section, true))) { + pa_log("[%s:%u] Switch makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->rvalue, "ignore")) + e->switch_use = PA_ALSA_SWITCH_IGNORE; + else if (pa_streq(state->rvalue, "mute")) + e->switch_use = PA_ALSA_SWITCH_MUTE; + else if (pa_streq(state->rvalue, "off")) + e->switch_use = PA_ALSA_SWITCH_OFF; + else if (pa_streq(state->rvalue, "on")) + e->switch_use = PA_ALSA_SWITCH_ON; + else if (pa_streq(state->rvalue, "select")) + e->switch_use = PA_ALSA_SWITCH_SELECT; + else { + pa_log("[%s:%u] Switch invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int element_parse_volume(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + + pa_assert(state); + + p = state->userdata; + + if (!(e = pa_alsa_element_get(p, state->section, true))) { + pa_log("[%s:%u] Volume makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->rvalue, "ignore")) + e->volume_use = PA_ALSA_VOLUME_IGNORE; + else if (pa_streq(state->rvalue, "merge")) + e->volume_use = PA_ALSA_VOLUME_MERGE; + else if (pa_streq(state->rvalue, "off")) + e->volume_use = PA_ALSA_VOLUME_OFF; + else if (pa_streq(state->rvalue, "zero")) + e->volume_use = PA_ALSA_VOLUME_ZERO; + else { + uint32_t constant; + + if (pa_atou(state->rvalue, &constant) >= 0) { + e->volume_use = PA_ALSA_VOLUME_CONSTANT; + e->constant_volume = constant; + } else { + pa_log("[%s:%u] Volume invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + } + + return 0; +} + +static int element_parse_enumeration(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + + pa_assert(state); + + p = state->userdata; + + if (!(e = pa_alsa_element_get(p, state->section, true))) { + pa_log("[%s:%u] Enumeration makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->rvalue, "ignore")) + e->enumeration_use = PA_ALSA_ENUMERATION_IGNORE; + else if (pa_streq(state->rvalue, "select")) + e->enumeration_use = PA_ALSA_ENUMERATION_SELECT; + else { + pa_log("[%s:%u] Enumeration invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int parse_type(pa_config_parser_state *state) { + struct device_port_types { + const char *name; + pa_device_port_type_t type; + } device_port_types[] = { + { "unknown", PA_DEVICE_PORT_TYPE_UNKNOWN }, + { "aux", PA_DEVICE_PORT_TYPE_AUX }, + { "speaker", PA_DEVICE_PORT_TYPE_SPEAKER }, + { "headphones", PA_DEVICE_PORT_TYPE_HEADPHONES }, + { "line", PA_DEVICE_PORT_TYPE_LINE }, + { "mic", PA_DEVICE_PORT_TYPE_MIC }, + { "headset", PA_DEVICE_PORT_TYPE_HEADSET }, + { "handset", PA_DEVICE_PORT_TYPE_HANDSET }, + { "earpiece", PA_DEVICE_PORT_TYPE_EARPIECE }, + { "spdif", PA_DEVICE_PORT_TYPE_SPDIF }, + { "hdmi", PA_DEVICE_PORT_TYPE_HDMI }, + { "tv", PA_DEVICE_PORT_TYPE_TV }, + { "radio", PA_DEVICE_PORT_TYPE_RADIO }, + { "video", PA_DEVICE_PORT_TYPE_VIDEO }, + { "usb", PA_DEVICE_PORT_TYPE_USB }, + { "bluetooth", PA_DEVICE_PORT_TYPE_BLUETOOTH }, + { "portable", PA_DEVICE_PORT_TYPE_PORTABLE }, + { "handsfree", PA_DEVICE_PORT_TYPE_HANDSFREE }, + { "car", PA_DEVICE_PORT_TYPE_CAR }, + { "hifi", PA_DEVICE_PORT_TYPE_HIFI }, + { "phone", PA_DEVICE_PORT_TYPE_PHONE }, + { "network", PA_DEVICE_PORT_TYPE_NETWORK }, + { "analog", PA_DEVICE_PORT_TYPE_ANALOG }, + }; + pa_alsa_path *path; + unsigned int idx; + + path = state->userdata; + + for (idx = 0; idx < PA_ELEMENTSOF(device_port_types); idx++) + if (pa_streq(state->rvalue, device_port_types[idx].name)) { + path->device_port_type = device_port_types[idx].type; + return 0; + } + + pa_log("[%s:%u] Invalid value for option 'type': %s", state->filename, state->lineno, state->rvalue); + return -1; +} + +static int parse_eld_device(pa_config_parser_state *state) { + pa_alsa_path *path; + uint32_t eld_device; + + path = state->userdata; + + if (pa_atou(state->rvalue, &eld_device) >= 0) { + path->autodetect_eld_device = false; + path->eld_device = eld_device; + return 0; + } + + if (pa_streq(state->rvalue, "auto")) { + path->autodetect_eld_device = true; + path->eld_device = -1; + return 0; + } + + pa_log("[%s:%u] Invalid value for option 'eld-device': %s", state->filename, state->lineno, state->rvalue); + return -1; +} + +static int option_parse_priority(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_option *o; + uint32_t prio; + + pa_assert(state); + + p = state->userdata; + + if (!(o = option_get(p, state->section))) { + pa_log("[%s:%u] Priority makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_atou(state->rvalue, &prio) < 0) { + pa_log("[%s:%u] Priority invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + o->priority = prio; + return 0; +} + +static int option_parse_name(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_option *o; + + pa_assert(state); + + p = state->userdata; + + if (!(o = option_get(p, state->section))) { + pa_log("[%s:%u] Name makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + pa_xfree(o->name); + o->name = pa_xstrdup(state->rvalue); + + return 0; +} + +static int element_parse_required(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + pa_alsa_option *o; + pa_alsa_jack *j; + pa_alsa_required_t req; + + pa_assert(state); + + p = state->userdata; + + e = pa_alsa_element_get(p, state->section, true); + o = option_get(p, state->section); + j = jack_get(p, state->section); + if (!e && !o && !j) { + pa_log("[%s:%u] Required makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->rvalue, "ignore")) + req = PA_ALSA_REQUIRED_IGNORE; + else if (pa_streq(state->rvalue, "switch") && e) + req = PA_ALSA_REQUIRED_SWITCH; + else if (pa_streq(state->rvalue, "volume") && e) + req = PA_ALSA_REQUIRED_VOLUME; + else if (pa_streq(state->rvalue, "enumeration")) + req = PA_ALSA_REQUIRED_ENUMERATION; + else if (pa_streq(state->rvalue, "any")) + req = PA_ALSA_REQUIRED_ANY; + else { + pa_log("[%s:%u] Required invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->lvalue, "required-absent")) { + if (e) + e->required_absent = req; + if (o) + o->required_absent = req; + if (j) + j->required_absent = req; + } + else if (pa_streq(state->lvalue, "required-any")) { + if (e) { + e->required_any = req; + e->path->has_req_any |= (req != PA_ALSA_REQUIRED_IGNORE); + } + if (o) { + o->required_any = req; + o->element->path->has_req_any |= (req != PA_ALSA_REQUIRED_IGNORE); + } + if (j) { + j->required_any = req; + j->path->has_req_any |= (req != PA_ALSA_REQUIRED_IGNORE); + } + + } + else { + if (e) + e->required = req; + if (o) + o->required = req; + if (j) + j->required = req; + } + + return 0; +} + +static int element_parse_direction(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + + pa_assert(state); + + p = state->userdata; + + if (!(e = pa_alsa_element_get(p, state->section, true))) { + pa_log("[%s:%u] Direction makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->rvalue, "playback")) + e->direction = PA_ALSA_DIRECTION_OUTPUT; + else if (pa_streq(state->rvalue, "capture")) + e->direction = PA_ALSA_DIRECTION_INPUT; + else { + pa_log("[%s:%u] Direction invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int element_parse_direction_try_other(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + int yes; + + pa_assert(state); + + p = state->userdata; + + if (!(e = pa_alsa_element_get(p, state->section, true))) { + pa_log("[%s:%u] Direction makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if ((yes = pa_parse_boolean(state->rvalue)) < 0) { + pa_log("[%s:%u] Direction invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + e->direction_try_other = !!yes; + return 0; +} + +static int element_parse_volume_limit(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + long volume_limit; + + pa_assert(state); + + p = state->userdata; + + if (!(e = pa_alsa_element_get(p, state->section, true))) { + pa_log("[%s:%u] volume-limit makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_atol(state->rvalue, &volume_limit) < 0 || volume_limit < 0) { + pa_log("[%s:%u] Invalid value for volume-limit", state->filename, state->lineno); + return -1; + } + + e->volume_limit = volume_limit; + return 0; +} + +static unsigned int parse_channel_position(const char *m) +{ + pa_channel_position_t p; + + if ((p = pa_channel_position_from_string(m)) == PA_CHANNEL_POSITION_INVALID) + return SND_MIXER_SCHN_UNKNOWN; + + return alsa_channel_ids[p]; +} + +static pa_channel_position_mask_t parse_mask(const char *m) { + pa_channel_position_mask_t v; + + if (pa_streq(m, "all-left")) + v = PA_CHANNEL_POSITION_MASK_LEFT; + else if (pa_streq(m, "all-right")) + v = PA_CHANNEL_POSITION_MASK_RIGHT; + else if (pa_streq(m, "all-center")) + v = PA_CHANNEL_POSITION_MASK_CENTER; + else if (pa_streq(m, "all-front")) + v = PA_CHANNEL_POSITION_MASK_FRONT; + else if (pa_streq(m, "all-rear")) + v = PA_CHANNEL_POSITION_MASK_REAR; + else if (pa_streq(m, "all-side")) + v = PA_CHANNEL_POSITION_MASK_SIDE_OR_TOP_CENTER; + else if (pa_streq(m, "all-top")) + v = PA_CHANNEL_POSITION_MASK_TOP; + else if (pa_streq(m, "all-no-lfe")) + v = PA_CHANNEL_POSITION_MASK_ALL ^ PA_CHANNEL_POSITION_MASK(PA_CHANNEL_POSITION_LFE); + else if (pa_streq(m, "all")) + v = PA_CHANNEL_POSITION_MASK_ALL; + else { + pa_channel_position_t p; + + if ((p = pa_channel_position_from_string(m)) == PA_CHANNEL_POSITION_INVALID) + return 0; + + v = PA_CHANNEL_POSITION_MASK(p); + } + + return v; +} + +static int element_parse_override_map(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_element *e; + const char *split_state = NULL; + char *s; + unsigned i = 0; + int channel_count = 0; + char *n; + + pa_assert(state); + + p = state->userdata; + + if (!(e = pa_alsa_element_get(p, state->section, true))) { + pa_log("[%s:%u] Override map makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + s = strstr(state->lvalue, "."); + if (s) { + pa_atoi(s + 1, &channel_count); + if (channel_count < 1 || channel_count > POSITION_MASK_CHANNELS) { + pa_log("[%s:%u] Override map index '%s' invalid in '%s'", state->filename, state->lineno, state->lvalue, state->section); + return 0; + } + } else { + pa_log("[%s:%u] Invalid override map syntax '%s' in '%s'", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + while ((n = pa_split(state->rvalue, ",", &split_state))) { + pa_channel_position_mask_t m; + snd_mixer_selem_channel_id_t channel_position; + + if (i >= (unsigned)channel_count) { + pa_log("[%s:%u] Invalid override map size (>%d) in '%s'", state->filename, state->lineno, channel_count, state->section); + return -1; + } + channel_position = alsa_channel_positions[i]; + + if (!*n) + m = 0; + else { + s = strstr(n, ":"); + if (s) { + *s = '\0'; + s++; + channel_position = parse_channel_position(n); + if (channel_position == SND_MIXER_SCHN_UNKNOWN) { + pa_log("[%s:%u] Override map position '%s' invalid in '%s'", state->filename, state->lineno, n, state->section); + pa_xfree(n); + return -1; + } + } + if ((m = parse_mask(s ? s : n)) == 0) { + pa_log("[%s:%u] Override map '%s' invalid in '%s'", state->filename, state->lineno, s ? s : n, state->section); + pa_xfree(n); + return -1; + } + } + + if (e->masks[channel_position][channel_count-1]) { + pa_log("[%s:%u] Override map '%s' duplicate position '%s' in '%s'", state->filename, state->lineno, s ? s : n, snd_mixer_selem_channel_name(channel_position), state->section); + pa_xfree(n); + return -1; + } + e->override_map |= (1 << (channel_count - 1)); + e->masks[channel_position][channel_count-1] = m; + pa_xfree(n); + i++; + } + + return 0; +} + +static int jack_parse_state(pa_config_parser_state *state) { + pa_alsa_path *p; + pa_alsa_jack *j; + pa_available_t pa; + + pa_assert(state); + + p = state->userdata; + + if (!(j = jack_get(p, state->section))) { + pa_log("[%s:%u] state makes no sense in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->rvalue, "yes")) + pa = PA_AVAILABLE_YES; + else if (pa_streq(state->rvalue, "no")) + pa = PA_AVAILABLE_NO; + else if (pa_streq(state->rvalue, "unknown")) + pa = PA_AVAILABLE_UNKNOWN; + else { + pa_log("[%s:%u] state must be 'yes', 'no' or 'unknown' in '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->lvalue, "state.unplugged")) + j->state_unplugged = pa; + else { + j->state_plugged = pa; + pa_assert(pa_streq(state->lvalue, "state.plugged")); + } + + return 0; +} + +static int jack_parse_append_pcm_to_name(pa_config_parser_state *state) { + pa_alsa_path *path; + pa_alsa_jack *jack; + int b; + + pa_assert(state); + + path = state->userdata; + if (!(jack = jack_get(path, state->section))) { + pa_log("[%s:%u] Option 'append_pcm_to_name' not expected in section '%s'", + state->filename, state->lineno, state->section); + return -1; + } + + b = pa_parse_boolean(state->rvalue); + if (b < 0) { + pa_log("[%s:%u] Invalid value for 'append_pcm_to_name': %s", state->filename, state->lineno, state->rvalue); + return -1; + } + + jack->append_pcm_to_name = b; + return 0; +} + +static int element_set_option(pa_alsa_element *e, snd_mixer_t *m, int alsa_idx) { + snd_mixer_selem_id_t *sid; + snd_mixer_elem_t *me; + char buf[64]; + int r; + + pa_assert(e); + pa_assert(m); + + SELEM_INIT(sid, &e->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return -1; + } + + if (e->switch_use == PA_ALSA_SWITCH_SELECT) { + + if (e->direction == PA_ALSA_DIRECTION_OUTPUT) + r = snd_mixer_selem_set_playback_switch_all(me, alsa_idx); + else + r = snd_mixer_selem_set_capture_switch_all(me, alsa_idx); + + if (r < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to set switch of %s: %s", buf, pa_alsa_strerror(errno)); + } + + } else { + pa_assert(e->enumeration_use == PA_ALSA_ENUMERATION_SELECT); + + if ((r = snd_mixer_selem_set_enum_item(me, 0, alsa_idx)) < 0) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Failed to set enumeration of %s: %s", buf, pa_alsa_strerror(errno)); + } + } + + return r; +} + +static int setting_select(pa_alsa_setting *s, snd_mixer_t *m) { + pa_alsa_option *o; + uint32_t idx; + + pa_assert(s); + pa_assert(m); + + PA_IDXSET_FOREACH(o, s->options, idx) + element_set_option(o->element, m, o->alsa_idx); + + return 0; +} + +static int option_verify(pa_alsa_option *o) { + static const struct description_map well_known_descriptions[] = { + { "input", N_("Input") }, + { "input-docking", N_("Docking Station Input") }, + { "input-docking-microphone", N_("Docking Station Microphone") }, + { "input-docking-linein", N_("Docking Station Line In") }, + { "input-linein", N_("Line In") }, + { "input-microphone", N_("Microphone") }, + { "input-microphone-front", N_("Front Microphone") }, + { "input-microphone-rear", N_("Rear Microphone") }, + { "input-microphone-external", N_("External Microphone") }, + { "input-microphone-internal", N_("Internal Microphone") }, + { "input-radio", N_("Radio") }, + { "input-video", N_("Video") }, + { "input-agc-on", N_("Automatic Gain Control") }, + { "input-agc-off", N_("No Automatic Gain Control") }, + { "input-boost-on", N_("Boost") }, + { "input-boost-off", N_("No Boost") }, + { "output-amplifier-on", N_("Amplifier") }, + { "output-amplifier-off", N_("No Amplifier") }, + { "output-bass-boost-on", N_("Bass Boost") }, + { "output-bass-boost-off", N_("No Bass Boost") }, + { "output-speaker", N_("Speaker") }, + { "output-headphones", N_("Headphones") } + }; + char buf[64]; + + pa_assert(o); + + if (!o->name) { + pa_log("No name set for option %s", o->alsa_name); + return -1; + } + + if (o->element->enumeration_use != PA_ALSA_ENUMERATION_SELECT && + o->element->switch_use != PA_ALSA_SWITCH_SELECT) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &o->element->alsa_id); + pa_log("Element %s of option %s not set for select.", buf, o->name); + return -1; + } + + if (o->element->switch_use == PA_ALSA_SWITCH_SELECT && + !pa_streq(o->alsa_name, "on") && + !pa_streq(o->alsa_name, "off")) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &o->element->alsa_id); + pa_log("Switch %s options need be named off or on ", buf); + return -1; + } + + if (!o->description) + o->description = pa_xstrdup(lookup_description(o->name, + well_known_descriptions, + PA_ELEMENTSOF(well_known_descriptions))); + if (!o->description) + o->description = pa_xstrdup(o->name); + + return 0; +} + +static int element_verify(pa_alsa_element *e) { + pa_alsa_option *o; + char buf[64]; + + pa_assert(e); + +// pa_log_debug("Element %s, path %s: r=%d, r-any=%d, r-abs=%d", e->alsa_name, e->path->name, e->required, e->required_any, e->required_absent); + if ((e->required != PA_ALSA_REQUIRED_IGNORE && e->required == e->required_absent) || + (e->required_any != PA_ALSA_REQUIRED_IGNORE && e->required_any == e->required_absent) || + (e->required_absent == PA_ALSA_REQUIRED_ANY && e->required_any != PA_ALSA_REQUIRED_IGNORE) || + (e->required_absent == PA_ALSA_REQUIRED_ANY && e->required != PA_ALSA_REQUIRED_IGNORE)) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log("Element %s cannot be required and absent at the same time.", buf); + return -1; + } + + if (e->switch_use == PA_ALSA_SWITCH_SELECT && e->enumeration_use == PA_ALSA_ENUMERATION_SELECT) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log("Element %s cannot set select for both switch and enumeration.", buf); + return -1; + } + + PA_LLIST_FOREACH(o, e->options) + if (option_verify(o) < 0) + return -1; + + return 0; +} + +static int path_verify(pa_alsa_path *p) { + static const struct description2_map well_known_descriptions[] = { + { "analog-input", N_("Analog Input"), PA_DEVICE_PORT_TYPE_ANALOG }, + { "analog-input-microphone", N_("Microphone"), PA_DEVICE_PORT_TYPE_MIC }, + { "analog-input-microphone-front", N_("Front Microphone"), PA_DEVICE_PORT_TYPE_MIC }, + { "analog-input-microphone-rear", N_("Rear Microphone"), PA_DEVICE_PORT_TYPE_MIC }, + { "analog-input-microphone-dock", N_("Dock Microphone"), PA_DEVICE_PORT_TYPE_MIC }, + { "analog-input-microphone-internal", N_("Internal Microphone"), PA_DEVICE_PORT_TYPE_MIC }, + { "analog-input-microphone-headset", N_("Headset Microphone"), PA_DEVICE_PORT_TYPE_HEADSET }, + { "analog-input-linein", N_("Line In"), PA_DEVICE_PORT_TYPE_LINE }, + { "analog-input-radio", N_("Radio"), PA_DEVICE_PORT_TYPE_RADIO }, + { "analog-input-video", N_("Video"), PA_DEVICE_PORT_TYPE_VIDEO }, + { "analog-output", N_("Analog Output"), PA_DEVICE_PORT_TYPE_ANALOG }, + { "analog-output-headphones", N_("Headphones"), PA_DEVICE_PORT_TYPE_HEADPHONES }, + { "analog-output-headphones-2", N_("Headphones 2"), PA_DEVICE_PORT_TYPE_HEADPHONES }, + { "analog-output-headphones-mono", N_("Headphones Mono Output"), PA_DEVICE_PORT_TYPE_HEADPHONES }, + { "analog-output-lineout", N_("Line Out"), PA_DEVICE_PORT_TYPE_LINE }, + { "analog-output-mono", N_("Analog Mono Output"), PA_DEVICE_PORT_TYPE_ANALOG }, + { "analog-output-speaker", N_("Speakers"), PA_DEVICE_PORT_TYPE_SPEAKER }, + { "hdmi-output", N_("HDMI / DisplayPort"), PA_DEVICE_PORT_TYPE_HDMI }, + { "iec958-stereo-output", N_("Digital Output (S/PDIF)"), PA_DEVICE_PORT_TYPE_SPDIF }, + { "iec958-stereo-input", N_("Digital Input (S/PDIF)"), PA_DEVICE_PORT_TYPE_SPDIF }, + { "multichannel-input", N_("Multichannel Input"), PA_DEVICE_PORT_TYPE_LINE }, + { "multichannel-output", N_("Multichannel Output"), PA_DEVICE_PORT_TYPE_LINE }, + { "steelseries-arctis-output-game-common", N_("Game Output"), PA_DEVICE_PORT_TYPE_HEADSET }, + { "steelseries-arctis-output-chat-common", N_("Chat Output"), PA_DEVICE_PORT_TYPE_HEADSET }, + }; + + pa_alsa_element *e; + const char *key = p->description_key ? p->description_key : p->name; + const struct description2_map *map = lookup_description2(key, + well_known_descriptions, + PA_ELEMENTSOF(well_known_descriptions)); + + pa_assert(p); + + PA_LLIST_FOREACH(e, p->elements) + if (element_verify(e) < 0) + return -1; + + if (map) { + if (p->device_port_type == PA_DEVICE_PORT_TYPE_UNKNOWN) + p->device_port_type = map->type; + if (!p->description) + p->description = pa_xstrdup(map->description); + } + + if (!p->description) { + if (p->description_key) + pa_log_warn("Path %s: Unrecognized description key: %s", p->name, p->description_key); + + p->description = pa_xstrdup(p->name); + } + + return 0; +} + +static const char *get_default_paths_dir(void) { +#ifdef HAVE_RUNNING_FROM_BUILD_TREE + if (pa_run_from_build_tree()) + return PA_SRCDIR "/modules/alsa/mixer/paths/"; + else +#endif + return PA_ALSA_PATHS_DIR; +} + +pa_alsa_path* pa_alsa_path_new(const char *paths_dir, const char *fname, pa_alsa_direction_t direction) { + pa_alsa_path *p; + char *fn; + int r; + const char *n; + bool mute_during_activation = false; + + pa_config_item items[] = { + /* [General] */ + { "priority", pa_config_parse_unsigned, NULL, "General" }, + { "description-key", pa_config_parse_string, NULL, "General" }, + { "description", pa_config_parse_string, NULL, "General" }, + { "mute-during-activation", pa_config_parse_bool, NULL, "General" }, + { "type", parse_type, NULL, "General" }, + { "eld-device", parse_eld_device, NULL, "General" }, + + /* [Option ...] */ + { "priority", option_parse_priority, NULL, NULL }, + { "name", option_parse_name, NULL, NULL }, + + /* [Jack ...] */ + { "state.plugged", jack_parse_state, NULL, NULL }, + { "state.unplugged", jack_parse_state, NULL, NULL }, + { "append-pcm-to-name", jack_parse_append_pcm_to_name, NULL, NULL }, + + /* [Element ...] */ + { "switch", element_parse_switch, NULL, NULL }, + { "volume", element_parse_volume, NULL, NULL }, + { "enumeration", element_parse_enumeration, NULL, NULL }, + { "override-map.1", element_parse_override_map, NULL, NULL }, + { "override-map.2", element_parse_override_map, NULL, NULL }, + { "override-map.3", element_parse_override_map, NULL, NULL }, + { "override-map.4", element_parse_override_map, NULL, NULL }, + { "override-map.5", element_parse_override_map, NULL, NULL }, + { "override-map.6", element_parse_override_map, NULL, NULL }, + { "override-map.7", element_parse_override_map, NULL, NULL }, + { "override-map.8", element_parse_override_map, NULL, NULL }, +#if POSITION_MASK_CHANNELS > 8 +#error "Add override-map.9+ definitions" +#endif + /* ... later on we might add override-map.3 and so on here ... */ + { "required", element_parse_required, NULL, NULL }, + { "required-any", element_parse_required, NULL, NULL }, + { "required-absent", element_parse_required, NULL, NULL }, + { "direction", element_parse_direction, NULL, NULL }, + { "direction-try-other", element_parse_direction_try_other, NULL, NULL }, + { "volume-limit", element_parse_volume_limit, NULL, NULL }, + { NULL, NULL, NULL, NULL } + }; + + pa_assert(fname); + + p = pa_xnew0(pa_alsa_path, 1); + n = pa_path_get_filename(fname); + p->name = pa_xstrndup(n, strcspn(n, ".")); + p->proplist = pa_proplist_new(); + p->direction = direction; + p->eld_device = -1; + + items[0].data = &p->priority; + items[1].data = &p->description_key; + items[2].data = &p->description; + items[3].data = &mute_during_activation; + + if (!paths_dir) + paths_dir = get_default_paths_dir(); + + fn = pa_maybe_prefix_path(fname, paths_dir); + + r = pa_config_parse(fn, NULL, items, p->proplist, false, p); + pa_xfree(fn); + + if (r < 0) + goto fail; + + p->mute_during_activation = mute_during_activation; + + if (path_verify(p) < 0) + goto fail; + + return p; + +fail: + pa_alsa_path_free(p); + return NULL; +} + +pa_alsa_path *pa_alsa_path_synthesize(const char *element, pa_alsa_direction_t direction) { + pa_alsa_path *p; + pa_alsa_element *e; + char *name; + int index; + + pa_assert(element); + + name = alloca(strlen(element) + 1); + if (alsa_id_decode(element, name, &index)) + return NULL; + + p = pa_xnew0(pa_alsa_path, 1); + p->name = pa_xstrdup(element); + p->direction = direction; + p->proplist = pa_proplist_new(); + + e = pa_xnew0(pa_alsa_element, 1); + e->path = p; + e->alsa_id.name = pa_xstrdup(name); + e->alsa_id.index = index; + e->direction = direction; + e->volume_limit = -1; + + e->switch_use = PA_ALSA_SWITCH_MUTE; + e->volume_use = PA_ALSA_VOLUME_MERGE; + + PA_LLIST_PREPEND(pa_alsa_element, p->elements, e); + p->last_element = e; + return p; +} + +static bool element_drop_unsupported(pa_alsa_element *e) { + pa_alsa_option *o, *n; + + pa_assert(e); + + for (o = e->options; o; o = n) { + n = o->next; + + if (o->alsa_idx < 0) { + PA_LLIST_REMOVE(pa_alsa_option, e->options, o); + option_free(o); + } + } + + return + e->switch_use != PA_ALSA_SWITCH_IGNORE || + e->volume_use != PA_ALSA_VOLUME_IGNORE || + e->enumeration_use != PA_ALSA_ENUMERATION_IGNORE; +} + +static void path_drop_unsupported(pa_alsa_path *p) { + pa_alsa_element *e, *n; + + pa_assert(p); + + for (e = p->elements; e; e = n) { + n = e->next; + + if (!element_drop_unsupported(e)) { + PA_LLIST_REMOVE(pa_alsa_element, p->elements, e); + element_free(e); + } + } +} + +static void path_make_options_unique(pa_alsa_path *p) { + pa_alsa_element *e; + pa_alsa_option *o, *u; + + PA_LLIST_FOREACH(e, p->elements) { + PA_LLIST_FOREACH(o, e->options) { + unsigned i; + char *m; + + for (u = o->next; u; u = u->next) + if (pa_streq(u->name, o->name)) + break; + + if (!u) + continue; + + m = pa_xstrdup(o->name); + + /* OK, this name is not unique, hence let's rename */ + for (i = 1, u = o; u; u = u->next) { + char *nn, *nd; + + if (!pa_streq(u->name, m)) + continue; + + nn = pa_sprintf_malloc("%s-%u", m, i); + pa_xfree(u->name); + u->name = nn; + + nd = pa_sprintf_malloc("%s %u", u->description, i); + pa_xfree(u->description); + u->description = nd; + + i++; + } + + pa_xfree(m); + } + } +} + +static bool element_create_settings(pa_alsa_element *e, pa_alsa_setting *template) { + pa_alsa_option *o; + + for (; e; e = e->next) + if (e->switch_use == PA_ALSA_SWITCH_SELECT || + e->enumeration_use == PA_ALSA_ENUMERATION_SELECT) + break; + + if (!e) + return false; + + for (o = e->options; o; o = o->next) { + pa_alsa_setting *s; + + if (template) { + s = pa_xnewdup(pa_alsa_setting, template, 1); + s->options = pa_idxset_copy(template->options, NULL); + s->name = pa_sprintf_malloc("%s+%s", template->name, o->name); + s->description = + (template->description[0] && o->description[0]) + ? pa_sprintf_malloc("%s / %s", template->description, o->description) + : (template->description[0] + ? pa_xstrdup(template->description) + : pa_xstrdup(o->description)); + + s->priority = PA_MAX(template->priority, o->priority); + } else { + s = pa_xnew0(pa_alsa_setting, 1); + s->options = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + s->name = pa_xstrdup(o->name); + s->description = pa_xstrdup(o->description); + s->priority = o->priority; + } + + pa_idxset_put(s->options, o, NULL); + + if (element_create_settings(e->next, s)) + /* This is not a leaf, so let's get rid of it */ + setting_free(s); + else { + /* This is a leaf, so let's add it */ + PA_LLIST_INSERT_AFTER(pa_alsa_setting, e->path->settings, e->path->last_setting, s); + + e->path->last_setting = s; + } + } + + return true; +} + +static void path_create_settings(pa_alsa_path *p) { + pa_assert(p); + + element_create_settings(p->elements, NULL); +} + +int pa_alsa_path_probe(pa_alsa_path *p, pa_alsa_mapping *mapping, snd_mixer_t *m, bool ignore_dB) { + pa_alsa_element *e; + pa_alsa_jack *j; + double min_dB[PA_CHANNEL_POSITION_MAX], max_dB[PA_CHANNEL_POSITION_MAX]; + pa_channel_position_t t; + pa_channel_position_mask_t path_volume_channels = 0; + bool min_dB_set, max_dB_set; + char buf[64]; + + pa_assert(p); + pa_assert(m); + + if (p->probed) + return p->supported ? 0 : -1; + p->probed = true; + + pa_zero(min_dB); + pa_zero(max_dB); + + pa_log_debug("Probing path '%s'", p->name); + + PA_LLIST_FOREACH(j, p->jacks) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &j->alsa_id); + if (jack_probe(j, mapping, m) < 0) { + p->supported = false; + pa_log_debug("Probe of jack %s failed.", buf); + return -1; + } + pa_log_debug("Probe of jack %s succeeded (%s)", buf, j->has_control ? "found!" : "not found"); + } + + PA_LLIST_FOREACH(e, p->elements) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + if (element_probe(e, m) < 0) { + p->supported = false; + pa_log_debug("Probe of element %s failed.", buf); + return -1; + } + pa_log_debug("Probe of element %s succeeded (volume=%d, switch=%d, enumeration=%d, has_dB=%d).", buf, e->volume_use, e->switch_use, e->enumeration_use, e->has_dB); + + if (ignore_dB) + e->has_dB = false; + + if (e->volume_use == PA_ALSA_VOLUME_MERGE) { + + if (!p->has_volume) { + p->min_volume = e->min_volume; + p->max_volume = e->max_volume; + } + + if (e->has_dB) { + if (!p->has_volume) { + for (t = 0; t < PA_CHANNEL_POSITION_MAX; t++) + if (PA_CHANNEL_POSITION_MASK(t) & e->merged_mask) { + min_dB[t] = e->min_dB; + max_dB[t] = e->max_dB; + path_volume_channels |= PA_CHANNEL_POSITION_MASK(t); + } + + p->has_dB = true; + } else { + + if (p->has_dB) { + for (t = 0; t < PA_CHANNEL_POSITION_MAX; t++) + if (PA_CHANNEL_POSITION_MASK(t) & e->merged_mask) { + min_dB[t] += e->min_dB; + max_dB[t] += e->max_dB; + path_volume_channels |= PA_CHANNEL_POSITION_MASK(t); + } + } else { + /* Hmm, there's another element before us + * which cannot do dB volumes, so we we need + * to 'neutralize' this slider */ + e->volume_use = PA_ALSA_VOLUME_ZERO; + pa_log_info("Zeroing volume of %s on path '%s'", buf, p->name); + } + } + } else if (p->has_volume) { + /* We can't use this volume, so let's ignore it */ + e->volume_use = PA_ALSA_VOLUME_IGNORE; + pa_log_info("Ignoring volume of %s on path '%s' (missing dB info)", buf, p->name); + } + p->has_volume = true; + } + + if (e->switch_use == PA_ALSA_SWITCH_MUTE) + p->has_mute = true; + } + + if (p->has_req_any && !p->req_any_present) { + p->supported = false; + pa_log_debug("Skipping path '%s', none of required-any elements preset.", p->name); + return -1; + } + + path_drop_unsupported(p); + path_make_options_unique(p); + path_create_settings(p); + + p->supported = true; + + p->min_dB = INFINITY; + min_dB_set = false; + p->max_dB = -INFINITY; + max_dB_set = false; + + for (t = 0; t < PA_CHANNEL_POSITION_MAX; t++) { + if (path_volume_channels & PA_CHANNEL_POSITION_MASK(t)) { + if (p->min_dB > min_dB[t]) { + p->min_dB = min_dB[t]; + min_dB_set = true; + } + + if (p->max_dB < max_dB[t]) { + p->max_dB = max_dB[t]; + max_dB_set = true; + } + } + } + + /* this is probably a wrong prediction, but it should be safe */ + if (!min_dB_set) + p->min_dB = -INFINITY; + if (!max_dB_set) + p->max_dB = 0; + + return 0; +} + +void pa_alsa_setting_dump(pa_alsa_setting *s) { + pa_assert(s); + + pa_log_debug("Setting %s (%s) priority=%u", + s->name, + pa_strnull(s->description), + s->priority); +} + +void pa_alsa_jack_dump(pa_alsa_jack *j) { + pa_assert(j); + + pa_log_debug("Jack %s, alsa_name='%s', index='%d', detection %s", j->name, j->alsa_id.name, j->alsa_id.index, j->has_control ? "possible" : "unavailable"); +} + +void pa_alsa_option_dump(pa_alsa_option *o) { + pa_assert(o); + + pa_log_debug("Option %s (%s/%s) index=%i, priority=%u", + o->alsa_name, + pa_strnull(o->name), + pa_strnull(o->description), + o->alsa_idx, + o->priority); +} + +void pa_alsa_element_dump(pa_alsa_element *e) { + char buf[64]; + + pa_alsa_option *o; + pa_assert(e); + + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_debug("Element %s, direction=%i, switch=%i, volume=%i, volume_limit=%li, enumeration=%i, required=%i, required_any=%i, required_absent=%i, mask=0x%llx, n_channels=%u, override_map=%02x", + buf, + e->direction, + e->switch_use, + e->volume_use, + e->volume_limit, + e->enumeration_use, + e->required, + e->required_any, + e->required_absent, + (long long unsigned) e->merged_mask, + e->n_channels, + e->override_map); + + PA_LLIST_FOREACH(o, e->options) + pa_alsa_option_dump(o); +} + +void pa_alsa_path_dump(pa_alsa_path *p) { + pa_alsa_element *e; + pa_alsa_jack *j; + pa_alsa_setting *s; + pa_assert(p); + + pa_log_debug("Path %s (%s), direction=%i, priority=%u, probed=%s, supported=%s, has_mute=%s, has_volume=%s, " + "has_dB=%s, min_volume=%li, max_volume=%li, min_dB=%g, max_dB=%g", + p->name, + pa_strnull(p->description), + p->direction, + p->priority, + pa_yes_no(p->probed), + pa_yes_no(p->supported), + pa_yes_no(p->has_mute), + pa_yes_no(p->has_volume), + pa_yes_no(p->has_dB), + p->min_volume, p->max_volume, + p->min_dB, p->max_dB); + + PA_LLIST_FOREACH(e, p->elements) + pa_alsa_element_dump(e); + + PA_LLIST_FOREACH(j, p->jacks) + pa_alsa_jack_dump(j); + + PA_LLIST_FOREACH(s, p->settings) + pa_alsa_setting_dump(s); +} + +static void element_set_callback(pa_alsa_element *e, snd_mixer_t *m, snd_mixer_elem_callback_t cb, void *userdata) { + snd_mixer_selem_id_t *sid; + snd_mixer_elem_t *me; + char buf[64]; + + pa_assert(e); + pa_assert(m); + pa_assert(cb); + + SELEM_INIT(sid, &e->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &e->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return; + } + + snd_mixer_elem_set_callback(me, cb); + snd_mixer_elem_set_callback_private(me, userdata); +} + +void pa_alsa_path_set_callback(pa_alsa_path *p, snd_mixer_t *m, snd_mixer_elem_callback_t cb, void *userdata) { + pa_alsa_element *e; + + pa_assert(p); + pa_assert(m); + pa_assert(cb); + + PA_LLIST_FOREACH(e, p->elements) + element_set_callback(e, m, cb, userdata); +} + +void pa_alsa_path_set_set_callback(pa_alsa_path_set *ps, snd_mixer_t *m, snd_mixer_elem_callback_t cb, void *userdata) { + pa_alsa_path *p; + void *state; + + pa_assert(ps); + pa_assert(m); + pa_assert(cb); + + PA_HASHMAP_FOREACH(p, ps->paths, state) + pa_alsa_path_set_callback(p, m, cb, userdata); +} + +static pa_alsa_path *profile_set_get_path(pa_alsa_profile_set *ps, const char *path_name) { + pa_alsa_path *path; + + pa_assert(ps); + pa_assert(path_name); + + if ((path = pa_hashmap_get(ps->output_paths, path_name))) + return path; + + return pa_hashmap_get(ps->input_paths, path_name); +} + +static void profile_set_add_path(pa_alsa_profile_set *ps, pa_alsa_path *path) { + pa_assert(ps); + pa_assert(path); + + switch (path->direction) { + case PA_ALSA_DIRECTION_OUTPUT: + pa_assert_se(pa_hashmap_put(ps->output_paths, path->name, path) >= 0); + break; + + case PA_ALSA_DIRECTION_INPUT: + pa_assert_se(pa_hashmap_put(ps->input_paths, path->name, path) >= 0); + break; + + default: + pa_assert_not_reached(); + } +} + +pa_alsa_path_set *pa_alsa_path_set_new(pa_alsa_mapping *m, pa_alsa_direction_t direction, const char *paths_dir) { + pa_alsa_path_set *ps; + char **pn = NULL, **en = NULL, **ie; + pa_alsa_decibel_fix *db_fix; + void *state, *state2; + char name[64]; + int index; + + pa_assert(m); + pa_assert(m->profile_set); + pa_assert(m->profile_set->decibel_fixes); + pa_assert(direction == PA_ALSA_DIRECTION_OUTPUT || direction == PA_ALSA_DIRECTION_INPUT); + + if (m->direction != PA_ALSA_DIRECTION_ANY && m->direction != direction) + return NULL; + + ps = pa_xnew0(pa_alsa_path_set, 1); + ps->direction = direction; + ps->paths = pa_hashmap_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + if (direction == PA_ALSA_DIRECTION_OUTPUT) + pn = m->output_path_names; + else + pn = m->input_path_names; + + if (pn) { + char **in; + + for (in = pn; *in; in++) { + pa_alsa_path *p = NULL; + bool duplicate = false; + char **kn; + + for (kn = pn; kn < in; kn++) + if (pa_streq(*kn, *in)) { + duplicate = true; + break; + } + + if (duplicate) + continue; + + p = profile_set_get_path(m->profile_set, *in); + + if (p && p->direction != direction) { + pa_log("Configuration error: Path %s is used both as an input and as an output path.", p->name); + goto fail; + } + + if (!p) { + char *fn = pa_sprintf_malloc("%s.conf", *in); + p = pa_alsa_path_new(paths_dir, fn, direction); + pa_xfree(fn); + if (p) + profile_set_add_path(m->profile_set, p); + } + + if (p) + pa_hashmap_put(ps->paths, p, p); + + } + + goto finish; + } + + if (direction == PA_ALSA_DIRECTION_OUTPUT) + en = m->output_element; + else + en = m->input_element; + + if (!en) + goto fail; + + for (ie = en; *ie; ie++) { + char **je; + pa_alsa_path *p; + + p = pa_alsa_path_synthesize(*ie, direction); + + /* Mark all other passed elements for require-absent */ + for (je = en; *je; je++) { + pa_alsa_element *e; + + if (je == ie) + continue; + + if (strlen(*je) + 1 >= sizeof(name)) { + pa_log("Element identifier %s is too long!", *je); + continue; + } + + if (alsa_id_decode(*je, name, &index)) + continue; + + e = pa_xnew0(pa_alsa_element, 1); + e->path = p; + e->alsa_id.name = pa_xstrdup(name); + e->alsa_id.index = index; + e->direction = direction; + e->required_absent = PA_ALSA_REQUIRED_ANY; + e->volume_limit = -1; + + PA_LLIST_INSERT_AFTER(pa_alsa_element, p->elements, p->last_element, e); + p->last_element = e; + } + + pa_hashmap_put(ps->paths, *ie, p); + } + +finish: + /* Assign decibel fixes to elements. */ + PA_HASHMAP_FOREACH(db_fix, m->profile_set->decibel_fixes, state) { + pa_alsa_path *p; + + PA_HASHMAP_FOREACH(p, ps->paths, state2) { + pa_alsa_element *e; + + PA_LLIST_FOREACH(e, p->elements) { + if (e->volume_use != PA_ALSA_VOLUME_IGNORE && pa_streq(db_fix->name, e->alsa_id.name) && + db_fix->index == e->alsa_id.index) { + /* The profile set that contains the dB fix may be freed + * before the element, so we have to copy the dB fix + * object. */ + e->db_fix = pa_xnewdup(pa_alsa_decibel_fix, db_fix, 1); + e->db_fix->profile_set = NULL; + e->db_fix->name = pa_xstrdup(db_fix->name); + e->db_fix->db_values = pa_xmemdup(db_fix->db_values, (db_fix->max_step - db_fix->min_step + 1) * sizeof(long)); + } + } + } + } + + return ps; + +fail: + if (ps) + pa_alsa_path_set_free(ps); + + return NULL; +} + +void pa_alsa_path_set_dump(pa_alsa_path_set *ps) { + pa_alsa_path *p; + void *state; + pa_assert(ps); + + pa_log_debug("Path Set %p, direction=%i", + (void*) ps, + ps->direction); + + PA_HASHMAP_FOREACH(p, ps->paths, state) + pa_alsa_path_dump(p); +} + +static bool options_have_option(pa_alsa_option *options, const char *alsa_name) { + pa_alsa_option *o; + + pa_assert(options); + pa_assert(alsa_name); + + PA_LLIST_FOREACH(o, options) { + if (pa_streq(o->alsa_name, alsa_name)) + return true; + } + return false; +} + +static bool enumeration_is_subset(pa_alsa_option *a_options, pa_alsa_option *b_options) { + pa_alsa_option *oa, *ob; + + if (!a_options) return true; + if (!b_options) return false; + + /* If there is an option A offers that B does not, then A is not a subset of B. */ + PA_LLIST_FOREACH(oa, a_options) { + bool found = false; + PA_LLIST_FOREACH(ob, b_options) { + if (pa_streq(oa->alsa_name, ob->alsa_name)) { + found = true; + break; + } + } + if (!found) + return false; + } + return true; +} + +/** + * Compares two elements to see if a is a subset of b + */ +static bool element_is_subset(pa_alsa_element *a, pa_alsa_element *b, snd_mixer_t *m) { + char buf[64]; + + pa_assert(a); + pa_assert(b); + pa_assert(m); + + /* General rules: + * Every state is a subset of itself (with caveats for volume_limits and options) + * IGNORE is a subset of every other state */ + + /* Check the volume_use */ + if (a->volume_use != PA_ALSA_VOLUME_IGNORE) { + + /* "Constant" is subset of "Constant" only when their constant values are equal */ + if (a->volume_use == PA_ALSA_VOLUME_CONSTANT && b->volume_use == PA_ALSA_VOLUME_CONSTANT && a->constant_volume != b->constant_volume) + return false; + + /* Different volume uses when b is not "Merge" means we are definitely not a subset */ + if (a->volume_use != b->volume_use && b->volume_use != PA_ALSA_VOLUME_MERGE) + return false; + + /* "Constant" is a subset of "Merge", if there is not a "volume-limit" in "Merge" below the actual constant. + * "Zero" and "Off" are just special cases of "Constant" when comparing to "Merge" + * "Merge" with a "volume-limit" is a subset of "Merge" without a "volume-limit" or with a higher "volume-limit" */ + if (b->volume_use == PA_ALSA_VOLUME_MERGE && b->volume_limit >= 0) { + long a_limit; + + if (a->volume_use == PA_ALSA_VOLUME_CONSTANT) + a_limit = a->constant_volume; + else if (a->volume_use == PA_ALSA_VOLUME_ZERO) { + long dB = 0; + + if (a->db_fix) { + int rounding = (a->direction == PA_ALSA_DIRECTION_OUTPUT ? +1 : -1); + a_limit = decibel_fix_get_step(a->db_fix, &dB, rounding); + } else { + snd_mixer_selem_id_t *sid; + snd_mixer_elem_t *me; + + SELEM_INIT(sid, &a->alsa_id); + if (!(me = snd_mixer_find_selem(m, sid))) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &a->alsa_id); + pa_log_warn("Element %s seems to have disappeared.", buf); + return false; + } + + if (a->direction == PA_ALSA_DIRECTION_OUTPUT) { + if (snd_mixer_selem_ask_playback_dB_vol(me, dB, +1, &a_limit) < 0) + return false; + } else { + if (snd_mixer_selem_ask_capture_dB_vol(me, dB, -1, &a_limit) < 0) + return false; + } + } + } else if (a->volume_use == PA_ALSA_VOLUME_OFF) + a_limit = a->min_volume; + else if (a->volume_use == PA_ALSA_VOLUME_MERGE) + a_limit = a->volume_limit; + else + pa_assert_not_reached(); + + if (a_limit > b->volume_limit) + return false; + } + + if (a->volume_use == PA_ALSA_VOLUME_MERGE) { + int s; + /* If override-maps are different, they're not subsets */ + if (a->n_channels != b->n_channels) + return false; + for (s = 0; s <= SND_MIXER_SCHN_LAST; s++) + if (a->masks[s][a->n_channels-1] != b->masks[s][b->n_channels-1]) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &a->alsa_id); + pa_log_debug("Element %s is not a subset - mask a: 0x%" PRIx64 ", mask b: 0x%" PRIx64 ", at channel %d", + buf, a->masks[s][a->n_channels-1], b->masks[s][b->n_channels-1], s); + return false; + } + } + } + + if (a->switch_use != PA_ALSA_SWITCH_IGNORE) { + /* "On" is a subset of "Mute". + * "Off" is a subset of "Mute". + * "On" is a subset of "Select", if there is an "Option:On" in B. + * "Off" is a subset of "Select", if there is an "Option:Off" in B. + * "Select" is a subset of "Select", if they have the same options (is this always true?). */ + + if (a->switch_use != b->switch_use) { + + if (a->switch_use == PA_ALSA_SWITCH_SELECT || a->switch_use == PA_ALSA_SWITCH_MUTE + || b->switch_use == PA_ALSA_SWITCH_OFF || b->switch_use == PA_ALSA_SWITCH_ON) + return false; + + if (b->switch_use == PA_ALSA_SWITCH_SELECT) { + if (a->switch_use == PA_ALSA_SWITCH_ON) { + if (!options_have_option(b->options, "on")) + return false; + } else if (a->switch_use == PA_ALSA_SWITCH_OFF) { + if (!options_have_option(b->options, "off")) + return false; + } + } + } else if (a->switch_use == PA_ALSA_SWITCH_SELECT) { + if (!enumeration_is_subset(a->options, b->options)) + return false; + } + } + + if (a->enumeration_use != PA_ALSA_ENUMERATION_IGNORE) { + if (b->enumeration_use == PA_ALSA_ENUMERATION_IGNORE) + return false; + if (!enumeration_is_subset(a->options, b->options)) + return false; + } + + return true; +} + +static void path_set_condense(pa_alsa_path_set *ps, snd_mixer_t *m) { + pa_alsa_path *p; + void *state; + + pa_assert(ps); + pa_assert(m); + + /* If we only have one path, then don't bother */ + if (pa_hashmap_size(ps->paths) < 2) + return; + + PA_HASHMAP_FOREACH(p, ps->paths, state) { + pa_alsa_path *p2; + void *state2; + + PA_HASHMAP_FOREACH(p2, ps->paths, state2) { + pa_alsa_element *ea, *eb; + pa_alsa_jack *ja, *jb; + bool is_subset = true; + + if (p == p2) + continue; + + /* If a has a jack that b does not have, a is not a subset */ + PA_LLIST_FOREACH(ja, p->jacks) { + bool exists = false; + + if (!ja->has_control) + continue; + + PA_LLIST_FOREACH(jb, p2->jacks) { + if (jb->has_control && pa_streq(ja->alsa_id.name, jb->alsa_id.name) && + (ja->alsa_id.index == jb->alsa_id.index) && + (ja->state_plugged == jb->state_plugged) && + (ja->state_unplugged == jb->state_unplugged)) { + exists = true; + break; + } + } + + if (!exists) { + is_subset = false; + break; + } + } + + /* Compare the elements of each set... */ + PA_LLIST_FOREACH(ea, p->elements) { + bool found_matching_element = false; + + if (!is_subset) + break; + + PA_LLIST_FOREACH(eb, p2->elements) { + if (pa_streq(ea->alsa_id.name, eb->alsa_id.name) && + ea->alsa_id.index == eb->alsa_id.index) { + found_matching_element = true; + is_subset = element_is_subset(ea, eb, m); + break; + } + } + + if (!found_matching_element) + is_subset = false; + } + + if (is_subset) { + pa_log_debug("Removing path '%s' as it is a subset of '%s'.", p->name, p2->name); + pa_hashmap_remove(ps->paths, p); + break; + } + } + } +} + +static pa_alsa_path* path_set_find_path_by_description(pa_alsa_path_set *ps, const char* description, pa_alsa_path *ignore) { + pa_alsa_path* p; + void *state; + + PA_HASHMAP_FOREACH(p, ps->paths, state) + if (p != ignore && pa_streq(p->description, description)) + return p; + + return NULL; +} + +static void path_set_make_path_descriptions_unique(pa_alsa_path_set *ps) { + pa_alsa_path *p, *q; + void *state, *state2; + + PA_HASHMAP_FOREACH(p, ps->paths, state) { + unsigned i; + char *old_description; + + q = path_set_find_path_by_description(ps, p->description, p); + + if (!q) + continue; + + old_description = pa_xstrdup(p->description); + + /* OK, this description is not unique, hence let's rename */ + i = 1; + PA_HASHMAP_FOREACH(q, ps->paths, state2) { + char *new_description; + + if (!pa_streq(q->description, old_description)) + continue; + + new_description = pa_sprintf_malloc("%s %u", q->description, i); + pa_xfree(q->description); + q->description = new_description; + + i++; + } + + pa_xfree(old_description); + } +} + +static void mapping_free(pa_alsa_mapping *m) { + pa_assert(m); + + pa_xfree(m->name); + pa_xfree(m->description); + pa_xfree(m->description_key); + + pa_proplist_free(m->proplist); + + pa_xstrfreev(m->device_strings); + pa_xstrfreev(m->input_path_names); + pa_xstrfreev(m->output_path_names); + pa_xstrfreev(m->input_element); + pa_xstrfreev(m->output_element); + if (m->input_path_set) + pa_alsa_path_set_free(m->input_path_set); + if (m->output_path_set) + pa_alsa_path_set_free(m->output_path_set); + + pa_assert(!m->input_pcm); + pa_assert(!m->output_pcm); + + pa_alsa_ucm_mapping_context_free(&m->ucm_context); + + pa_xfree(m); +} + +static void profile_free(pa_alsa_profile *p) { + pa_assert(p); + + pa_xfree(p->name); + pa_xfree(p->description); + pa_xfree(p->description_key); + pa_xfree(p->input_name); + pa_xfree(p->output_name); + + pa_xstrfreev(p->input_mapping_names); + pa_xstrfreev(p->output_mapping_names); + + if (p->input_mappings) + pa_idxset_free(p->input_mappings, NULL); + + if (p->output_mappings) + pa_idxset_free(p->output_mappings, NULL); + + pa_xfree(p); +} + +void pa_alsa_profile_set_free(pa_alsa_profile_set *ps) { + pa_assert(ps); + + if (ps->input_paths) + pa_hashmap_free(ps->input_paths); + + if (ps->output_paths) + pa_hashmap_free(ps->output_paths); + + if (ps->profiles) + pa_hashmap_free(ps->profiles); + + if (ps->mappings) + pa_hashmap_free(ps->mappings); + + if (ps->decibel_fixes) + pa_hashmap_free(ps->decibel_fixes); + + pa_xfree(ps); +} + +pa_alsa_mapping *pa_alsa_mapping_get(pa_alsa_profile_set *ps, const char *name) { + pa_alsa_mapping *m; + + if (!pa_startswith(name, "Mapping ")) + return NULL; + + name += 8; + + if ((m = pa_hashmap_get(ps->mappings, name))) + return m; + + m = pa_xnew0(pa_alsa_mapping, 1); + m->profile_set = ps; + m->exact_channels = true; + m->name = pa_xstrdup(name); + pa_sample_spec_init(&m->sample_spec); + pa_channel_map_init(&m->channel_map); + m->proplist = pa_proplist_new(); + m->hw_device_index = -1; + + pa_hashmap_put(ps->mappings, m->name, m); + + return m; +} + +static pa_alsa_profile *profile_get(pa_alsa_profile_set *ps, const char *name) { + pa_alsa_profile *p; + + if (!pa_startswith(name, "Profile ")) + return NULL; + + name += 8; + + if ((p = pa_hashmap_get(ps->profiles, name))) + return p; + + p = pa_xnew0(pa_alsa_profile, 1); + p->profile_set = ps; + p->name = pa_xstrdup(name); + + pa_hashmap_put(ps->profiles, p->name, p); + + return p; +} + +static pa_alsa_decibel_fix *decibel_fix_get(pa_alsa_profile_set *ps, const char *alsa_id) { + pa_alsa_decibel_fix *db_fix; + char *name; + int index; + + if (!pa_startswith(alsa_id, "DecibelFix ")) + return NULL; + + alsa_id += 11; + + if ((db_fix = pa_hashmap_get(ps->decibel_fixes, alsa_id))) + return db_fix; + + name = alloca(strlen(alsa_id) + 1); + if (alsa_id_decode(alsa_id, name, &index)) + return NULL; + + db_fix = pa_xnew0(pa_alsa_decibel_fix, 1); + db_fix->profile_set = ps; + db_fix->name = pa_xstrdup(name); + db_fix->index = index; + db_fix->key = pa_xstrdup(alsa_id); + + pa_hashmap_put(ps->decibel_fixes, db_fix->key, db_fix); + + return db_fix; +} + +static int mapping_parse_device_strings(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if (!(m = pa_alsa_mapping_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + pa_xstrfreev(m->device_strings); + if (!(m->device_strings = pa_split_spaces_strv(state->rvalue))) { + pa_log("[%s:%u] Device string list empty of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int mapping_parse_channel_map(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if (!(m = pa_alsa_mapping_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + if (!(pa_channel_map_parse(&m->channel_map, state->rvalue))) { + pa_log("[%s:%u] Channel map invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int mapping_parse_paths(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if (!(m = pa_alsa_mapping_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + if (pa_streq(state->lvalue, "paths-input")) { + pa_xstrfreev(m->input_path_names); + m->input_path_names = pa_split_spaces_strv(state->rvalue); + } else { + pa_xstrfreev(m->output_path_names); + m->output_path_names = pa_split_spaces_strv(state->rvalue); + } + + return 0; +} + +static int mapping_parse_exact_channels(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_mapping *m; + int b; + + pa_assert(state); + + ps = state->userdata; + + if (!(m = pa_alsa_mapping_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + if ((b = pa_parse_boolean(state->rvalue)) < 0) { + pa_log("[%s:%u] %s has invalid value '%s'", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + m->exact_channels = b; + + return 0; +} + +static int mapping_parse_element(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if (!(m = pa_alsa_mapping_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + if (pa_streq(state->lvalue, "element-input")) { + pa_xstrfreev(m->input_element); + m->input_element = pa_split_spaces_strv(state->rvalue); + } else { + pa_xstrfreev(m->output_element); + m->output_element = pa_split_spaces_strv(state->rvalue); + } + + return 0; +} + +static int mapping_parse_direction(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if (!(m = pa_alsa_mapping_get(ps, state->section))) { + pa_log("[%s:%u] Section name %s invalid.", state->filename, state->lineno, state->section); + return -1; + } + + if (pa_streq(state->rvalue, "input")) + m->direction = PA_ALSA_DIRECTION_INPUT; + else if (pa_streq(state->rvalue, "output")) + m->direction = PA_ALSA_DIRECTION_OUTPUT; + else if (pa_streq(state->rvalue, "any")) + m->direction = PA_ALSA_DIRECTION_ANY; + else { + pa_log("[%s:%u] Direction %s invalid.", state->filename, state->lineno, state->rvalue); + return -1; + } + + return 0; +} + +static int mapping_parse_description(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_profile *p; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if ((m = pa_alsa_mapping_get(ps, state->section))) { + pa_xfree(m->description); + m->description = pa_xstrdup(state->rvalue); + } else if ((p = profile_get(ps, state->section))) { + pa_xfree(p->description); + p->description = pa_xstrdup(state->rvalue); + } else { + pa_log("[%s:%u] Section name %s invalid.", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int mapping_parse_description_key(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_profile *p; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if ((m = pa_alsa_mapping_get(ps, state->section))) { + pa_xfree(m->description_key); + m->description_key = pa_xstrdup(state->rvalue); + } else if ((p = profile_get(ps, state->section))) { + pa_xfree(p->description_key); + p->description_key = pa_xstrdup(state->rvalue); + } else { + pa_log("[%s:%u] Section name %s invalid.", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + + +static int mapping_parse_priority(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_profile *p; + pa_alsa_mapping *m; + uint32_t prio; + + pa_assert(state); + + ps = state->userdata; + + if (pa_atou(state->rvalue, &prio) < 0) { + pa_log("[%s:%u] Priority invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if ((m = pa_alsa_mapping_get(ps, state->section))) + m->priority = prio; + else if ((p = profile_get(ps, state->section))) + p->priority = prio; + else { + pa_log("[%s:%u] Section name %s invalid.", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int mapping_parse_fallback(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_profile *p; + pa_alsa_mapping *m; + int k; + + pa_assert(state); + + ps = state->userdata; + + if ((k = pa_parse_boolean(state->rvalue)) < 0) { + pa_log("[%s:%u] Fallback invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + if ((m = pa_alsa_mapping_get(ps, state->section))) + m->fallback = k; + else if ((p = profile_get(ps, state->section))) + p->fallback_input = p->fallback_output = k; + else { + pa_log("[%s:%u] Section name %s invalid.", state->filename, state->lineno, state->section); + return -1; + } + + return 0; +} + +static int mapping_parse_intended_roles(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_mapping *m; + + pa_assert(state); + + ps = state->userdata; + + if (!(m = pa_alsa_mapping_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + pa_proplist_sets(m->proplist, PA_PROP_DEVICE_INTENDED_ROLES, state->rvalue); + + return 0; +} + + +static int profile_parse_mappings(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_profile *p; + + pa_assert(state); + + ps = state->userdata; + + if (!(p = profile_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + if (pa_streq(state->lvalue, "input-mappings")) { + pa_xstrfreev(p->input_mapping_names); + p->input_mapping_names = pa_split_spaces_strv(state->rvalue); + } else { + pa_xstrfreev(p->output_mapping_names); + p->output_mapping_names = pa_split_spaces_strv(state->rvalue); + } + + return 0; +} + +static int profile_parse_skip_probe(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_profile *p; + int b; + + pa_assert(state); + + ps = state->userdata; + + if (!(p = profile_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + if ((b = pa_parse_boolean(state->rvalue)) < 0) { + pa_log("[%s:%u] Skip probe invalid of '%s'", state->filename, state->lineno, state->section); + return -1; + } + + p->supported = b; + + return 0; +} + +static int decibel_fix_parse_db_values(pa_config_parser_state *state) { + pa_alsa_profile_set *ps; + pa_alsa_decibel_fix *db_fix; + char **items; + char *item; + long *db_values; + unsigned n = 8; /* Current size of the db_values table. */ + unsigned min_step = 0; + unsigned max_step = 0; + unsigned i = 0; /* Index to the items table. */ + unsigned prev_step = 0; + double prev_db = 0; + + pa_assert(state); + + ps = state->userdata; + + if (!(db_fix = decibel_fix_get(ps, state->section))) { + pa_log("[%s:%u] %s invalid in section %s", state->filename, state->lineno, state->lvalue, state->section); + return -1; + } + + if (!(items = pa_split_spaces_strv(state->rvalue))) { + pa_log("[%s:%u] Value missing", state->filename, state->lineno); + return -1; + } + + db_values = pa_xnew(long, n); + + while ((item = items[i++])) { + char *s = item; /* Step value string. */ + char *d = item; /* dB value string. */ + uint32_t step; + double db; + + /* Move d forward until it points to a colon or to the end of the item. */ + for (; *d && *d != ':'; ++d); + + if (d == s) { + /* item started with colon. */ + pa_log("[%s:%u] No step value found in %s", state->filename, state->lineno, item); + goto fail; + } + + if (!*d || !*(d + 1)) { + /* No colon found, or it was the last character in item. */ + pa_log("[%s:%u] No dB value found in %s", state->filename, state->lineno, item); + goto fail; + } + + /* pa_atou() needs a null-terminating string. Let's replace the colon + * with a zero byte. */ + *d++ = '\0'; + + if (pa_atou(s, &step) < 0) { + pa_log("[%s:%u] Invalid step value: %s", state->filename, state->lineno, s); + goto fail; + } + + if (pa_atod(d, &db) < 0) { + pa_log("[%s:%u] Invalid dB value: %s", state->filename, state->lineno, d); + goto fail; + } + + if (step <= prev_step && i != 1) { + pa_log("[%s:%u] Step value %u not greater than the previous value %u", state->filename, state->lineno, step, prev_step); + goto fail; + } + + if (db < prev_db && i != 1) { + pa_log("[%s:%u] Decibel value %0.2f less than the previous value %0.2f", state->filename, state->lineno, db, prev_db); + goto fail; + } + + if (i == 1) { + min_step = step; + db_values[0] = (long) (db * 100.0); + prev_step = step; + prev_db = db; + } else { + /* Interpolate linearly. */ + double db_increment = (db - prev_db) / (step - prev_step); + + for (; prev_step < step; ++prev_step, prev_db += db_increment) { + + /* Reallocate the db_values table if it's about to overflow. */ + if (prev_step + 1 - min_step == n) { + n *= 2; + db_values = pa_xrenew(long, db_values, n); + } + + db_values[prev_step + 1 - min_step] = (long) ((prev_db + db_increment) * 100.0); + } + } + + max_step = step; + } + + db_fix->min_step = min_step; + db_fix->max_step = max_step; + pa_xfree(db_fix->db_values); + db_fix->db_values = db_values; + + pa_xstrfreev(items); + + return 0; + +fail: + pa_xstrfreev(items); + pa_xfree(db_values); + + return -1; +} + +/* the logic is simple: if we see the jack in multiple paths */ +/* assign all those paths to one availability_group */ +static void profile_set_set_availability_groups(pa_alsa_profile_set *ps) { + pa_dynarray *paths; + pa_alsa_path *p; + void *state; + unsigned idx1; + uint32_t num = 1; + + /* Merge ps->input_paths and ps->output_paths into one dynarray. */ + paths = pa_dynarray_new(NULL); + PA_HASHMAP_FOREACH(p, ps->input_paths, state) + pa_dynarray_append(paths, p); + PA_HASHMAP_FOREACH(p, ps->output_paths, state) + pa_dynarray_append(paths, p); + + PA_DYNARRAY_FOREACH(p, paths, idx1) { + pa_alsa_jack *j; + const char *found = NULL; + bool has_control = false; + + PA_LLIST_FOREACH(j, p->jacks) { + pa_alsa_path *p2; + unsigned idx2; + + if (!j->has_control || j->state_plugged == PA_AVAILABLE_NO) + continue; + has_control = true; + PA_DYNARRAY_FOREACH(p2, paths, idx2) { + pa_alsa_jack *j2; + + if (p2 == p) + break; + PA_LLIST_FOREACH(j2, p2->jacks) { + if (!j2->has_control || j2->state_plugged == PA_AVAILABLE_NO) + continue; + if (pa_streq(j->alsa_id.name, j2->alsa_id.name) && + j->alsa_id.index == j2->alsa_id.index) { + j->state_plugged = PA_AVAILABLE_UNKNOWN; + j2->state_plugged = PA_AVAILABLE_UNKNOWN; + found = p2->availability_group; + break; + } + } + } + if (found) + break; + } + if (!has_control) + continue; + if (!found) { + p->availability_group = pa_sprintf_malloc("Legacy %d", num); + } else { + p->availability_group = pa_xstrdup(found); + } + if (!found) + num++; + } + + pa_dynarray_free(paths); +} + +static void mapping_paths_probe(pa_alsa_mapping *m, pa_alsa_profile *profile, + pa_alsa_direction_t direction, pa_hashmap *used_paths, + pa_hashmap *mixers) { + + pa_alsa_path *p; + void *state; + snd_pcm_t *pcm_handle; + pa_alsa_path_set *ps; + snd_mixer_t *mixer_handle; + + if (direction == PA_ALSA_DIRECTION_OUTPUT) { + if (m->output_path_set) + return; /* Already probed */ + m->output_path_set = ps = pa_alsa_path_set_new(m, direction, NULL); /* FIXME: Handle paths_dir */ + pcm_handle = m->output_pcm; + } else { + if (m->input_path_set) + return; /* Already probed */ + m->input_path_set = ps = pa_alsa_path_set_new(m, direction, NULL); /* FIXME: Handle paths_dir */ + pcm_handle = m->input_pcm; + } + + if (!ps) + return; /* No paths */ + + pa_assert(pcm_handle); + + mixer_handle = pa_alsa_open_mixer_for_pcm(mixers, pcm_handle, true); + if (!mixer_handle) { + /* Cannot open mixer, remove all entries */ + pa_hashmap_remove_all(ps->paths); + return; + } + + PA_HASHMAP_FOREACH(p, ps->paths, state) { + if (p->autodetect_eld_device) + p->eld_device = m->hw_device_index; + + if (pa_alsa_path_probe(p, m, mixer_handle, m->profile_set->ignore_dB) < 0) + pa_hashmap_remove(ps->paths, p); + } + + path_set_condense(ps, mixer_handle); + path_set_make_path_descriptions_unique(ps); + + PA_HASHMAP_FOREACH(p, ps->paths, state) + pa_hashmap_put(used_paths, p, p); + + pa_log_debug("Available mixer paths (after tidying):"); + pa_alsa_path_set_dump(ps); +} + +static int mapping_verify(pa_alsa_mapping *m, const pa_channel_map *bonus) { + + static const struct description_map well_known_descriptions[] = { + { "analog-mono", N_("Analog Mono") }, + { "analog-stereo", N_("Analog Stereo") }, + { "mono-fallback", N_("Mono") }, + { "stereo-fallback", N_("Stereo") }, + /* Note: Not translated to "Analog Stereo Input", because the source + * name gets "Input" appended to it automatically, so adding "Input" + * here would lead to the source name to become "Analog Stereo Input + * Input". The same logic applies to analog-stereo-output, + * multichannel-input and multichannel-output. */ + { "analog-stereo-input", N_("Analog Stereo") }, + { "analog-stereo-output", N_("Analog Stereo") }, + { "multichannel-input", N_("Multichannel") }, + { "multichannel-output", N_("Multichannel") }, + { "analog-surround-21", N_("Analog Surround 2.1") }, + { "analog-surround-30", N_("Analog Surround 3.0") }, + { "analog-surround-31", N_("Analog Surround 3.1") }, + { "analog-surround-40", N_("Analog Surround 4.0") }, + { "analog-surround-41", N_("Analog Surround 4.1") }, + { "analog-surround-50", N_("Analog Surround 5.0") }, + { "analog-surround-51", N_("Analog Surround 5.1") }, + { "analog-surround-61", N_("Analog Surround 6.0") }, + { "analog-surround-61", N_("Analog Surround 6.1") }, + { "analog-surround-70", N_("Analog Surround 7.0") }, + { "analog-surround-71", N_("Analog Surround 7.1") }, + { "iec958-stereo", N_("Digital Stereo (IEC958)") }, + { "iec958-ac3-surround-40", N_("Digital Surround 4.0 (IEC958/AC3)") }, + { "iec958-ac3-surround-51", N_("Digital Surround 5.1 (IEC958/AC3)") }, + { "iec958-dts-surround-51", N_("Digital Surround 5.1 (IEC958/DTS)") }, + { "hdmi-stereo", N_("Digital Stereo (HDMI)") }, + { "hdmi-surround-51", N_("Digital Surround 5.1 (HDMI)") }, + { "gaming-headset-chat", N_("Chat") }, + { "gaming-headset-game", N_("Game") }, + }; + const char *description_key = m->description_key ? m->description_key : m->name; + + pa_assert(m); + + if (!pa_channel_map_valid(&m->channel_map)) { + pa_log("Mapping %s is missing channel map.", m->name); + return -1; + } + + if (!m->device_strings) { + pa_log("Mapping %s is missing device strings.", m->name); + return -1; + } + + if ((m->input_path_names && m->input_element) || + (m->output_path_names && m->output_element)) { + pa_log("Mapping %s must have either mixer path or mixer element, not both.", m->name); + return -1; + } + + if (!m->description) + m->description = pa_xstrdup(lookup_description(description_key, + well_known_descriptions, + PA_ELEMENTSOF(well_known_descriptions))); + + if (!m->description) + m->description = pa_xstrdup(m->name); + + if (bonus) { + if (pa_channel_map_equal(&m->channel_map, bonus)) + m->priority += 50; + else if (m->channel_map.channels == bonus->channels) + m->priority += 30; + } + + return 0; +} + +void pa_alsa_mapping_dump(pa_alsa_mapping *m) { + char cm[PA_CHANNEL_MAP_SNPRINT_MAX]; + + pa_assert(m); + + pa_log_debug("Mapping %s (%s), priority=%u, channel_map=%s, supported=%s, direction=%i", + m->name, + pa_strnull(m->description), + m->priority, + pa_channel_map_snprint(cm, sizeof(cm), &m->channel_map), + pa_yes_no(m->supported), + m->direction); +} + +static void profile_set_add_auto_pair( + pa_alsa_profile_set *ps, + pa_alsa_mapping *m, /* output */ + pa_alsa_mapping *n /* input */) { + + char *name; + pa_alsa_profile *p; + + pa_assert(ps); + pa_assert(m || n); + + if (m && m->direction == PA_ALSA_DIRECTION_INPUT) + return; + + if (n && n->direction == PA_ALSA_DIRECTION_OUTPUT) + return; + + if (m && n) + name = pa_sprintf_malloc("output:%s+input:%s", m->name, n->name); + else if (m) + name = pa_sprintf_malloc("output:%s", m->name); + else + name = pa_sprintf_malloc("input:%s", n->name); + + if (pa_hashmap_get(ps->profiles, name)) { + pa_xfree(name); + return; + } + + p = pa_xnew0(pa_alsa_profile, 1); + p->profile_set = ps; + p->name = name; + + if (m) { + p->output_name = pa_xstrdup(m->name); + p->output_mappings = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + pa_idxset_put(p->output_mappings, m, NULL); + p->priority += m->priority * 100; + p->fallback_output = m->fallback; + } + + if (n) { + p->input_name = pa_xstrdup(n->name); + p->input_mappings = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + pa_idxset_put(p->input_mappings, n, NULL); + p->priority += n->priority; + p->fallback_input = n->fallback; + } + + pa_hashmap_put(ps->profiles, p->name, p); +} + +static void profile_set_add_auto(pa_alsa_profile_set *ps) { + pa_alsa_mapping *m, *n; + void *m_state, *n_state; + + pa_assert(ps); + + /* The order is important here: + 1) try single inputs and outputs before trying their + combination, because if the half-duplex test failed, we don't have + to try full duplex. + 2) try the output right before the input combinations with + that output, because then the output_pcm is not closed between tests. + */ + PA_HASHMAP_FOREACH(n, ps->mappings, n_state) + profile_set_add_auto_pair(ps, NULL, n); + + PA_HASHMAP_FOREACH(m, ps->mappings, m_state) { + profile_set_add_auto_pair(ps, m, NULL); + + PA_HASHMAP_FOREACH(n, ps->mappings, n_state) + profile_set_add_auto_pair(ps, m, n); + } + +} + +static int profile_verify(pa_alsa_profile *p) { + + static const struct description_map well_known_descriptions[] = { + { "output:analog-mono+input:analog-mono", N_("Analog Mono Duplex") }, + { "output:analog-stereo+input:analog-stereo", N_("Analog Stereo Duplex") }, + { "output:iec958-stereo+input:iec958-stereo", N_("Digital Stereo Duplex (IEC958)") }, + { "output:multichannel-output+input:multichannel-input", N_("Multichannel Duplex") }, + { "output:unknown-stereo+input:unknown-stereo", N_("Stereo Duplex") }, + { "off", N_("Off") } + }; + const char *description_key = p->description_key ? p->description_key : p->name; + + pa_assert(p); + + /* Replace the output mapping names by the actual mappings */ + if (p->output_mapping_names) { + char **name; + + pa_assert(!p->output_mappings); + p->output_mappings = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + for (name = p->output_mapping_names; *name; name++) { + pa_alsa_mapping *m; + char **in; + bool duplicate = false; + + for (in = name + 1; *in; in++) + if (pa_streq(*name, *in)) { + duplicate = true; + break; + } + + if (duplicate) + continue; + + if (!(m = pa_hashmap_get(p->profile_set->mappings, *name)) || m->direction == PA_ALSA_DIRECTION_INPUT) { + pa_log("Profile '%s' refers to nonexistent mapping '%s'.", p->name, *name); + return -1; + } + + pa_idxset_put(p->output_mappings, m, NULL); + + if (p->supported) + m->supported++; + } + + pa_xstrfreev(p->output_mapping_names); + p->output_mapping_names = NULL; + } + + /* Replace the input mapping names by the actual mappings */ + if (p->input_mapping_names) { + char **name; + + pa_assert(!p->input_mappings); + p->input_mappings = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + for (name = p->input_mapping_names; *name; name++) { + pa_alsa_mapping *m; + char **in; + bool duplicate = false; + + for (in = name + 1; *in; in++) + if (pa_streq(*name, *in)) { + duplicate = true; + break; + } + + if (duplicate) + continue; + + if (!(m = pa_hashmap_get(p->profile_set->mappings, *name)) || m->direction == PA_ALSA_DIRECTION_OUTPUT) { + pa_log("Profile '%s' refers to nonexistent mapping '%s'.", p->name, *name); + return -1; + } + + pa_idxset_put(p->input_mappings, m, NULL); + + if (p->supported) + m->supported++; + } + + pa_xstrfreev(p->input_mapping_names); + p->input_mapping_names = NULL; + } + + if (!p->input_mappings && !p->output_mappings) { + pa_log("Profile '%s' lacks mappings.", p->name); + return -1; + } + + if (!p->description) + p->description = pa_xstrdup(lookup_description(description_key, + well_known_descriptions, + PA_ELEMENTSOF(well_known_descriptions))); + + if (!p->description) { + pa_strbuf *sb; + uint32_t idx; + pa_alsa_mapping *m; + + sb = pa_strbuf_new(); + + if (p->output_mappings) + PA_IDXSET_FOREACH(m, p->output_mappings, idx) { + if (!pa_strbuf_isempty(sb)) + pa_strbuf_puts(sb, " + "); + + pa_strbuf_printf(sb, _("%s Output"), m->description); + } + + if (p->input_mappings) + PA_IDXSET_FOREACH(m, p->input_mappings, idx) { + if (!pa_strbuf_isempty(sb)) + pa_strbuf_puts(sb, " + "); + + pa_strbuf_printf(sb, _("%s Input"), m->description); + } + + p->description = pa_strbuf_to_string_free(sb); + } + + return 0; +} + +void pa_alsa_profile_dump(pa_alsa_profile *p) { + uint32_t idx; + pa_alsa_mapping *m; + pa_assert(p); + + pa_log_debug("Profile %s (%s), input=%s, output=%s priority=%u, supported=%s n_input_mappings=%u, n_output_mappings=%u", + p->name, + pa_strnull(p->description), + pa_strnull(p->input_name), + pa_strnull(p->output_name), + p->priority, + pa_yes_no(p->supported), + p->input_mappings ? pa_idxset_size(p->input_mappings) : 0, + p->output_mappings ? pa_idxset_size(p->output_mappings) : 0); + + if (p->input_mappings) + PA_IDXSET_FOREACH(m, p->input_mappings, idx) + pa_log_debug("Input %s", m->name); + + if (p->output_mappings) + PA_IDXSET_FOREACH(m, p->output_mappings, idx) + pa_log_debug("Output %s", m->name); +} + +static int decibel_fix_verify(pa_alsa_decibel_fix *db_fix) { + pa_assert(db_fix); + + /* Check that the dB mapping has been configured. Since "db-values" is + * currently the only option in the DecibelFix section, and decibel fix + * objects don't get created if a DecibelFix section is empty, this is + * actually a redundant check. Having this may prevent future bugs, + * however. */ + if (!db_fix->db_values) { + pa_log("Decibel fix for element %s lacks the dB values.", db_fix->name); + return -1; + } + + return 0; +} + +void pa_alsa_decibel_fix_dump(pa_alsa_decibel_fix *db_fix) { + char *db_values = NULL; + + pa_assert(db_fix); + + if (db_fix->db_values) { + pa_strbuf *buf; + unsigned long i, nsteps; + + pa_assert(db_fix->min_step <= db_fix->max_step); + nsteps = db_fix->max_step - db_fix->min_step + 1; + + buf = pa_strbuf_new(); + for (i = 0; i < nsteps; ++i) + pa_strbuf_printf(buf, "[%li]:%0.2f ", i + db_fix->min_step, db_fix->db_values[i] / 100.0); + + db_values = pa_strbuf_to_string_free(buf); + } + + pa_log_debug("Decibel fix %s, min_step=%li, max_step=%li, db_values=%s", + db_fix->name, db_fix->min_step, db_fix->max_step, pa_strnull(db_values)); + + pa_xfree(db_values); +} + +pa_alsa_profile_set* pa_alsa_profile_set_new(const char *fname, const pa_channel_map *bonus) { + pa_alsa_profile_set *ps; + pa_alsa_profile *p; + pa_alsa_mapping *m; + pa_alsa_decibel_fix *db_fix; + char *fn; + int r; + void *state; + + static pa_config_item items[] = { + /* [General] */ + { "auto-profiles", pa_config_parse_bool, NULL, "General" }, + + /* [Mapping ...] */ + { "device-strings", mapping_parse_device_strings, NULL, NULL }, + { "channel-map", mapping_parse_channel_map, NULL, NULL }, + { "paths-input", mapping_parse_paths, NULL, NULL }, + { "paths-output", mapping_parse_paths, NULL, NULL }, + { "element-input", mapping_parse_element, NULL, NULL }, + { "element-output", mapping_parse_element, NULL, NULL }, + { "direction", mapping_parse_direction, NULL, NULL }, + { "exact-channels", mapping_parse_exact_channels, NULL, NULL }, + { "intended-roles", mapping_parse_intended_roles, NULL, NULL }, + + /* Shared by [Mapping ...] and [Profile ...] */ + { "description", mapping_parse_description, NULL, NULL }, + { "description-key", mapping_parse_description_key,NULL, NULL }, + { "priority", mapping_parse_priority, NULL, NULL }, + { "fallback", mapping_parse_fallback, NULL, NULL }, + + /* [Profile ...] */ + { "input-mappings", profile_parse_mappings, NULL, NULL }, + { "output-mappings", profile_parse_mappings, NULL, NULL }, + { "skip-probe", profile_parse_skip_probe, NULL, NULL }, + + /* [DecibelFix ...] */ + { "db-values", decibel_fix_parse_db_values, NULL, NULL }, + { NULL, NULL, NULL, NULL } + }; + + ps = pa_xnew0(pa_alsa_profile_set, 1); + ps->mappings = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, NULL, (pa_free_cb_t) mapping_free); + ps->profiles = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, NULL, (pa_free_cb_t) profile_free); + ps->decibel_fixes = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, NULL, (pa_free_cb_t) decibel_fix_free); + ps->input_paths = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, NULL, (pa_free_cb_t) pa_alsa_path_free); + ps->output_paths = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, NULL, (pa_free_cb_t) pa_alsa_path_free); + + items[0].data = &ps->auto_profiles; + + if (!fname) + fname = "default.conf"; + + fn = pa_maybe_prefix_path(fname, +#ifdef HAVE_RUNNING_FROM_BUILD_TREE + pa_run_from_build_tree() ? PA_SRCDIR "/modules/alsa/mixer/profile-sets/" : +#endif + PA_ALSA_PROFILE_SETS_DIR); + + r = pa_config_parse(fn, NULL, items, NULL, false, ps); + pa_xfree(fn); + + if (r < 0) + goto fail; + + PA_HASHMAP_FOREACH(m, ps->mappings, state) + if (mapping_verify(m, bonus) < 0) + goto fail; + + if (ps->auto_profiles) + profile_set_add_auto(ps); + + PA_HASHMAP_FOREACH(p, ps->profiles, state) + if (profile_verify(p) < 0) + goto fail; + + PA_HASHMAP_FOREACH(db_fix, ps->decibel_fixes, state) + if (decibel_fix_verify(db_fix) < 0) + goto fail; + + return ps; + +fail: + pa_alsa_profile_set_free(ps); + return NULL; +} + +static void profile_finalize_probing(pa_alsa_profile *to_be_finalized, pa_alsa_profile *next) { + pa_alsa_mapping *m; + uint32_t idx; + + if (!to_be_finalized) + return; + + if (to_be_finalized->output_mappings) + PA_IDXSET_FOREACH(m, to_be_finalized->output_mappings, idx) { + + if (!m->output_pcm) + continue; + + if (to_be_finalized->supported) + m->supported++; + + /* If this mapping is also in the next profile, we won't close the + * pcm handle here, because it would get immediately reopened + * anyway. */ + if (next && next->output_mappings && pa_idxset_get_by_data(next->output_mappings, m, NULL)) + continue; + + snd_pcm_close(m->output_pcm); + m->output_pcm = NULL; + } + + if (to_be_finalized->input_mappings) + PA_IDXSET_FOREACH(m, to_be_finalized->input_mappings, idx) { + + if (!m->input_pcm) + continue; + + if (to_be_finalized->supported) + m->supported++; + + /* If this mapping is also in the next profile, we won't close the + * pcm handle here, because it would get immediately reopened + * anyway. */ + if (next && next->input_mappings && pa_idxset_get_by_data(next->input_mappings, m, NULL)) + continue; + + snd_pcm_close(m->input_pcm); + m->input_pcm = NULL; + } +} + +static snd_pcm_t* mapping_open_pcm(pa_alsa_mapping *m, + const pa_sample_spec *ss, + const char *dev_id, + bool exact_channels, + int mode, + unsigned default_n_fragments, + unsigned default_fragment_size_msec) { + + snd_pcm_t* handle; + pa_sample_spec try_ss = *ss; + pa_channel_map try_map = m->channel_map; + snd_pcm_uframes_t try_period_size, try_buffer_size; + + try_ss.channels = try_map.channels; + + try_period_size = + pa_usec_to_bytes(default_fragment_size_msec * PA_USEC_PER_MSEC, &try_ss) / + pa_frame_size(&try_ss); + try_buffer_size = default_n_fragments * try_period_size; + + handle = pa_alsa_open_by_template( + m->device_strings, dev_id, NULL, &try_ss, + &try_map, mode, &try_period_size, + &try_buffer_size, 0, NULL, NULL, exact_channels); + if (handle && !exact_channels && m->channel_map.channels != try_map.channels) { + char buf[PA_CHANNEL_MAP_SNPRINT_MAX]; + pa_log_debug("Channel map for mapping '%s' permanently changed to '%s'", m->name, + pa_channel_map_snprint(buf, sizeof(buf), &try_map)); + m->channel_map = try_map; + } + return handle; +} + +static void paths_drop_unused(pa_hashmap* h, pa_hashmap *keep) { + + void* state = NULL; + const void* key; + pa_alsa_path* p; + + pa_assert(h); + pa_assert(keep); + + p = pa_hashmap_iterate(h, &state, &key); + while (p) { + if (pa_hashmap_get(keep, p) == NULL) + pa_hashmap_remove_and_free(h, key); + p = pa_hashmap_iterate(h, &state, &key); + } +} + +static int add_profiles_to_probe( + pa_alsa_profile **list, + pa_hashmap *profiles, + bool fallback_output, + bool fallback_input) { + + int i = 0; + void *state; + pa_alsa_profile *p; + PA_HASHMAP_FOREACH(p, profiles, state) + if (p->fallback_input == fallback_input && p->fallback_output == fallback_output) { + *list = p; + list++; + i++; + } + return i; +} + +static void mapping_query_hw_device(pa_alsa_mapping *mapping, snd_pcm_t *pcm) { + int r; + snd_pcm_info_t* pcm_info; + snd_pcm_info_alloca(&pcm_info); + + r = snd_pcm_info(pcm, pcm_info); + if (r < 0) { + pa_log("Mapping %s: snd_pcm_info() failed %s: ", mapping->name, pa_alsa_strerror(r)); + return; + } + + /* XXX: It's not clear what snd_pcm_info_get_device() does if the device is + * not backed by a hw device or if it's backed by multiple hw devices. We + * only use hw_device_index for HDMI devices, however, and for those the + * return value is expected to be always valid, so this shouldn't be a + * significant problem. */ + mapping->hw_device_index = snd_pcm_info_get_device(pcm_info); +} + +void pa_alsa_profile_set_probe( + pa_alsa_profile_set *ps, + pa_hashmap *mixers, + const char *dev_id, + const pa_sample_spec *ss, + unsigned default_n_fragments, + unsigned default_fragment_size_msec) { + + bool found_output = false, found_input = false; + + pa_alsa_profile *p, *last = NULL; + pa_alsa_profile **pp, **probe_order; + pa_alsa_mapping *m; + pa_hashmap *broken_inputs, *broken_outputs, *used_paths; + + pa_assert(ps); + pa_assert(dev_id); + pa_assert(ss); + + if (ps->probed) + return; + + broken_inputs = pa_hashmap_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + broken_outputs = pa_hashmap_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + used_paths = pa_hashmap_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + pp = probe_order = pa_xnew0(pa_alsa_profile *, pa_hashmap_size(ps->profiles) + 1); + + pp += add_profiles_to_probe(pp, ps->profiles, false, false); + pp += add_profiles_to_probe(pp, ps->profiles, false, true); + pp += add_profiles_to_probe(pp, ps->profiles, true, false); + pp += add_profiles_to_probe(pp, ps->profiles, true, true); + + for (pp = probe_order; *pp; pp++) { + uint32_t idx; + p = *pp; + + /* Skip if fallback and already found something */ + if (found_input && p->fallback_input) + continue; + if (found_output && p->fallback_output) + continue; + + /* Skip if this is already marked that it is supported (i.e. from the config file) */ + if (!p->supported) { + + profile_finalize_probing(last, p); + p->supported = true; + + if (p->output_mappings) { + PA_IDXSET_FOREACH(m, p->output_mappings, idx) { + if (pa_hashmap_get(broken_outputs, m) == m) { + pa_log_debug("Skipping profile %s - will not be able to open output:%s", p->name, m->name); + p->supported = false; + break; + } + } + } + + if (p->input_mappings && p->supported) { + PA_IDXSET_FOREACH(m, p->input_mappings, idx) { + if (pa_hashmap_get(broken_inputs, m) == m) { + pa_log_debug("Skipping profile %s - will not be able to open input:%s", p->name, m->name); + p->supported = false; + break; + } + } + } + + if (p->supported) + pa_log_debug("Looking at profile %s", p->name); + + /* Check if we can open all new ones */ + if (p->output_mappings && p->supported) + PA_IDXSET_FOREACH(m, p->output_mappings, idx) { + + if (m->output_pcm) + continue; + + pa_log_debug("Checking for playback on %s (%s)", m->description, m->name); + if (!(m->output_pcm = mapping_open_pcm(m, ss, dev_id, m->exact_channels, + SND_PCM_STREAM_PLAYBACK, + default_n_fragments, + default_fragment_size_msec))) { + p->supported = false; + if (pa_idxset_size(p->output_mappings) == 1 && + ((!p->input_mappings) || pa_idxset_size(p->input_mappings) == 0)) { + pa_log_debug("Caching failure to open output:%s", m->name); + pa_hashmap_put(broken_outputs, m, m); + } + break; + } + + if (m->hw_device_index < 0) + mapping_query_hw_device(m, m->output_pcm); + } + + if (p->input_mappings && p->supported) + PA_IDXSET_FOREACH(m, p->input_mappings, idx) { + + if (m->input_pcm) + continue; + + pa_log_debug("Checking for recording on %s (%s)", m->description, m->name); + if (!(m->input_pcm = mapping_open_pcm(m, ss, dev_id, m->exact_channels, + SND_PCM_STREAM_CAPTURE, + default_n_fragments, + default_fragment_size_msec))) { + p->supported = false; + if (pa_idxset_size(p->input_mappings) == 1 && + ((!p->output_mappings) || pa_idxset_size(p->output_mappings) == 0)) { + pa_log_debug("Caching failure to open input:%s", m->name); + pa_hashmap_put(broken_inputs, m, m); + } + break; + } + + if (m->hw_device_index < 0) + mapping_query_hw_device(m, m->input_pcm); + } + + last = p; + + if (!p->supported) + continue; + } + + pa_log_debug("Profile %s supported.", p->name); + + if (p->output_mappings) + PA_IDXSET_FOREACH(m, p->output_mappings, idx) + if (m->output_pcm) { + found_output |= !p->fallback_output; + mapping_paths_probe(m, p, PA_ALSA_DIRECTION_OUTPUT, used_paths, mixers); + } + + if (p->input_mappings) + PA_IDXSET_FOREACH(m, p->input_mappings, idx) + if (m->input_pcm) { + found_input |= !p->fallback_input; + mapping_paths_probe(m, p, PA_ALSA_DIRECTION_INPUT, used_paths, mixers); + } + } + + /* Clean up */ + profile_finalize_probing(last, NULL); + + pa_alsa_profile_set_drop_unsupported(ps); + + paths_drop_unused(ps->input_paths, used_paths); + paths_drop_unused(ps->output_paths, used_paths); + pa_hashmap_free(broken_inputs); + pa_hashmap_free(broken_outputs); + pa_hashmap_free(used_paths); + pa_xfree(probe_order); + + profile_set_set_availability_groups(ps); + + ps->probed = true; +} + +void pa_alsa_profile_set_dump(pa_alsa_profile_set *ps) { + pa_alsa_profile *p; + pa_alsa_mapping *m; + pa_alsa_decibel_fix *db_fix; + void *state; + + pa_assert(ps); + + pa_log_debug("Profile set %p, auto_profiles=%s, probed=%s, n_mappings=%u, n_profiles=%u, n_decibel_fixes=%u", + (void*) + ps, + pa_yes_no(ps->auto_profiles), + pa_yes_no(ps->probed), + pa_hashmap_size(ps->mappings), + pa_hashmap_size(ps->profiles), + pa_hashmap_size(ps->decibel_fixes)); + + PA_HASHMAP_FOREACH(m, ps->mappings, state) + pa_alsa_mapping_dump(m); + + PA_HASHMAP_FOREACH(p, ps->profiles, state) + pa_alsa_profile_dump(p); + + PA_HASHMAP_FOREACH(db_fix, ps->decibel_fixes, state) + pa_alsa_decibel_fix_dump(db_fix); +} + +void pa_alsa_profile_set_drop_unsupported(pa_alsa_profile_set *ps) { + pa_alsa_profile *p; + pa_alsa_mapping *m; + void *state; + + PA_HASHMAP_FOREACH(p, ps->profiles, state) { + if (!p->supported) + pa_hashmap_remove_and_free(ps->profiles, p->name); + } + + PA_HASHMAP_FOREACH(m, ps->mappings, state) { + if (m->supported <= 0) + pa_hashmap_remove_and_free(ps->mappings, m->name); + } +} + +static pa_device_port* device_port_alsa_init(pa_hashmap *ports, /* card ports */ + const char* name, + const char* description, + pa_alsa_path *path, + pa_alsa_setting *setting, + pa_card_profile *cp, + pa_hashmap *extra, /* sink/source ports */ + pa_core *core) { + + pa_device_port *p; + + pa_assert(path); + + p = pa_hashmap_get(ports, name); + + if (!p) { + pa_alsa_port_data *data; + pa_device_port_new_data port_data; + + pa_device_port_new_data_init(&port_data); + pa_device_port_new_data_set_name(&port_data, name); + pa_device_port_new_data_set_description(&port_data, description); + pa_device_port_new_data_set_direction(&port_data, path->direction == PA_ALSA_DIRECTION_OUTPUT ? PA_DIRECTION_OUTPUT : PA_DIRECTION_INPUT); + pa_device_port_new_data_set_type(&port_data, path->device_port_type); + pa_device_port_new_data_set_availability_group(&port_data, path->availability_group); + + p = pa_device_port_new(core, &port_data, sizeof(pa_alsa_port_data)); + pa_device_port_new_data_done(&port_data); + pa_assert(p); + pa_hashmap_put(ports, p->name, p); + pa_proplist_update(p->proplist, PA_UPDATE_REPLACE, path->proplist); + + data = PA_DEVICE_PORT_DATA(p); + /* Ownership of the path and setting is not transferred to the port data, so we don't deal with freeing them */ + data->path = path; + data->setting = setting; + path->port = p; + } + + if (cp) + pa_hashmap_put(p->profiles, cp->name, cp); + + if (extra) { + pa_hashmap_put(extra, p->name, p); + pa_device_port_ref(p); + } + + return p; +} + +void pa_alsa_path_set_add_ports( + pa_alsa_path_set *ps, + pa_card_profile *cp, + pa_hashmap *ports, /* card ports */ + pa_hashmap *extra, /* sink/source ports */ + pa_core *core) { + + pa_alsa_path *path; + void *state; + + pa_assert(ports); + + if (!ps) + return; + + PA_HASHMAP_FOREACH(path, ps->paths, state) { + if (!path->settings || !path->settings->next) { + /* If there is no or just one setting we only need a + * single entry */ + pa_device_port *port = device_port_alsa_init(ports, path->name, + path->description, path, path->settings, cp, extra, core); + port->priority = path->priority * 100; + + } else { + pa_alsa_setting *s; + PA_LLIST_FOREACH(s, path->settings) { + pa_device_port *port; + char *n, *d; + + n = pa_sprintf_malloc("%s;%s", path->name, s->name); + + if (s->description[0]) + d = pa_sprintf_malloc("%s / %s", path->description, s->description); + else + d = pa_xstrdup(path->description); + + port = device_port_alsa_init(ports, n, d, path, s, cp, extra, core); + port->priority = path->priority * 100 + s->priority; + + pa_xfree(n); + pa_xfree(d); + } + } + } +} + +void pa_alsa_add_ports(void *sink_or_source_new_data, pa_alsa_path_set *ps, pa_card *card) { + pa_hashmap *ports; + + pa_assert(sink_or_source_new_data); + pa_assert(ps); + + if (ps->direction == PA_ALSA_DIRECTION_OUTPUT) + ports = ((pa_sink_new_data *) sink_or_source_new_data)->ports; + else + ports = ((pa_source_new_data *) sink_or_source_new_data)->ports; + + if (ps->paths && pa_hashmap_size(ps->paths) > 0) { + pa_assert(card); + pa_alsa_path_set_add_ports(ps, NULL, card->ports, ports, card->core); + } + + pa_log_debug("Added %u ports", pa_hashmap_size(ports)); +} diff --git a/src/modules/alsa/alsa-mixer.h b/src/modules/alsa/alsa-mixer.h new file mode 100644 index 0000000..db83102 --- /dev/null +++ b/src/modules/alsa/alsa-mixer.h @@ -0,0 +1,411 @@ +#ifndef fooalsamixerhfoo +#define fooalsamixerhfoo + +/*** + This file is part of PulseAudio. + + Copyright 2004-2006 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#include <alsa/asoundlib.h> + +#include <pulse/sample.h> +#include <pulse/mainloop-api.h> +#include <pulse/channelmap.h> +#include <pulse/volume.h> + +#include <pulsecore/llist.h> +#include <pulsecore/rtpoll.h> + +typedef struct pa_alsa_fdlist pa_alsa_fdlist; +typedef struct pa_alsa_mixer pa_alsa_mixer; +typedef struct pa_alsa_mixer_pdata pa_alsa_mixer_pdata; +typedef struct pa_alsa_setting pa_alsa_setting; +typedef struct pa_alsa_mixer_id pa_alsa_mixer_id; +typedef struct pa_alsa_option pa_alsa_option; +typedef struct pa_alsa_element pa_alsa_element; +typedef struct pa_alsa_jack pa_alsa_jack; +typedef struct pa_alsa_path pa_alsa_path; +typedef struct pa_alsa_path_set pa_alsa_path_set; +typedef struct pa_alsa_mapping pa_alsa_mapping; +typedef struct pa_alsa_profile pa_alsa_profile; +typedef struct pa_alsa_decibel_fix pa_alsa_decibel_fix; +typedef struct pa_alsa_profile_set pa_alsa_profile_set; +typedef struct pa_alsa_port_data pa_alsa_port_data; + +#include "alsa-util.h" +#include "alsa-ucm.h" + +#define POSITION_MASK_CHANNELS 8 + +typedef enum pa_alsa_switch_use { + PA_ALSA_SWITCH_IGNORE, + PA_ALSA_SWITCH_MUTE, /* make this switch follow mute status */ + PA_ALSA_SWITCH_OFF, /* set this switch to 'off' unconditionally */ + PA_ALSA_SWITCH_ON, /* set this switch to 'on' unconditionally */ + PA_ALSA_SWITCH_SELECT /* allow the user to select switch status through a setting */ +} pa_alsa_switch_use_t; + +typedef enum pa_alsa_volume_use { + PA_ALSA_VOLUME_IGNORE, + PA_ALSA_VOLUME_MERGE, /* merge this volume slider into the global volume slider */ + PA_ALSA_VOLUME_OFF, /* set this volume to minimal unconditionally */ + PA_ALSA_VOLUME_ZERO, /* set this volume to 0dB unconditionally */ + PA_ALSA_VOLUME_CONSTANT /* set this volume to a constant value unconditionally */ +} pa_alsa_volume_use_t; + +typedef enum pa_alsa_enumeration_use { + PA_ALSA_ENUMERATION_IGNORE, + PA_ALSA_ENUMERATION_SELECT +} pa_alsa_enumeration_use_t; + +typedef enum pa_alsa_required { + PA_ALSA_REQUIRED_IGNORE, + PA_ALSA_REQUIRED_SWITCH, + PA_ALSA_REQUIRED_VOLUME, + PA_ALSA_REQUIRED_ENUMERATION, + PA_ALSA_REQUIRED_ANY +} pa_alsa_required_t; + +typedef enum pa_alsa_direction { + PA_ALSA_DIRECTION_ANY, + PA_ALSA_DIRECTION_OUTPUT, + PA_ALSA_DIRECTION_INPUT +} pa_alsa_direction_t; + +/* A setting combines a couple of options into a single entity that + * may be selected. Only one setting can be active at the same + * time. */ +struct pa_alsa_setting { + pa_alsa_path *path; + PA_LLIST_FIELDS(pa_alsa_setting); + + pa_idxset *options; + + char *name; + char *description; + unsigned priority; +}; + +/* An entry for one ALSA mixer */ +struct pa_alsa_mixer { + snd_mixer_t *mixer_handle; + int card_index; + pa_alsa_fdlist *fdl; + bool used_for_probe_only:1; +}; + +/* ALSA mixer element identifier */ +struct pa_alsa_mixer_id { + char *name; + int index; +}; + +char *pa_alsa_mixer_id_to_string(char *dst, size_t dst_len, pa_alsa_mixer_id *id); + +/* An option belongs to an element and refers to one enumeration item + * of the element is an enumeration item, or a switch status if the + * element is a switch item. */ +struct pa_alsa_option { + pa_alsa_element *element; + PA_LLIST_FIELDS(pa_alsa_option); + + char *alsa_name; + int alsa_idx; + + char *name; + char *description; + unsigned priority; + + pa_alsa_required_t required; + pa_alsa_required_t required_any; + pa_alsa_required_t required_absent; +}; + +/* An element wraps one specific ALSA element. A series of elements + * make up a path (see below). If the element is an enumeration or switch + * element it may include a list of options. */ +struct pa_alsa_element { + pa_alsa_path *path; + PA_LLIST_FIELDS(pa_alsa_element); + + struct pa_alsa_mixer_id alsa_id; + pa_alsa_direction_t direction; + + pa_alsa_switch_use_t switch_use; + pa_alsa_volume_use_t volume_use; + pa_alsa_enumeration_use_t enumeration_use; + + pa_alsa_required_t required; + pa_alsa_required_t required_any; + pa_alsa_required_t required_absent; + + long constant_volume; + + unsigned int override_map; + bool direction_try_other:1; + + bool has_dB:1; + long min_volume, max_volume; + long volume_limit; /* -1 for no configured limit */ + double min_dB, max_dB; + + pa_channel_position_mask_t masks[SND_MIXER_SCHN_LAST + 1][POSITION_MASK_CHANNELS]; + unsigned n_channels; + + pa_channel_position_mask_t merged_mask; + + PA_LLIST_HEAD(pa_alsa_option, options); + + pa_alsa_decibel_fix *db_fix; +}; + +struct pa_alsa_jack { + pa_alsa_path *path; + PA_LLIST_FIELDS(pa_alsa_jack); + + snd_mixer_t *mixer_handle; + char *mixer_device_name; + + struct pa_alsa_mixer_id alsa_id; + char *name; /* E g "Headphone" */ + bool has_control; /* is the jack itself present? */ + bool plugged_in; /* is this jack currently plugged in? */ + snd_mixer_elem_t *melem; /* Jack detection handle */ + pa_available_t state_unplugged, state_plugged; + + pa_alsa_required_t required; + pa_alsa_required_t required_any; + pa_alsa_required_t required_absent; + + pa_dynarray *ucm_devices; /* pa_alsa_ucm_device */ + pa_dynarray *ucm_hw_mute_devices; /* pa_alsa_ucm_device */ + + bool append_pcm_to_name; +}; + +pa_alsa_jack *pa_alsa_jack_new(pa_alsa_path *path, const char *mixer_device_name, const char *name, int index); +void pa_alsa_jack_free(pa_alsa_jack *jack); +void pa_alsa_jack_set_has_control(pa_alsa_jack *jack, bool has_control); +void pa_alsa_jack_set_plugged_in(pa_alsa_jack *jack, bool plugged_in); +void pa_alsa_jack_add_ucm_device(pa_alsa_jack *jack, pa_alsa_ucm_device *device); +void pa_alsa_jack_add_ucm_hw_mute_device(pa_alsa_jack *jack, pa_alsa_ucm_device *device); + +/* A path wraps a series of elements into a single entity which can be + * used to control it as if it had a single volume slider, a single + * mute switch and a single list of selectable options. */ +struct pa_alsa_path { + pa_alsa_direction_t direction; + pa_device_port* port; + + char *name; + char *description_key; + char *description; + char *availability_group; + pa_device_port_type_t device_port_type; + unsigned priority; + bool autodetect_eld_device; + pa_alsa_mixer *eld_mixer_handle; + int eld_device; + pa_proplist *proplist; + + bool probed:1; + bool supported:1; + bool has_mute:1; + bool has_volume:1; + bool has_dB:1; + bool mute_during_activation:1; + /* These two are used during probing only */ + bool has_req_any:1; + bool req_any_present:1; + + long min_volume, max_volume; + double min_dB, max_dB; + + /* This is used during parsing only, as a shortcut so that we + * don't have to iterate the list all the time */ + pa_alsa_element *last_element; + pa_alsa_option *last_option; + pa_alsa_setting *last_setting; + pa_alsa_jack *last_jack; + + PA_LLIST_HEAD(pa_alsa_element, elements); + PA_LLIST_HEAD(pa_alsa_setting, settings); + PA_LLIST_HEAD(pa_alsa_jack, jacks); +}; + +/* A path set is simply a set of paths that are applicable to a + * device */ +struct pa_alsa_path_set { + pa_hashmap *paths; + pa_alsa_direction_t direction; +}; + +void pa_alsa_setting_dump(pa_alsa_setting *s); + +void pa_alsa_option_dump(pa_alsa_option *o); +void pa_alsa_jack_dump(pa_alsa_jack *j); +void pa_alsa_element_dump(pa_alsa_element *e); + +pa_alsa_path *pa_alsa_path_new(const char *paths_dir, const char *fname, pa_alsa_direction_t direction); +pa_alsa_path *pa_alsa_path_synthesize(const char *element, pa_alsa_direction_t direction); +pa_alsa_element *pa_alsa_element_get(pa_alsa_path *p, const char *section, bool prefixed); +int pa_alsa_path_probe(pa_alsa_path *p, pa_alsa_mapping *mapping, snd_mixer_t *m, bool ignore_dB); +void pa_alsa_path_dump(pa_alsa_path *p); +int pa_alsa_path_get_volume(pa_alsa_path *p, snd_mixer_t *m, const pa_channel_map *cm, pa_cvolume *v); +int pa_alsa_path_get_mute(pa_alsa_path *path, snd_mixer_t *m, bool *muted); +int pa_alsa_path_set_volume(pa_alsa_path *path, snd_mixer_t *m, const pa_channel_map *cm, pa_cvolume *v, bool deferred_volume, bool write_to_hw); +int pa_alsa_path_set_mute(pa_alsa_path *path, snd_mixer_t *m, bool muted); +int pa_alsa_path_select(pa_alsa_path *p, pa_alsa_setting *s, snd_mixer_t *m, bool device_is_muted); +void pa_alsa_path_set_callback(pa_alsa_path *p, snd_mixer_t *m, snd_mixer_elem_callback_t cb, void *userdata); +void pa_alsa_path_free(pa_alsa_path *p); + +pa_alsa_path_set *pa_alsa_path_set_new(pa_alsa_mapping *m, pa_alsa_direction_t direction, const char *paths_dir); +void pa_alsa_path_set_dump(pa_alsa_path_set *s); +void pa_alsa_path_set_set_callback(pa_alsa_path_set *ps, snd_mixer_t *m, snd_mixer_elem_callback_t cb, void *userdata); +void pa_alsa_path_set_free(pa_alsa_path_set *s); +int pa_alsa_path_set_is_empty(pa_alsa_path_set *s); + +struct pa_alsa_mapping { + pa_alsa_profile_set *profile_set; + + char *name; + char *description; + char *description_key; + unsigned priority; + pa_alsa_direction_t direction; + /* These are copied over to the resultant sink/source */ + pa_proplist *proplist; + + pa_sample_spec sample_spec; + pa_channel_map channel_map; + + char **device_strings; + + char **input_path_names; + char **output_path_names; + char **input_element; /* list of fallbacks */ + char **output_element; + pa_alsa_path_set *input_path_set; + pa_alsa_path_set *output_path_set; + + unsigned supported; + bool exact_channels:1; + bool fallback:1; + + /* The "y" in "hw:x,y". This is set to -1 before the device index has been + * queried, or if the query failed. */ + int hw_device_index; + + /* Temporarily used during probing */ + snd_pcm_t *input_pcm; + snd_pcm_t *output_pcm; + + pa_sink *sink; + pa_source *source; + + /* ucm device context*/ + pa_alsa_ucm_mapping_context ucm_context; +}; + +struct pa_alsa_profile { + pa_alsa_profile_set *profile_set; + + char *name; + char *description; + char *description_key; + unsigned priority; + + char *input_name; + char *output_name; + + bool supported:1; + bool fallback_input:1; + bool fallback_output:1; + + char **input_mapping_names; + char **output_mapping_names; + + pa_idxset *input_mappings; + pa_idxset *output_mappings; +}; + +struct pa_alsa_decibel_fix { + char *key; + + pa_alsa_profile_set *profile_set; + + char *name; /* Alsa volume element name. */ + int index; /* Alsa volume element index. */ + long min_step; + long max_step; + + /* An array that maps alsa volume element steps to decibels. The steps can + * be used as indices to this array, after subtracting min_step from the + * real value. + * + * The values are actually stored as integers representing millibels, + * because that's the format the alsa API uses. */ + long *db_values; +}; + +struct pa_alsa_profile_set { + pa_hashmap *mappings; + pa_hashmap *profiles; + pa_hashmap *decibel_fixes; + pa_hashmap *input_paths; + pa_hashmap *output_paths; + + bool auto_profiles; + bool ignore_dB:1; + bool probed:1; +}; + +void pa_alsa_mapping_dump(pa_alsa_mapping *m); +void pa_alsa_profile_dump(pa_alsa_profile *p); +void pa_alsa_decibel_fix_dump(pa_alsa_decibel_fix *db_fix); +pa_alsa_mapping *pa_alsa_mapping_get(pa_alsa_profile_set *ps, const char *name); + +pa_alsa_profile_set* pa_alsa_profile_set_new(const char *fname, const pa_channel_map *bonus); +void pa_alsa_profile_set_probe(pa_alsa_profile_set *ps, pa_hashmap *mixers, const char *dev_id, const pa_sample_spec *ss, unsigned default_n_fragments, unsigned default_fragment_size_msec); +void pa_alsa_profile_set_free(pa_alsa_profile_set *s); +void pa_alsa_profile_set_dump(pa_alsa_profile_set *s); +void pa_alsa_profile_set_drop_unsupported(pa_alsa_profile_set *s); + +pa_alsa_fdlist *pa_alsa_fdlist_new(void); +void pa_alsa_fdlist_free(pa_alsa_fdlist *fdl); +int pa_alsa_fdlist_set_handle(pa_alsa_fdlist *fdl, snd_mixer_t *mixer_handle, snd_hctl_t *hctl_handle, pa_mainloop_api* m); + +/* Alternative for handling alsa mixer events in io-thread. */ + +pa_alsa_mixer_pdata *pa_alsa_mixer_pdata_new(void); +void pa_alsa_mixer_pdata_free(pa_alsa_mixer_pdata *pd); +int pa_alsa_set_mixer_rtpoll(struct pa_alsa_mixer_pdata *pd, snd_mixer_t *mixer, pa_rtpoll *rtp); + +/* Data structure for inclusion in pa_device_port for alsa + * sinks/sources. This contains nothing that needs to be freed + * individually */ +struct pa_alsa_port_data { + pa_alsa_path *path; + pa_alsa_setting *setting; + bool suspend_when_unavailable; +}; + +void pa_alsa_add_ports(void *sink_or_source_new_data, pa_alsa_path_set *ps, pa_card *card); +void pa_alsa_path_set_add_ports(pa_alsa_path_set *ps, pa_card_profile *cp, pa_hashmap *ports, pa_hashmap *extra, pa_core *core); + +#endif diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c new file mode 100644 index 0000000..f7fef8a --- /dev/null +++ b/src/modules/alsa/alsa-sink.c @@ -0,0 +1,2832 @@ +/*** + This file is part of PulseAudio. + + Copyright 2004-2008 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <signal.h> +#include <stdio.h> + +#include <alsa/asoundlib.h> + +#ifdef HAVE_VALGRIND_MEMCHECK_H +#include <valgrind/memcheck.h> +#endif + +#include <pulse/rtclock.h> +#include <pulse/timeval.h> +#include <pulse/util.h> +#include <pulse/volume.h> +#include <pulse/xmalloc.h> +#include <pulse/internal.h> + +#include <pulsecore/core.h> +#include <pulsecore/i18n.h> +#include <pulsecore/module.h> +#include <pulsecore/memchunk.h> +#include <pulsecore/sink.h> +#include <pulsecore/modargs.h> +#include <pulsecore/core-rtclock.h> +#include <pulsecore/core-util.h> +#include <pulsecore/sample-util.h> +#include <pulsecore/log.h> +#include <pulsecore/macro.h> +#include <pulsecore/thread.h> +#include <pulsecore/thread-mq.h> +#include <pulsecore/rtpoll.h> +#include <pulsecore/time-smoother.h> + +#include <modules/reserve-wrap.h> + +#include "alsa-util.h" +#include "alsa-sink.h" + +/* #define DEBUG_TIMING */ + +#define DEFAULT_DEVICE "default" + +#define DEFAULT_TSCHED_BUFFER_USEC (2*PA_USEC_PER_SEC) /* 2s -- Overall buffer size */ +#define DEFAULT_TSCHED_WATERMARK_USEC (20*PA_USEC_PER_MSEC) /* 20ms -- Fill up when only this much is left in the buffer */ + +#define TSCHED_WATERMARK_INC_STEP_USEC (10*PA_USEC_PER_MSEC) /* 10ms -- On underrun, increase watermark by this */ +#define TSCHED_WATERMARK_DEC_STEP_USEC (5*PA_USEC_PER_MSEC) /* 5ms -- When everything's great, decrease watermark by this */ +#define TSCHED_WATERMARK_VERIFY_AFTER_USEC (20*PA_USEC_PER_SEC) /* 20s -- How long after a drop out recheck if things are good now */ +#define TSCHED_WATERMARK_INC_THRESHOLD_USEC (0*PA_USEC_PER_MSEC) /* 0ms -- If the buffer level ever below this threshold, increase the watermark */ +#define TSCHED_WATERMARK_DEC_THRESHOLD_USEC (100*PA_USEC_PER_MSEC) /* 100ms -- If the buffer level didn't drop below this threshold in the verification time, decrease the watermark */ + +/* Note that TSCHED_WATERMARK_INC_THRESHOLD_USEC == 0 means that we + * will increase the watermark only if we hit a real underrun. */ + +#define TSCHED_MIN_SLEEP_USEC (10*PA_USEC_PER_MSEC) /* 10ms -- Sleep at least 10ms on each iteration */ +#define TSCHED_MIN_WAKEUP_USEC (4*PA_USEC_PER_MSEC) /* 4ms -- Wakeup at least this long before the buffer runs empty*/ + +#define SMOOTHER_WINDOW_USEC (10*PA_USEC_PER_SEC) /* 10s -- smoother windows size */ +#define SMOOTHER_ADJUST_USEC (1*PA_USEC_PER_SEC) /* 1s -- smoother adjust time */ + +#define SMOOTHER_MIN_INTERVAL (2*PA_USEC_PER_MSEC) /* 2ms -- min smoother update interval */ +#define SMOOTHER_MAX_INTERVAL (200*PA_USEC_PER_MSEC) /* 200ms -- max smoother update interval */ + +#define VOLUME_ACCURACY (PA_VOLUME_NORM/100) /* don't require volume adjustments to be perfectly correct. don't necessarily extend granularity in software unless the differences get greater than this level */ + +#define DEFAULT_REWIND_SAFEGUARD_BYTES (256U) /* 1.33ms @48kHz, we'll never rewind less than this */ +#define DEFAULT_REWIND_SAFEGUARD_USEC (1330) /* 1.33ms, depending on channels/rate/sample we may rewind more than 256 above */ + +#define DEFAULT_WRITE_ITERATION_THRESHOLD 0.03 /* don't iterate write if < 3% of the buffer is available */ + +struct userdata { + pa_core *core; + pa_module *module; + pa_sink *sink; + + pa_thread *thread; + pa_thread_mq thread_mq; + pa_rtpoll *rtpoll; + + snd_pcm_t *pcm_handle; + + char *paths_dir; + pa_alsa_fdlist *mixer_fdl; + pa_alsa_mixer_pdata *mixer_pd; + pa_hashmap *mixers; + snd_mixer_t *mixer_handle; + pa_alsa_path_set *mixer_path_set; + pa_alsa_path *mixer_path; + + pa_cvolume hardware_volume; + + pa_sample_spec verified_sample_spec; + pa_sample_format_t *supported_formats; + unsigned int *supported_rates; + struct { + size_t fragment_size; + size_t nfrags; + size_t tsched_size; + size_t tsched_watermark; + size_t rewind_safeguard; + } initial_info; + + size_t + frame_size, + fragment_size, + hwbuf_size, + tsched_size, + tsched_watermark, + tsched_watermark_ref, + hwbuf_unused, + min_sleep, + min_wakeup, + watermark_inc_step, + watermark_dec_step, + watermark_inc_threshold, + watermark_dec_threshold, + rewind_safeguard; + + snd_pcm_uframes_t frames_per_block; + + pa_usec_t watermark_dec_not_before; + pa_usec_t min_latency_ref; + pa_usec_t tsched_watermark_usec; + + pa_memchunk memchunk; + + char *device_name; /* name of the PCM device */ + char *control_device; /* name of the control device */ + + bool use_mmap:1, use_tsched:1, deferred_volume:1, fixed_latency_range:1; + + bool first, after_rewind; + + pa_rtpoll_item *alsa_rtpoll_item; + + pa_smoother *smoother; + uint64_t write_count; + uint64_t since_start; + pa_usec_t smoother_interval; + pa_usec_t last_smoother_update; + + pa_idxset *formats; + + pa_reserve_wrapper *reserve; + pa_hook_slot *reserve_slot; + pa_reserve_monitor_wrapper *monitor; + pa_hook_slot *monitor_slot; + + /* ucm context */ + pa_alsa_ucm_mapping_context *ucm_context; +}; + +enum { + SINK_MESSAGE_SYNC_MIXER = PA_SINK_MESSAGE_MAX +}; + +static void userdata_free(struct userdata *u); +static int unsuspend(struct userdata *u, bool recovering); + +/* FIXME: Is there a better way to do this than device names? */ +static bool is_iec958(struct userdata *u) { + return (strncmp("iec958", u->device_name, 6) == 0); +} + +static bool is_hdmi(struct userdata *u) { + return (strncmp("hdmi", u->device_name, 4) == 0); +} + +static pa_hook_result_t reserve_cb(pa_reserve_wrapper *r, void *forced, struct userdata *u) { + pa_assert(r); + pa_assert(u); + + pa_log_debug("Suspending sink %s, because another application requested us to release the device.", u->sink->name); + + if (pa_sink_suspend(u->sink, true, PA_SUSPEND_APPLICATION) < 0) + return PA_HOOK_CANCEL; + + return PA_HOOK_OK; +} + +static void reserve_done(struct userdata *u) { + pa_assert(u); + + if (u->reserve_slot) { + pa_hook_slot_free(u->reserve_slot); + u->reserve_slot = NULL; + } + + if (u->reserve) { + pa_reserve_wrapper_unref(u->reserve); + u->reserve = NULL; + } +} + +static void reserve_update(struct userdata *u) { + const char *description; + pa_assert(u); + + if (!u->sink || !u->reserve) + return; + + if ((description = pa_proplist_gets(u->sink->proplist, PA_PROP_DEVICE_DESCRIPTION))) + pa_reserve_wrapper_set_application_device_name(u->reserve, description); +} + +static int reserve_init(struct userdata *u, const char *dname) { + char *rname; + + pa_assert(u); + pa_assert(dname); + + if (u->reserve) + return 0; + + if (pa_in_system_mode()) + return 0; + + if (!(rname = pa_alsa_get_reserve_name(dname))) + return 0; + + /* We are resuming, try to lock the device */ + u->reserve = pa_reserve_wrapper_get(u->core, rname); + pa_xfree(rname); + + if (!(u->reserve)) + return -1; + + reserve_update(u); + + pa_assert(!u->reserve_slot); + u->reserve_slot = pa_hook_connect(pa_reserve_wrapper_hook(u->reserve), PA_HOOK_NORMAL, (pa_hook_cb_t) reserve_cb, u); + + return 0; +} + +static pa_hook_result_t monitor_cb(pa_reserve_monitor_wrapper *w, void* busy, struct userdata *u) { + pa_assert(w); + pa_assert(u); + + if (PA_PTR_TO_UINT(busy) && !u->reserve) { + pa_log_debug("Suspending sink %s, because another application is blocking the access to the device.", u->sink->name); + pa_sink_suspend(u->sink, true, PA_SUSPEND_APPLICATION); + } else { + pa_log_debug("Resuming sink %s, because other applications aren't blocking access to the device any more.", u->sink->name); + pa_sink_suspend(u->sink, false, PA_SUSPEND_APPLICATION); + } + + return PA_HOOK_OK; +} + +static void monitor_done(struct userdata *u) { + pa_assert(u); + + if (u->monitor_slot) { + pa_hook_slot_free(u->monitor_slot); + u->monitor_slot = NULL; + } + + if (u->monitor) { + pa_reserve_monitor_wrapper_unref(u->monitor); + u->monitor = NULL; + } +} + +static int reserve_monitor_init(struct userdata *u, const char *dname) { + char *rname; + + pa_assert(u); + pa_assert(dname); + + if (pa_in_system_mode()) + return 0; + + if (!(rname = pa_alsa_get_reserve_name(dname))) + return 0; + + /* We are resuming, try to lock the device */ + u->monitor = pa_reserve_monitor_wrapper_get(u->core, rname); + pa_xfree(rname); + + if (!(u->monitor)) + return -1; + + pa_assert(!u->monitor_slot); + u->monitor_slot = pa_hook_connect(pa_reserve_monitor_wrapper_hook(u->monitor), PA_HOOK_NORMAL, (pa_hook_cb_t) monitor_cb, u); + + return 0; +} + +static void fix_min_sleep_wakeup(struct userdata *u) { + size_t max_use, max_use_2; + + pa_assert(u); + pa_assert(u->use_tsched); + + max_use = u->hwbuf_size - u->hwbuf_unused; + max_use_2 = pa_frame_align(max_use/2, &u->sink->sample_spec); + + u->min_sleep = pa_usec_to_bytes(TSCHED_MIN_SLEEP_USEC, &u->sink->sample_spec); + u->min_sleep = PA_CLAMP(u->min_sleep, u->frame_size, max_use_2); + + u->min_wakeup = pa_usec_to_bytes(TSCHED_MIN_WAKEUP_USEC, &u->sink->sample_spec); + u->min_wakeup = PA_CLAMP(u->min_wakeup, u->frame_size, max_use_2); +} + +static void fix_tsched_watermark(struct userdata *u) { + size_t max_use; + pa_assert(u); + pa_assert(u->use_tsched); + + max_use = u->hwbuf_size - u->hwbuf_unused; + + if (u->tsched_watermark > max_use - u->min_sleep) + u->tsched_watermark = max_use - u->min_sleep; + + if (u->tsched_watermark < u->min_wakeup) + u->tsched_watermark = u->min_wakeup; + + u->tsched_watermark_usec = pa_bytes_to_usec(u->tsched_watermark, &u->sink->sample_spec); +} + +static void increase_watermark(struct userdata *u) { + size_t old_watermark; + pa_usec_t old_min_latency, new_min_latency; + + pa_assert(u); + pa_assert(u->use_tsched); + + /* First, just try to increase the watermark */ + old_watermark = u->tsched_watermark; + u->tsched_watermark = PA_MIN(u->tsched_watermark * 2, u->tsched_watermark + u->watermark_inc_step); + fix_tsched_watermark(u); + + if (old_watermark != u->tsched_watermark) { + pa_log_info("Increasing wakeup watermark to %0.2f ms", + (double) u->tsched_watermark_usec / PA_USEC_PER_MSEC); + return; + } + + /* Hmm, we cannot increase the watermark any further, hence let's + raise the latency, unless doing so was disabled in + configuration */ + if (u->fixed_latency_range) + return; + + old_min_latency = u->sink->thread_info.min_latency; + new_min_latency = PA_MIN(old_min_latency * 2, old_min_latency + TSCHED_WATERMARK_INC_STEP_USEC); + new_min_latency = PA_MIN(new_min_latency, u->sink->thread_info.max_latency); + + if (old_min_latency != new_min_latency) { + pa_log_info("Increasing minimal latency to %0.2f ms", + (double) new_min_latency / PA_USEC_PER_MSEC); + + pa_sink_set_latency_range_within_thread(u->sink, new_min_latency, u->sink->thread_info.max_latency); + } + + /* When we reach this we're officially fucked! */ +} + +static void decrease_watermark(struct userdata *u) { + size_t old_watermark; + pa_usec_t now; + + pa_assert(u); + pa_assert(u->use_tsched); + + now = pa_rtclock_now(); + + if (u->watermark_dec_not_before <= 0) + goto restart; + + if (u->watermark_dec_not_before > now) + return; + + old_watermark = u->tsched_watermark; + + if (u->tsched_watermark < u->watermark_dec_step) + u->tsched_watermark = u->tsched_watermark / 2; + else + u->tsched_watermark = PA_MAX(u->tsched_watermark / 2, u->tsched_watermark - u->watermark_dec_step); + + fix_tsched_watermark(u); + + if (old_watermark != u->tsched_watermark) + pa_log_info("Decreasing wakeup watermark to %0.2f ms", + (double) u->tsched_watermark_usec / PA_USEC_PER_MSEC); + + /* We don't change the latency range*/ + +restart: + u->watermark_dec_not_before = now + TSCHED_WATERMARK_VERIFY_AFTER_USEC; +} + +/* Called from IO Context on unsuspend or from main thread when creating sink */ +static void reset_watermark(struct userdata *u, size_t tsched_watermark, pa_sample_spec *ss, + bool in_thread) { + u->tsched_watermark = pa_convert_size(tsched_watermark, ss, &u->sink->sample_spec); + + u->watermark_inc_step = pa_usec_to_bytes(TSCHED_WATERMARK_INC_STEP_USEC, &u->sink->sample_spec); + u->watermark_dec_step = pa_usec_to_bytes(TSCHED_WATERMARK_DEC_STEP_USEC, &u->sink->sample_spec); + + u->watermark_inc_threshold = pa_usec_to_bytes_round_up(TSCHED_WATERMARK_INC_THRESHOLD_USEC, &u->sink->sample_spec); + u->watermark_dec_threshold = pa_usec_to_bytes_round_up(TSCHED_WATERMARK_DEC_THRESHOLD_USEC, &u->sink->sample_spec); + + fix_min_sleep_wakeup(u); + fix_tsched_watermark(u); + + if (in_thread) + pa_sink_set_latency_range_within_thread(u->sink, + u->min_latency_ref, + pa_bytes_to_usec(u->hwbuf_size, ss)); + else { + pa_sink_set_latency_range(u->sink, + 0, + pa_bytes_to_usec(u->hwbuf_size, ss)); + + /* work-around assert in pa_sink_set_latency_within_thead, + keep track of min_latency and reuse it when + this routine is called from IO context */ + u->min_latency_ref = u->sink->thread_info.min_latency; + } + + pa_log_info("Time scheduling watermark is %0.2fms", + (double) u->tsched_watermark_usec / PA_USEC_PER_MSEC); +} + +static void hw_sleep_time(struct userdata *u, pa_usec_t *sleep_usec, pa_usec_t*process_usec) { + pa_usec_t usec, wm; + + pa_assert(sleep_usec); + pa_assert(process_usec); + + pa_assert(u); + pa_assert(u->use_tsched); + + usec = pa_sink_get_requested_latency_within_thread(u->sink); + + if (usec == (pa_usec_t) -1) + usec = pa_bytes_to_usec(u->hwbuf_size, &u->sink->sample_spec); + + wm = u->tsched_watermark_usec; + + if (wm > usec) + wm = usec/2; + + *sleep_usec = usec - wm; + *process_usec = wm; + +#ifdef DEBUG_TIMING + pa_log_debug("Buffer time: %lu ms; Sleep time: %lu ms; Process time: %lu ms", + (unsigned long) (usec / PA_USEC_PER_MSEC), + (unsigned long) (*sleep_usec / PA_USEC_PER_MSEC), + (unsigned long) (*process_usec / PA_USEC_PER_MSEC)); +#endif +} + +/* Reset smoother and counters */ +static void reset_vars(struct userdata *u) { + + pa_smoother_reset(u->smoother, pa_rtclock_now(), true); + u->smoother_interval = SMOOTHER_MIN_INTERVAL; + u->last_smoother_update = 0; + + u->first = true; + u->since_start = 0; + u->write_count = 0; +} + +/* Called from IO context */ +static void close_pcm(struct userdata *u) { + /* Let's suspend -- we don't call snd_pcm_drain() here since that might + * take awfully long with our long buffer sizes today. */ + snd_pcm_close(u->pcm_handle); + u->pcm_handle = NULL; + + if (u->alsa_rtpoll_item) { + pa_rtpoll_item_free(u->alsa_rtpoll_item); + u->alsa_rtpoll_item = NULL; + } +} + +static int try_recover(struct userdata *u, const char *call, int err) { + pa_assert(u); + pa_assert(call); + pa_assert(err < 0); + + pa_log_debug("%s: %s", call, pa_alsa_strerror(err)); + + pa_assert(err != -EAGAIN); + + if (err == -EPIPE) + pa_log_debug("%s: Buffer underrun!", call); + + if (err == -ESTRPIPE) + pa_log_debug("%s: System suspended!", call); + + if ((err = snd_pcm_recover(u->pcm_handle, err, 1)) < 0) { + pa_log("%s: %s, trying to restart PCM", call, pa_alsa_strerror(err)); + + /* As a last measure, restart the PCM and inform the caller about it. */ + close_pcm(u); + if (unsuspend(u, true) < 0) + return -1; + + return 1; + } + + reset_vars(u); + return 0; +} + +static size_t check_left_to_play(struct userdata *u, size_t n_bytes, bool on_timeout) { + size_t left_to_play; + bool underrun = false; + + /* We use <= instead of < for this check here because an underrun + * only happens after the last sample was processed, not already when + * it is removed from the buffer. This is particularly important + * when block transfer is used. */ + + if (n_bytes <= u->hwbuf_size) + left_to_play = u->hwbuf_size - n_bytes; + else { + + /* We got a dropout. What a mess! */ + left_to_play = 0; + underrun = true; + +#if 0 + PA_DEBUG_TRAP; +#endif + + if (!u->first && !u->after_rewind) + if (pa_log_ratelimit(PA_LOG_INFO)) + pa_log_info("Underrun!"); + } + +#ifdef DEBUG_TIMING + pa_log_debug("%0.2f ms left to play; inc threshold = %0.2f ms; dec threshold = %0.2f ms", + (double) pa_bytes_to_usec(left_to_play, &u->sink->sample_spec) / PA_USEC_PER_MSEC, + (double) pa_bytes_to_usec(u->watermark_inc_threshold, &u->sink->sample_spec) / PA_USEC_PER_MSEC, + (double) pa_bytes_to_usec(u->watermark_dec_threshold, &u->sink->sample_spec) / PA_USEC_PER_MSEC); +#endif + + if (u->use_tsched) { + bool reset_not_before = true; + + if (!u->first && !u->after_rewind) { + if (underrun || left_to_play < u->watermark_inc_threshold) + increase_watermark(u); + else if (left_to_play > u->watermark_dec_threshold) { + reset_not_before = false; + + /* We decrease the watermark only if have actually + * been woken up by a timeout. If something else woke + * us up it's too easy to fulfill the deadlines... */ + + if (on_timeout) + decrease_watermark(u); + } + } + + if (reset_not_before) + u->watermark_dec_not_before = 0; + } + + return left_to_play; +} + +static int mmap_write(struct userdata *u, pa_usec_t *sleep_usec, bool polled, bool on_timeout) { + bool work_done = false; + pa_usec_t max_sleep_usec = 0, process_usec = 0; + size_t left_to_play, input_underrun; + unsigned j = 0; + + pa_assert(u); + pa_sink_assert_ref(u->sink); + + if (u->use_tsched) + hw_sleep_time(u, &max_sleep_usec, &process_usec); + + for (;;) { + snd_pcm_sframes_t n; + size_t n_bytes; + int r; + bool after_avail = true; + + /* First we determine how many samples are missing to fill the + * buffer up to 100% */ + + if (PA_UNLIKELY((n = pa_alsa_safe_avail(u->pcm_handle, u->hwbuf_size, &u->sink->sample_spec)) < 0)) { + + if ((r = try_recover(u, "snd_pcm_avail", (int) n)) >= 0) + continue; + + return r; + } + + n_bytes = (size_t) n * u->frame_size; + +#ifdef DEBUG_TIMING + pa_log_debug("avail: %lu", (unsigned long) n_bytes); +#endif + + left_to_play = check_left_to_play(u, n_bytes, on_timeout); + on_timeout = false; + + if (u->use_tsched) + + /* We won't fill up the playback buffer before at least + * half the sleep time is over because otherwise we might + * ask for more data from the clients then they expect. We + * need to guarantee that clients only have to keep around + * a single hw buffer length. */ + + if (!polled && + pa_bytes_to_usec(left_to_play, &u->sink->sample_spec) > process_usec+max_sleep_usec/2) { +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because too early."); +#endif + break; + } + + if (PA_UNLIKELY(n_bytes <= u->hwbuf_unused)) { + + if (polled) + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(u->pcm_handle); + pa_log(_("ALSA woke us up to write new data to the device, but there was actually nothing to write.\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.\n" + "We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail."), + pa_strnull(dn)); + pa_xfree(dn); + } PA_ONCE_END; + +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because not necessary."); +#endif + break; + } + + j++; + + if (j > 10) { +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because already too many iterations."); +#endif + + break; + } else if (j >= 2 && (n_bytes < (DEFAULT_WRITE_ITERATION_THRESHOLD * (u->hwbuf_size - u->hwbuf_unused)))) { +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because <%g%% available.", DEFAULT_WRITE_ITERATION_THRESHOLD * 100); +#endif + break; + } + + n_bytes -= u->hwbuf_unused; + polled = false; + +#ifdef DEBUG_TIMING + pa_log_debug("Filling up"); +#endif + + for (;;) { + pa_memchunk chunk; + void *p; + int err; + const snd_pcm_channel_area_t *areas; + snd_pcm_uframes_t offset, frames; + snd_pcm_sframes_t sframes; + size_t written; + + frames = (snd_pcm_uframes_t) (n_bytes / u->frame_size); +/* pa_log_debug("%lu frames to write", (unsigned long) frames); */ + + if (PA_UNLIKELY((err = pa_alsa_safe_mmap_begin(u->pcm_handle, &areas, &offset, &frames, u->hwbuf_size, &u->sink->sample_spec)) < 0)) { + + if (!after_avail && err == -EAGAIN) + break; + + if ((r = try_recover(u, "snd_pcm_mmap_begin", err)) == 0) + continue; + + if (r == 1) + break; + + return r; + } + + /* Make sure that if these memblocks need to be copied they will fit into one slot */ + frames = PA_MIN(frames, u->frames_per_block); + + if (!after_avail && frames == 0) + break; + + pa_assert(frames > 0); + after_avail = false; + + /* Check these are multiples of 8 bit */ + pa_assert((areas[0].first & 7) == 0); + pa_assert((areas[0].step & 7) == 0); + + /* We assume a single interleaved memory buffer */ + pa_assert((areas[0].first >> 3) == 0); + pa_assert((areas[0].step >> 3) == u->frame_size); + + p = (uint8_t*) areas[0].addr + (offset * u->frame_size); + + written = frames * u->frame_size; + chunk.memblock = pa_memblock_new_fixed(u->core->mempool, p, written, true); + chunk.length = pa_memblock_get_length(chunk.memblock); + chunk.index = 0; + + pa_sink_render_into_full(u->sink, &chunk); + pa_memblock_unref_fixed(chunk.memblock); + + if (PA_UNLIKELY((sframes = snd_pcm_mmap_commit(u->pcm_handle, offset, frames)) < 0)) { + + if ((int) sframes == -EAGAIN) + break; + + if ((r = try_recover(u, "snd_pcm_mmap_commit", (int) sframes)) == 0) + continue; + + if (r == 1) + break; + + return r; + } + + work_done = true; + + u->write_count += written; + u->since_start += written; + +#ifdef DEBUG_TIMING + pa_log_debug("Wrote %lu bytes (of possible %lu bytes)", (unsigned long) written, (unsigned long) n_bytes); +#endif + + if (written >= n_bytes) + break; + + n_bytes -= written; + } + } + + input_underrun = pa_sink_process_input_underruns(u->sink, left_to_play); + + if (u->use_tsched) { + pa_usec_t underrun_sleep = pa_bytes_to_usec_round_up(input_underrun, &u->sink->sample_spec); + + *sleep_usec = pa_bytes_to_usec(left_to_play, &u->sink->sample_spec); + process_usec = u->tsched_watermark_usec; + + if (*sleep_usec > process_usec) + *sleep_usec -= process_usec; + else + *sleep_usec = 0; + + *sleep_usec = PA_MIN(*sleep_usec, underrun_sleep); + } else + *sleep_usec = 0; + + return work_done ? 1 : 0; +} + +static int unix_write(struct userdata *u, pa_usec_t *sleep_usec, bool polled, bool on_timeout) { + bool work_done = false; + pa_usec_t max_sleep_usec = 0, process_usec = 0; + size_t left_to_play, input_underrun; + unsigned j = 0; + + pa_assert(u); + pa_sink_assert_ref(u->sink); + + if (u->use_tsched) + hw_sleep_time(u, &max_sleep_usec, &process_usec); + + for (;;) { + snd_pcm_sframes_t n; + size_t n_bytes; + int r; + bool after_avail = true; + + if (PA_UNLIKELY((n = pa_alsa_safe_avail(u->pcm_handle, u->hwbuf_size, &u->sink->sample_spec)) < 0)) { + + if ((r = try_recover(u, "snd_pcm_avail", (int) n)) >= 0) + continue; + + return r; + } + + n_bytes = (size_t) n * u->frame_size; + +#ifdef DEBUG_TIMING + pa_log_debug("avail: %lu", (unsigned long) n_bytes); +#endif + + left_to_play = check_left_to_play(u, n_bytes, on_timeout); + on_timeout = false; + + if (u->use_tsched) + + /* We won't fill up the playback buffer before at least + * half the sleep time is over because otherwise we might + * ask for more data from the clients then they expect. We + * need to guarantee that clients only have to keep around + * a single hw buffer length. */ + + if (!polled && + pa_bytes_to_usec(left_to_play, &u->sink->sample_spec) > process_usec+max_sleep_usec/2) + break; + + if (PA_UNLIKELY(n_bytes <= u->hwbuf_unused)) { + + if (polled) + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(u->pcm_handle); + pa_log(_("ALSA woke us up to write new data to the device, but there was actually nothing to write.\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.\n" + "We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail."), + pa_strnull(dn)); + pa_xfree(dn); + } PA_ONCE_END; + + break; + } + + j++; + + if (j > 10) { +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because already too many iterations."); +#endif + + break; + } else if (j >= 2 && (n_bytes < (DEFAULT_WRITE_ITERATION_THRESHOLD * (u->hwbuf_size - u->hwbuf_unused)))) { +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because <%g%% available.", DEFAULT_WRITE_ITERATION_THRESHOLD * 100); +#endif + break; + } + + n_bytes -= u->hwbuf_unused; + polled = false; + + for (;;) { + snd_pcm_sframes_t frames; + void *p; + size_t written; + +/* pa_log_debug("%lu frames to write", (unsigned long) frames); */ + + if (u->memchunk.length <= 0) + pa_sink_render(u->sink, n_bytes, &u->memchunk); + + pa_assert(u->memchunk.length > 0); + + frames = (snd_pcm_sframes_t) (u->memchunk.length / u->frame_size); + + if (frames > (snd_pcm_sframes_t) (n_bytes/u->frame_size)) + frames = (snd_pcm_sframes_t) (n_bytes/u->frame_size); + + p = pa_memblock_acquire(u->memchunk.memblock); + frames = snd_pcm_writei(u->pcm_handle, (const uint8_t*) p + u->memchunk.index, (snd_pcm_uframes_t) frames); + pa_memblock_release(u->memchunk.memblock); + + if (PA_UNLIKELY(frames < 0)) { + + if (!after_avail && (int) frames == -EAGAIN) + break; + + if ((r = try_recover(u, "snd_pcm_writei", (int) frames)) == 0) + continue; + + if (r == 1) + break; + + return r; + } + + if (!after_avail && frames == 0) + break; + + pa_assert(frames > 0); + after_avail = false; + + written = frames * u->frame_size; + u->memchunk.index += written; + u->memchunk.length -= written; + + if (u->memchunk.length <= 0) { + pa_memblock_unref(u->memchunk.memblock); + pa_memchunk_reset(&u->memchunk); + } + + work_done = true; + + u->write_count += written; + u->since_start += written; + +/* pa_log_debug("wrote %lu frames", (unsigned long) frames); */ + + if (written >= n_bytes) + break; + + n_bytes -= written; + } + } + + input_underrun = pa_sink_process_input_underruns(u->sink, left_to_play); + + if (u->use_tsched) { + pa_usec_t underrun_sleep = pa_bytes_to_usec_round_up(input_underrun, &u->sink->sample_spec); + + *sleep_usec = pa_bytes_to_usec(left_to_play, &u->sink->sample_spec); + process_usec = u->tsched_watermark_usec; + + if (*sleep_usec > process_usec) + *sleep_usec -= process_usec; + else + *sleep_usec = 0; + + *sleep_usec = PA_MIN(*sleep_usec, underrun_sleep); + } else + *sleep_usec = 0; + + return work_done ? 1 : 0; +} + +static void update_smoother(struct userdata *u) { + snd_pcm_sframes_t delay = 0; + int64_t position; + int err; + pa_usec_t now1 = 0, now2; + snd_pcm_status_t *status; + snd_htimestamp_t htstamp = { 0, 0 }; + + snd_pcm_status_alloca(&status); + + pa_assert(u); + pa_assert(u->pcm_handle); + + /* Let's update the time smoother */ + + if (PA_UNLIKELY((err = pa_alsa_safe_delay(u->pcm_handle, status, &delay, u->hwbuf_size, &u->sink->sample_spec, false)) < 0)) { + pa_log_warn("Failed to query DSP status data: %s", pa_alsa_strerror(err)); + return; + } + + snd_pcm_status_get_htstamp(status, &htstamp); + now1 = pa_timespec_load(&htstamp); + + /* Hmm, if the timestamp is 0, then it wasn't set and we take the current time */ + if (now1 <= 0) + now1 = pa_rtclock_now(); + + /* check if the time since the last update is bigger than the interval */ + if (u->last_smoother_update > 0) + if (u->last_smoother_update + u->smoother_interval > now1) + return; + + position = (int64_t) u->write_count - ((int64_t) delay * (int64_t) u->frame_size); + + if (PA_UNLIKELY(position < 0)) + position = 0; + + now2 = pa_bytes_to_usec((uint64_t) position, &u->sink->sample_spec); + + pa_smoother_put(u->smoother, now1, now2); + + u->last_smoother_update = now1; + /* exponentially increase the update interval up to the MAX limit */ + u->smoother_interval = PA_MIN (u->smoother_interval * 2, SMOOTHER_MAX_INTERVAL); +} + +static int64_t sink_get_latency(struct userdata *u) { + int64_t delay; + pa_usec_t now1, now2; + + pa_assert(u); + + now1 = pa_rtclock_now(); + now2 = pa_smoother_get(u->smoother, now1); + + delay = (int64_t) pa_bytes_to_usec(u->write_count, &u->sink->sample_spec) - (int64_t) now2; + + if (u->memchunk.memblock) + delay += pa_bytes_to_usec(u->memchunk.length, &u->sink->sample_spec); + + return delay; +} + +static int build_pollfd(struct userdata *u) { + pa_assert(u); + pa_assert(u->pcm_handle); + + if (u->alsa_rtpoll_item) + pa_rtpoll_item_free(u->alsa_rtpoll_item); + + if (!(u->alsa_rtpoll_item = pa_alsa_build_pollfd(u->pcm_handle, u->rtpoll))) + return -1; + + return 0; +} + +/* Called from IO context */ +static void suspend(struct userdata *u) { + pa_assert(u); + + /* Handle may have been invalidated due to a device failure. + * In that case there is nothing to do. */ + if (!u->pcm_handle) + return; + + pa_smoother_pause(u->smoother, pa_rtclock_now()); + + /* Close PCM device */ + close_pcm(u); + + /* We reset max_rewind/max_request here to make sure that while we + * are suspended the old max_request/max_rewind values set before + * the suspend can influence the per-stream buffer of newly + * created streams, without their requirements having any + * influence on them. */ + pa_sink_set_max_rewind_within_thread(u->sink, 0); + pa_sink_set_max_request_within_thread(u->sink, 0); + + pa_log_info("Device suspended..."); +} + +/* Called from IO context */ +static int update_sw_params(struct userdata *u, bool may_need_rewind) { + size_t old_unused; + snd_pcm_uframes_t avail_min; + int err; + + pa_assert(u); + + /* Use the full buffer if no one asked us for anything specific */ + old_unused = u->hwbuf_unused; + u->hwbuf_unused = 0; + + if (u->use_tsched) { + pa_usec_t latency; + + if ((latency = pa_sink_get_requested_latency_within_thread(u->sink)) != (pa_usec_t) -1) { + size_t b; + + pa_log_debug("Latency set to %0.2fms", (double) latency / PA_USEC_PER_MSEC); + + b = pa_usec_to_bytes(latency, &u->sink->sample_spec); + + /* We need at least one sample in our buffer */ + + if (PA_UNLIKELY(b < u->frame_size)) + b = u->frame_size; + + u->hwbuf_unused = PA_LIKELY(b < u->hwbuf_size) ? (u->hwbuf_size - b) : 0; + } + + fix_min_sleep_wakeup(u); + fix_tsched_watermark(u); + } + + pa_log_debug("hwbuf_unused=%lu", (unsigned long) u->hwbuf_unused); + + /* We need at last one frame in the used part of the buffer */ + avail_min = (snd_pcm_uframes_t) u->hwbuf_unused / u->frame_size + 1; + + if (u->use_tsched) { + pa_usec_t sleep_usec, process_usec; + + hw_sleep_time(u, &sleep_usec, &process_usec); + avail_min += pa_usec_to_bytes(sleep_usec, &u->sink->sample_spec) / u->frame_size; + } + + pa_log_debug("setting avail_min=%lu", (unsigned long) avail_min); + + if ((err = pa_alsa_set_sw_params(u->pcm_handle, avail_min, !u->use_tsched)) < 0) { + pa_log("Failed to set software parameters: %s", pa_alsa_strerror(err)); + return err; + } + + /* If we're lowering the latency, we need to do a rewind, because otherwise + * we might end up in a situation where the hw buffer contains more data + * than the new configured latency. The rewind has to be requested before + * updating max_rewind, because the rewind amount is limited to max_rewind. + * + * If may_need_rewind is false, it means that we're just starting playback, + * and rewinding is never needed in that situation. */ + if (may_need_rewind && u->hwbuf_unused > old_unused) { + pa_log_debug("Requesting rewind due to latency change."); + pa_sink_request_rewind(u->sink, (size_t) -1); + } + + pa_sink_set_max_request_within_thread(u->sink, u->hwbuf_size - u->hwbuf_unused); + if (pa_alsa_pcm_is_hw(u->pcm_handle)) + pa_sink_set_max_rewind_within_thread(u->sink, u->hwbuf_size - u->hwbuf_unused); + else { + pa_log_info("Disabling rewind_within_thread for device %s", u->device_name); + pa_sink_set_max_rewind_within_thread(u->sink, 0); + } + + return 0; +} + +/* Called from IO Context on unsuspend */ +static void update_size(struct userdata *u, pa_sample_spec *ss) { + pa_assert(u); + pa_assert(ss); + + u->frame_size = pa_frame_size(ss); + u->frames_per_block = pa_mempool_block_size_max(u->core->mempool) / u->frame_size; + + /* use initial values including module arguments */ + u->fragment_size = u->initial_info.fragment_size; + u->hwbuf_size = u->initial_info.nfrags * u->fragment_size; + u->tsched_size = u->initial_info.tsched_size; + u->tsched_watermark = u->initial_info.tsched_watermark; + u->rewind_safeguard = u->initial_info.rewind_safeguard; + + u->tsched_watermark_ref = u->tsched_watermark; + + pa_log_info("Updated frame_size %zu, frames_per_block %lu, fragment_size %zu, hwbuf_size %zu, tsched(size %zu, watermark %zu), rewind_safeguard %zu", + u->frame_size, (unsigned long) u->frames_per_block, u->fragment_size, u->hwbuf_size, u->tsched_size, u->tsched_watermark, u->rewind_safeguard); +} + +/* Called from IO context */ +static int unsuspend(struct userdata *u, bool recovering) { + pa_sample_spec ss; + int err, i; + bool b, d; + snd_pcm_uframes_t period_frames, buffer_frames; + snd_pcm_uframes_t tsched_frames = 0; + char *device_name = NULL; + bool frame_size_changed = false; + + pa_assert(u); + pa_assert(!u->pcm_handle); + + pa_log_info("Trying resume..."); + + if ((is_iec958(u) || is_hdmi(u)) && pa_sink_is_passthrough(u->sink)) { + /* Need to open device in NONAUDIO mode */ + int len = strlen(u->device_name) + 8; + + device_name = pa_xmalloc(len); + pa_snprintf(device_name, len, "%s,AES0=6", u->device_name); + } + + /* + * On some machines, during the system suspend and resume, the thread_func could receive + * POLLERR events before the dev nodes in /dev/snd/ are accessible, and thread_func calls + * the unsuspend() to try to recover the PCM, this will make the snd_pcm_open() fail, here + * we add msleep and retry to make sure those nodes are accessible. + */ + for (i = 0; i < 4; i++) { + if ((err = snd_pcm_open(&u->pcm_handle, device_name ? device_name : u->device_name, SND_PCM_STREAM_PLAYBACK, + SND_PCM_NONBLOCK| + SND_PCM_NO_AUTO_RESAMPLE| + SND_PCM_NO_AUTO_CHANNELS| + SND_PCM_NO_AUTO_FORMAT)) < 0 && recovering) + pa_msleep(25); + else + break; + } + + if (err < 0) { + pa_log("Error opening PCM device %s: %s", u->device_name, pa_alsa_strerror(err)); + goto fail; + } + + if (pa_frame_size(&u->sink->sample_spec) != u->frame_size) { + update_size(u, &u->sink->sample_spec); + tsched_frames = u->tsched_size / u->frame_size; + frame_size_changed = true; + } + + ss = u->sink->sample_spec; + period_frames = u->fragment_size / u->frame_size; + buffer_frames = u->hwbuf_size / u->frame_size; + b = u->use_mmap; + d = u->use_tsched; + + if ((err = pa_alsa_set_hw_params(u->pcm_handle, &ss, &period_frames, &buffer_frames, tsched_frames, &b, &d, true)) < 0) { + pa_log("Failed to set hardware parameters: %s", pa_alsa_strerror(err)); + goto fail; + } + + if (b != u->use_mmap || d != u->use_tsched) { + pa_log_warn("Resume failed, couldn't get original access mode."); + goto fail; + } + + if (!pa_sample_spec_equal(&ss, &u->sink->sample_spec)) { + pa_log_warn("Resume failed, couldn't restore original sample settings."); + goto fail; + } + + if (frame_size_changed) { + u->fragment_size = (size_t)(period_frames * u->frame_size); + u->hwbuf_size = (size_t)(buffer_frames * u->frame_size); + pa_proplist_setf(u->sink->proplist, PA_PROP_DEVICE_BUFFERING_BUFFER_SIZE, "%zu", u->hwbuf_size); + pa_proplist_setf(u->sink->proplist, PA_PROP_DEVICE_BUFFERING_FRAGMENT_SIZE, "%zu", u->fragment_size); + + } else if (period_frames * u->frame_size != u->fragment_size || + buffer_frames * u->frame_size != u->hwbuf_size) { + pa_log_warn("Resume failed, couldn't restore original fragment settings. (Old: %zu/%zu, New %lu/%lu)", + u->hwbuf_size, u->fragment_size, + (unsigned long) buffer_frames * u->frame_size, (unsigned long) period_frames * u->frame_size); + goto fail; + } + + if (update_sw_params(u, false) < 0) + goto fail; + + if (build_pollfd(u) < 0) + goto fail; + + reset_vars(u); + + /* reset the watermark to the value defined when sink was created */ + if (u->use_tsched && !recovering) + reset_watermark(u, u->tsched_watermark_ref, &u->sink->sample_spec, true); + + pa_log_info("Resumed successfully..."); + + pa_xfree(device_name); + return 0; + +fail: + if (u->pcm_handle) { + snd_pcm_close(u->pcm_handle); + u->pcm_handle = NULL; + } + + pa_xfree(device_name); + + return -PA_ERR_IO; +} + +/* Called from the IO thread or the main thread depending on whether deferred + * volume is enabled or not (with deferred volume all mixer handling is done + * from the IO thread). + * + * Sets the mixer settings to match the current sink and port state (the port + * is given as an argument, because active_port may still point to the old + * port, if we're switching ports). */ +static void sync_mixer(struct userdata *u, pa_device_port *port) { + pa_alsa_setting *setting = NULL; + + pa_assert(u); + + if (!u->mixer_path) + return; + + /* port may be NULL, because if we use a synthesized mixer path, then the + * sink has no ports. */ + if (port && !u->ucm_context) { + pa_alsa_port_data *data; + + data = PA_DEVICE_PORT_DATA(port); + setting = data->setting; + } + + pa_alsa_path_select(u->mixer_path, setting, u->mixer_handle, u->sink->muted); + + if (u->sink->set_mute) + u->sink->set_mute(u->sink); + if (u->sink->flags & PA_SINK_DEFERRED_VOLUME) { + if (u->sink->write_volume) + u->sink->write_volume(u->sink); + } else { + if (u->sink->set_volume) + u->sink->set_volume(u->sink); + } +} + +/* Called from IO context */ +static int sink_process_msg(pa_msgobject *o, int code, void *data, int64_t offset, pa_memchunk *chunk) { + struct userdata *u = PA_SINK(o)->userdata; + + switch (code) { + + case PA_SINK_MESSAGE_GET_LATENCY: { + int64_t r = 0; + + if (u->pcm_handle) + r = sink_get_latency(u); + + *((int64_t*) data) = r; + + return 0; + } + + case SINK_MESSAGE_SYNC_MIXER: { + pa_device_port *port = data; + + sync_mixer(u, port); + return 0; + } + } + + return pa_sink_process_msg(o, code, data, offset, chunk); +} + +/* Called from main context */ +static int sink_set_state_in_main_thread_cb(pa_sink *s, pa_sink_state_t new_state, pa_suspend_cause_t new_suspend_cause) { + pa_sink_state_t old_state; + struct userdata *u; + + pa_sink_assert_ref(s); + pa_assert_se(u = s->userdata); + + /* When our session becomes active, we need to sync the mixer, because + * another user may have changed the mixer settings. + * + * If deferred volume is enabled, the syncing is done in the + * set_state_in_io_thread() callback instead. */ + if (!(s->flags & PA_SINK_DEFERRED_VOLUME) + && (s->suspend_cause & PA_SUSPEND_SESSION) + && !(new_suspend_cause & PA_SUSPEND_SESSION)) + sync_mixer(u, s->active_port); + + old_state = u->sink->state; + + if (PA_SINK_IS_OPENED(old_state) && new_state == PA_SINK_SUSPENDED) + reserve_done(u); + else if (old_state == PA_SINK_SUSPENDED && PA_SINK_IS_OPENED(new_state)) + if (reserve_init(u, u->device_name) < 0) + return -PA_ERR_BUSY; + + return 0; +} + +/* Called from the IO thread. */ +static int sink_set_state_in_io_thread_cb(pa_sink *s, pa_sink_state_t new_state, pa_suspend_cause_t new_suspend_cause) { + struct userdata *u; + + pa_assert(s); + pa_assert_se(u = s->userdata); + + /* When our session becomes active, we need to sync the mixer, because + * another user may have changed the mixer settings. + * + * If deferred volume is disabled, the syncing is done in the + * set_state_in_main_thread() callback instead. */ + if ((s->flags & PA_SINK_DEFERRED_VOLUME) + && (s->suspend_cause & PA_SUSPEND_SESSION) + && !(new_suspend_cause & PA_SUSPEND_SESSION)) + sync_mixer(u, s->active_port); + + /* It may be that only the suspend cause is changing, in which case there's + * nothing more to do. */ + if (new_state == s->thread_info.state) + return 0; + + switch (new_state) { + + case PA_SINK_SUSPENDED: { + pa_assert(PA_SINK_IS_OPENED(s->thread_info.state)); + + suspend(u); + + break; + } + + case PA_SINK_IDLE: + case PA_SINK_RUNNING: { + int r; + + if (s->thread_info.state == PA_SINK_INIT) { + if (build_pollfd(u) < 0) + /* FIXME: This will cause an assertion failure, because + * with the current design pa_sink_put() is not allowed + * to fail and pa_sink_put() has no fallback code that + * would start the sink suspended if opening the device + * fails. */ + return -PA_ERR_IO; + } + + if (s->thread_info.state == PA_SINK_SUSPENDED) { + if ((r = unsuspend(u, false)) < 0) + return r; + } + + break; + } + + case PA_SINK_UNLINKED: + case PA_SINK_INIT: + case PA_SINK_INVALID_STATE: + break; + } + + return 0; +} + +static int ctl_mixer_callback(snd_mixer_elem_t *elem, unsigned int mask) { + struct userdata *u = snd_mixer_elem_get_callback_private(elem); + + pa_assert(u); + pa_assert(u->mixer_handle); + + if (mask == SND_CTL_EVENT_MASK_REMOVE) + return 0; + + if (!PA_SINK_IS_LINKED(u->sink->state)) + return 0; + + if (u->sink->suspend_cause & PA_SUSPEND_SESSION) + return 0; + + if (mask & SND_CTL_EVENT_MASK_VALUE) { + pa_sink_get_volume(u->sink, true); + pa_sink_get_mute(u->sink, true); + } + + return 0; +} + +static int io_mixer_callback(snd_mixer_elem_t *elem, unsigned int mask) { + struct userdata *u = snd_mixer_elem_get_callback_private(elem); + + pa_assert(u); + pa_assert(u->mixer_handle); + + if (mask == SND_CTL_EVENT_MASK_REMOVE) + return 0; + + if (u->sink->suspend_cause & PA_SUSPEND_SESSION) + return 0; + + if (mask & SND_CTL_EVENT_MASK_VALUE) + pa_sink_update_volume_and_mute(u->sink); + + return 0; +} + +static void sink_get_volume_cb(pa_sink *s) { + struct userdata *u = s->userdata; + pa_cvolume r; + char volume_buf[PA_CVOLUME_SNPRINT_VERBOSE_MAX]; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + if (pa_alsa_path_get_volume(u->mixer_path, u->mixer_handle, &s->channel_map, &r) < 0) + return; + + /* Shift down by the base volume, so that 0dB becomes maximum volume */ + pa_sw_cvolume_multiply_scalar(&r, &r, s->base_volume); + + pa_log_debug("Read hardware volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &r, &s->channel_map, u->mixer_path->has_dB)); + + if (pa_cvolume_equal(&u->hardware_volume, &r)) + return; + + s->real_volume = u->hardware_volume = r; + + /* Hmm, so the hardware volume changed, let's reset our software volume */ + if (u->mixer_path->has_dB) + pa_sink_set_soft_volume(s, NULL); +} + +static void sink_set_volume_cb(pa_sink *s) { + struct userdata *u = s->userdata; + pa_cvolume r; + char volume_buf[PA_CVOLUME_SNPRINT_VERBOSE_MAX]; + bool deferred_volume = !!(s->flags & PA_SINK_DEFERRED_VOLUME); + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + /* Shift up by the base volume */ + pa_sw_cvolume_divide_scalar(&r, &s->real_volume, s->base_volume); + + if (pa_alsa_path_set_volume(u->mixer_path, u->mixer_handle, &s->channel_map, &r, deferred_volume, !deferred_volume) < 0) + return; + + /* Shift down by the base volume, so that 0dB becomes maximum volume */ + pa_sw_cvolume_multiply_scalar(&r, &r, s->base_volume); + + u->hardware_volume = r; + + if (u->mixer_path->has_dB) { + pa_cvolume new_soft_volume; + bool accurate_enough; + + /* Match exactly what the user requested by software */ + pa_sw_cvolume_divide(&new_soft_volume, &s->real_volume, &u->hardware_volume); + + /* If the adjustment to do in software is only minimal we + * can skip it. That saves us CPU at the expense of a bit of + * accuracy */ + accurate_enough = + (pa_cvolume_min(&new_soft_volume) >= (PA_VOLUME_NORM - VOLUME_ACCURACY)) && + (pa_cvolume_max(&new_soft_volume) <= (PA_VOLUME_NORM + VOLUME_ACCURACY)); + + pa_log_debug("Requested volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &s->real_volume, &s->channel_map, true)); + pa_log_debug("Got hardware volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &u->hardware_volume, &s->channel_map, true)); + pa_log_debug("Calculated software volume: %s (accurate-enough=%s)", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &new_soft_volume, &s->channel_map, true), + pa_yes_no(accurate_enough)); + + if (!accurate_enough) + s->soft_volume = new_soft_volume; + + } else { + pa_log_debug("Wrote hardware volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &r, &s->channel_map, false)); + + /* We can't match exactly what the user requested, hence let's + * at least tell the user about it */ + + s->real_volume = r; + } +} + +static void sink_write_volume_cb(pa_sink *s) { + struct userdata *u = s->userdata; + pa_cvolume hw_vol = s->thread_info.current_hw_volume; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + pa_assert(s->flags & PA_SINK_DEFERRED_VOLUME); + + /* Shift up by the base volume */ + pa_sw_cvolume_divide_scalar(&hw_vol, &hw_vol, s->base_volume); + + if (pa_alsa_path_set_volume(u->mixer_path, u->mixer_handle, &s->channel_map, &hw_vol, true, true) < 0) + pa_log_error("Writing HW volume failed"); + else { + pa_cvolume tmp_vol; + bool accurate_enough; + + /* Shift down by the base volume, so that 0dB becomes maximum volume */ + pa_sw_cvolume_multiply_scalar(&hw_vol, &hw_vol, s->base_volume); + + pa_sw_cvolume_divide(&tmp_vol, &hw_vol, &s->thread_info.current_hw_volume); + accurate_enough = + (pa_cvolume_min(&tmp_vol) >= (PA_VOLUME_NORM - VOLUME_ACCURACY)) && + (pa_cvolume_max(&tmp_vol) <= (PA_VOLUME_NORM + VOLUME_ACCURACY)); + + if (!accurate_enough) { + char volume_buf[2][PA_CVOLUME_SNPRINT_VERBOSE_MAX]; + + pa_log_debug("Written HW volume did not match with the request: %s (request) != %s", + pa_cvolume_snprint_verbose(volume_buf[0], + sizeof(volume_buf[0]), + &s->thread_info.current_hw_volume, + &s->channel_map, + true), + pa_cvolume_snprint_verbose(volume_buf[1], sizeof(volume_buf[1]), &hw_vol, &s->channel_map, true)); + } + } +} + +static int sink_get_mute_cb(pa_sink *s, bool *mute) { + struct userdata *u = s->userdata; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + if (pa_alsa_path_get_mute(u->mixer_path, u->mixer_handle, mute) < 0) + return -1; + + return 0; +} + +static void sink_set_mute_cb(pa_sink *s) { + struct userdata *u = s->userdata; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + pa_alsa_path_set_mute(u->mixer_path, u->mixer_handle, s->muted); +} + +static void mixer_volume_init(struct userdata *u) { + pa_assert(u); + + if (!u->mixer_path || !u->mixer_path->has_volume) { + pa_sink_set_write_volume_callback(u->sink, NULL); + pa_sink_set_get_volume_callback(u->sink, NULL); + pa_sink_set_set_volume_callback(u->sink, NULL); + + pa_log_info("Driver does not support hardware volume control, falling back to software volume control."); + } else { + pa_sink_set_get_volume_callback(u->sink, sink_get_volume_cb); + pa_sink_set_set_volume_callback(u->sink, sink_set_volume_cb); + + if (u->mixer_path->has_dB && u->deferred_volume) { + pa_sink_set_write_volume_callback(u->sink, sink_write_volume_cb); + pa_log_info("Successfully enabled deferred volume."); + } else + pa_sink_set_write_volume_callback(u->sink, NULL); + + if (u->mixer_path->has_dB) { + pa_sink_enable_decibel_volume(u->sink, true); + pa_log_info("Hardware volume ranges from %0.2f dB to %0.2f dB.", u->mixer_path->min_dB, u->mixer_path->max_dB); + + u->sink->base_volume = pa_sw_volume_from_dB(-u->mixer_path->max_dB); + u->sink->n_volume_steps = PA_VOLUME_NORM+1; + + pa_log_info("Fixing base volume to %0.2f dB", pa_sw_volume_to_dB(u->sink->base_volume)); + } else { + pa_sink_enable_decibel_volume(u->sink, false); + pa_log_info("Hardware volume ranges from %li to %li.", u->mixer_path->min_volume, u->mixer_path->max_volume); + + u->sink->base_volume = PA_VOLUME_NORM; + u->sink->n_volume_steps = u->mixer_path->max_volume - u->mixer_path->min_volume + 1; + } + + pa_log_info("Using hardware volume control. Hardware dB scale %s.", u->mixer_path->has_dB ? "supported" : "not supported"); + } + + if (!u->mixer_path || !u->mixer_path->has_mute) { + pa_sink_set_get_mute_callback(u->sink, NULL); + pa_sink_set_set_mute_callback(u->sink, NULL); + pa_log_info("Driver does not support hardware mute control, falling back to software mute control."); + } else { + pa_sink_set_get_mute_callback(u->sink, sink_get_mute_cb); + pa_sink_set_set_mute_callback(u->sink, sink_set_mute_cb); + pa_log_info("Using hardware mute control."); + } +} + +static int sink_set_port_ucm_cb(pa_sink *s, pa_device_port *p) { + struct userdata *u = s->userdata; + pa_alsa_ucm_port_data *data; + + pa_assert(u); + pa_assert(p); + pa_assert(u->ucm_context); + + data = PA_DEVICE_PORT_DATA(p); + u->mixer_path = data->path; + mixer_volume_init(u); + + if (s->flags & PA_SINK_DEFERRED_VOLUME) + pa_asyncmsgq_send(u->sink->asyncmsgq, PA_MSGOBJECT(u->sink), SINK_MESSAGE_SYNC_MIXER, p, 0, NULL); + else + sync_mixer(u, p); + + return pa_alsa_ucm_set_port(u->ucm_context, p, true); +} + +static int sink_set_port_cb(pa_sink *s, pa_device_port *p) { + struct userdata *u = s->userdata; + pa_alsa_port_data *data; + + pa_assert(u); + pa_assert(p); + pa_assert(u->mixer_handle); + pa_assert(!u->ucm_context); + + data = PA_DEVICE_PORT_DATA(p); + pa_assert_se(u->mixer_path = data->path); + mixer_volume_init(u); + + if (s->flags & PA_SINK_DEFERRED_VOLUME) + pa_asyncmsgq_send(u->sink->asyncmsgq, PA_MSGOBJECT(u->sink), SINK_MESSAGE_SYNC_MIXER, p, 0, NULL); + else + sync_mixer(u, p); + + if (data->suspend_when_unavailable && p->available == PA_AVAILABLE_NO) + pa_sink_suspend(s, true, PA_SUSPEND_UNAVAILABLE); + else + pa_sink_suspend(s, false, PA_SUSPEND_UNAVAILABLE); + + return 0; +} + +static void sink_update_requested_latency_cb(pa_sink *s) { + struct userdata *u = s->userdata; + pa_assert(u); + pa_assert(u->use_tsched); /* only when timer scheduling is used + * we can dynamically adjust the + * latency */ + + if (!u->pcm_handle) + return; + + update_sw_params(u, true); +} + +static pa_idxset* sink_get_formats(pa_sink *s) { + struct userdata *u = s->userdata; + + pa_assert(u); + + return pa_idxset_copy(u->formats, (pa_copy_func_t) pa_format_info_copy); +} + +static bool sink_set_formats(pa_sink *s, pa_idxset *formats) { + struct userdata *u = s->userdata; + pa_format_info *f, *g; + uint32_t idx, n; + + pa_assert(u); + + /* FIXME: also validate sample rates against what the device supports */ + PA_IDXSET_FOREACH(f, formats, idx) { + if (is_iec958(u) && f->encoding == PA_ENCODING_EAC3_IEC61937) + /* EAC3 cannot be sent over over S/PDIF */ + return false; + } + + pa_idxset_free(u->formats, (pa_free_cb_t) pa_format_info_free); + u->formats = pa_idxset_new(NULL, NULL); + + /* Note: the logic below won't apply if we're using software encoding. + * This is fine for now since we don't support that via the passthrough + * framework, but this must be changed if we do. */ + + /* Count how many sample rates we support */ + for (idx = 0, n = 0; u->supported_rates[idx]; idx++) + n++; + + /* First insert non-PCM formats since we prefer those. */ + PA_IDXSET_FOREACH(f, formats, idx) { + if (!pa_format_info_is_pcm(f)) { + g = pa_format_info_copy(f); + pa_format_info_set_prop_int_array(g, PA_PROP_FORMAT_RATE, (int *) u->supported_rates, n); + pa_idxset_put(u->formats, g, NULL); + } + } + + /* Now add any PCM formats */ + PA_IDXSET_FOREACH(f, formats, idx) { + if (pa_format_info_is_pcm(f)) { + /* We don't set rates here since we'll just tack on a resampler for + * unsupported rates */ + pa_idxset_put(u->formats, pa_format_info_copy(f), NULL); + } + } + + return true; +} + +static void sink_reconfigure_cb(pa_sink *s, pa_sample_spec *spec, bool passthrough) { + struct userdata *u = s->userdata; + int i; + bool format_supported = false; + bool rate_supported = false; + + pa_assert(u); + + for (i = 0; u->supported_formats[i] != PA_SAMPLE_MAX; i++) { + if (u->supported_formats[i] == spec->format) { + pa_sink_set_sample_format(u->sink, spec->format); + format_supported = true; + break; + } + } + + if (!format_supported) { + pa_log_info("Sink does not support sample format of %s, set it to a verified value", + pa_sample_format_to_string(spec->format)); + pa_sink_set_sample_format(u->sink, u->verified_sample_spec.format); + } + + for (i = 0; u->supported_rates[i]; i++) { + if (u->supported_rates[i] == spec->rate) { + pa_sink_set_sample_rate(u->sink, spec->rate); + rate_supported = true; + break; + } + } + + if (!rate_supported) { + pa_log_info("Sink does not support sample rate of %u, set it to a verified value", spec->rate); + pa_sink_set_sample_rate(u->sink, u->verified_sample_spec.rate); + } + + /* Passthrough status change is handled during unsuspend */ +} + +static int process_rewind(struct userdata *u) { + snd_pcm_sframes_t unused; + size_t rewind_nbytes, unused_nbytes, limit_nbytes; + int err; + pa_assert(u); + + if (!PA_SINK_IS_OPENED(u->sink->thread_info.state)) { + pa_sink_process_rewind(u->sink, 0); + return 0; + } + + /* Figure out how much we shall rewind and reset the counter */ + rewind_nbytes = u->sink->thread_info.rewind_nbytes; + + pa_log_debug("Requested to rewind %lu bytes.", (unsigned long) rewind_nbytes); + + if (PA_UNLIKELY((unused = pa_alsa_safe_avail(u->pcm_handle, u->hwbuf_size, &u->sink->sample_spec)) < 0)) { + if ((err = try_recover(u, "snd_pcm_avail", (int) unused)) < 0) { + pa_log_warn("Trying to recover from underrun failed during rewind"); + return -1; + } + if (err == 1) + goto rewind_done; + } + + unused_nbytes = (size_t) unused * u->frame_size; + + /* make sure rewind doesn't go too far, can cause issues with DMAs */ + unused_nbytes += u->rewind_safeguard; + + if (u->hwbuf_size > unused_nbytes) + limit_nbytes = u->hwbuf_size - unused_nbytes; + else + limit_nbytes = 0; + + if (rewind_nbytes > limit_nbytes) + rewind_nbytes = limit_nbytes; + + if (rewind_nbytes > 0) { + snd_pcm_sframes_t in_frames, out_frames; + + pa_log_debug("Limited to %lu bytes.", (unsigned long) rewind_nbytes); + + in_frames = (snd_pcm_sframes_t) (rewind_nbytes / u->frame_size); + pa_log_debug("before: %lu", (unsigned long) in_frames); + if ((out_frames = snd_pcm_rewind(u->pcm_handle, (snd_pcm_uframes_t) in_frames)) < 0) { + pa_log("snd_pcm_rewind() failed: %s", pa_alsa_strerror((int) out_frames)); + if ((err = try_recover(u, "process_rewind", out_frames)) < 0) + return -1; + if (err == 1) + goto rewind_done; + out_frames = 0; + } + + pa_log_debug("after: %lu", (unsigned long) out_frames); + + rewind_nbytes = (size_t) out_frames * u->frame_size; + + if (rewind_nbytes <= 0) + pa_log_info("Tried rewind, but was apparently not possible."); + else { + u->write_count -= rewind_nbytes; + pa_log_debug("Rewound %lu bytes.", (unsigned long) rewind_nbytes); + pa_sink_process_rewind(u->sink, rewind_nbytes); + + u->after_rewind = true; + return 0; + } + } else + pa_log_debug("Mhmm, actually there is nothing to rewind."); + +rewind_done: + pa_sink_process_rewind(u->sink, 0); + return 0; +} + +static void thread_func(void *userdata) { + struct userdata *u = userdata; + unsigned short revents = 0; + + pa_assert(u); + + pa_log_debug("Thread starting up"); + + if (u->core->realtime_scheduling) + pa_thread_make_realtime(u->core->realtime_priority); + + pa_thread_mq_install(&u->thread_mq); + + for (;;) { + int ret; + pa_usec_t rtpoll_sleep = 0, real_sleep; + +#ifdef DEBUG_TIMING + pa_log_debug("Loop"); +#endif + + if (PA_UNLIKELY(u->sink->thread_info.rewind_requested)) { + if (process_rewind(u) < 0) + goto fail; + } + + /* Render some data and write it to the dsp */ + if (PA_SINK_IS_OPENED(u->sink->thread_info.state)) { + int work_done; + pa_usec_t sleep_usec = 0; + bool on_timeout = pa_rtpoll_timer_elapsed(u->rtpoll); + + if (u->use_mmap) + work_done = mmap_write(u, &sleep_usec, revents & POLLOUT, on_timeout); + else + work_done = unix_write(u, &sleep_usec, revents & POLLOUT, on_timeout); + + if (work_done < 0) + goto fail; + +/* pa_log_debug("work_done = %i", work_done); */ + + if (work_done) { + + if (u->first) { + pa_log_info("Starting playback."); + snd_pcm_start(u->pcm_handle); + + pa_smoother_resume(u->smoother, pa_rtclock_now(), true); + + u->first = false; + } + + update_smoother(u); + } + + if (u->use_tsched) { + pa_usec_t cusec; + + if (u->since_start <= u->hwbuf_size) { + + /* USB devices on ALSA seem to hit a buffer + * underrun during the first iterations much + * quicker then we calculate here, probably due to + * the transport latency. To accommodate for that + * we artificially decrease the sleep time until + * we have filled the buffer at least once + * completely.*/ + + if (pa_log_ratelimit(PA_LOG_DEBUG)) + pa_log_debug("Cutting sleep time for the initial iterations by half."); + sleep_usec /= 2; + } + + /* OK, the playback buffer is now full, let's + * calculate when to wake up next */ +#ifdef DEBUG_TIMING + pa_log_debug("Waking up in %0.2fms (sound card clock).", (double) sleep_usec / PA_USEC_PER_MSEC); +#endif + + /* Convert from the sound card time domain to the + * system time domain */ + cusec = pa_smoother_translate(u->smoother, pa_rtclock_now(), sleep_usec); + +#ifdef DEBUG_TIMING + pa_log_debug("Waking up in %0.2fms (system clock).", (double) cusec / PA_USEC_PER_MSEC); +#endif + + /* We don't trust the conversion, so we wake up whatever comes first */ + rtpoll_sleep = PA_MIN(sleep_usec, cusec); + } + + u->after_rewind = false; + + } + + if (u->sink->flags & PA_SINK_DEFERRED_VOLUME) { + pa_usec_t volume_sleep; + pa_sink_volume_change_apply(u->sink, &volume_sleep); + if (volume_sleep > 0) { + if (rtpoll_sleep > 0) + rtpoll_sleep = PA_MIN(volume_sleep, rtpoll_sleep); + else + rtpoll_sleep = volume_sleep; + } + } + + if (rtpoll_sleep > 0) { + pa_rtpoll_set_timer_relative(u->rtpoll, rtpoll_sleep); + real_sleep = pa_rtclock_now(); + } + else + pa_rtpoll_set_timer_disabled(u->rtpoll); + + /* Hmm, nothing to do. Let's sleep */ + if ((ret = pa_rtpoll_run(u->rtpoll)) < 0) + goto fail; + + if (rtpoll_sleep > 0) { + real_sleep = pa_rtclock_now() - real_sleep; +#ifdef DEBUG_TIMING + pa_log_debug("Expected sleep: %0.2fms, real sleep: %0.2fms (diff %0.2f ms)", + (double) rtpoll_sleep / PA_USEC_PER_MSEC, (double) real_sleep / PA_USEC_PER_MSEC, + (double) ((int64_t) real_sleep - (int64_t) rtpoll_sleep) / PA_USEC_PER_MSEC); +#endif + if (u->use_tsched && real_sleep > rtpoll_sleep + u->tsched_watermark_usec) + pa_log_info("Scheduling delay of %0.2f ms > %0.2f ms, you might want to investigate this to improve latency...", + (double) (real_sleep - rtpoll_sleep) / PA_USEC_PER_MSEC, + (double) (u->tsched_watermark_usec) / PA_USEC_PER_MSEC); + } + + if (u->sink->flags & PA_SINK_DEFERRED_VOLUME) + pa_sink_volume_change_apply(u->sink, NULL); + + if (ret == 0) + goto finish; + + /* Tell ALSA about this and process its response */ + if (PA_SINK_IS_OPENED(u->sink->thread_info.state)) { + struct pollfd *pollfd; + int err; + unsigned n; + + pollfd = pa_rtpoll_item_get_pollfd(u->alsa_rtpoll_item, &n); + + if ((err = snd_pcm_poll_descriptors_revents(u->pcm_handle, pollfd, n, &revents)) < 0) { + pa_log("snd_pcm_poll_descriptors_revents() failed: %s", pa_alsa_strerror(err)); + goto fail; + } + + if (revents & ~POLLOUT) { + if ((err = pa_alsa_recover_from_poll(u->pcm_handle, revents)) < 0) + goto fail; + + /* Stream needs to be restarted */ + if (err == 1) { + close_pcm(u); + if (unsuspend(u, true) < 0) + goto fail; + } else + reset_vars(u); + + revents = 0; + } else if (revents && u->use_tsched && pa_log_ratelimit(PA_LOG_DEBUG)) + pa_log_debug("Wakeup from ALSA!"); + + } else + revents = 0; + } + +fail: + /* If this was no regular exit from the loop we have to continue + * processing messages until we received PA_MESSAGE_SHUTDOWN */ + pa_asyncmsgq_post(u->thread_mq.outq, PA_MSGOBJECT(u->core), PA_CORE_MESSAGE_UNLOAD_MODULE, u->module, 0, NULL, NULL); + pa_asyncmsgq_wait_for(u->thread_mq.inq, PA_MESSAGE_SHUTDOWN); + +finish: + pa_log_debug("Thread shutting down"); +} + +static void set_sink_name(pa_sink_new_data *data, pa_modargs *ma, const char *device_id, const char *device_name, pa_alsa_mapping *mapping) { + const char *n; + char *t; + + pa_assert(data); + pa_assert(ma); + pa_assert(device_name); + + if ((n = pa_modargs_get_value(ma, "sink_name", NULL))) { + pa_sink_new_data_set_name(data, n); + data->namereg_fail = true; + return; + } + + if ((n = pa_modargs_get_value(ma, "name", NULL))) + data->namereg_fail = true; + else { + n = device_id ? device_id : device_name; + data->namereg_fail = false; + } + + if (mapping) + t = pa_sprintf_malloc("alsa_output.%s.%s", n, mapping->name); + else + t = pa_sprintf_malloc("alsa_output.%s", n); + + pa_sink_new_data_set_name(data, t); + pa_xfree(t); +} + +static void find_mixer(struct userdata *u, pa_alsa_mapping *mapping, const char *element, bool ignore_dB) { + const char *mdev; + + if (!mapping && !element) + return; + + if (!element && mapping && pa_alsa_path_set_is_empty(mapping->output_path_set)) + return; + + u->mixers = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, + NULL, (pa_free_cb_t) pa_alsa_mixer_free); + + mdev = pa_proplist_gets(mapping->proplist, "alsa.mixer_device"); + if (mdev) { + u->mixer_handle = pa_alsa_open_mixer_by_name(u->mixers, mdev, true); + } else { + u->mixer_handle = pa_alsa_open_mixer_for_pcm(u->mixers, u->pcm_handle, true); + } + if (!u->mixer_handle) { + pa_log_info("Failed to find a working mixer device."); + return; + } + + if (element) { + + if (!(u->mixer_path = pa_alsa_path_synthesize(element, PA_ALSA_DIRECTION_OUTPUT))) + goto fail; + + if (pa_alsa_path_probe(u->mixer_path, NULL, u->mixer_handle, ignore_dB) < 0) + goto fail; + + pa_log_debug("Probed mixer path %s:", u->mixer_path->name); + pa_alsa_path_dump(u->mixer_path); + } else { + u->mixer_path_set = mapping->output_path_set; + } + + return; + +fail: + + if (u->mixer_path) { + pa_alsa_path_free(u->mixer_path); + u->mixer_path = NULL; + } + + u->mixer_handle = NULL; + pa_hashmap_free(u->mixers); + u->mixers = NULL; +} + +static int setup_mixer(struct userdata *u, bool ignore_dB) { + bool need_mixer_callback = false; + + pa_assert(u); + + /* This code is before the u->mixer_handle check, because if the UCM + * configuration doesn't specify volume or mute controls, u->mixer_handle + * will be NULL, but the UCM device enable sequence will still need to be + * executed. */ + if (u->sink->active_port && u->ucm_context) { + if (pa_alsa_ucm_set_port(u->ucm_context, u->sink->active_port, true) < 0) + return -1; + } + + if (!u->mixer_handle) + return 0; + + if (u->sink->active_port) { + if (!u->ucm_context) { + pa_alsa_port_data *data; + + /* We have a list of supported paths, so let's activate the + * one that has been chosen as active */ + + data = PA_DEVICE_PORT_DATA(u->sink->active_port); + u->mixer_path = data->path; + + pa_alsa_path_select(data->path, data->setting, u->mixer_handle, u->sink->muted); + } else { + pa_alsa_ucm_port_data *data; + + data = PA_DEVICE_PORT_DATA(u->sink->active_port); + + /* Now activate volume controls, if any */ + if (data->path) { + u->mixer_path = data->path; + pa_alsa_path_select(u->mixer_path, NULL, u->mixer_handle, u->sink->muted); + } + } + } else { + + if (!u->mixer_path && u->mixer_path_set) + u->mixer_path = pa_hashmap_first(u->mixer_path_set->paths); + + if (u->mixer_path) { + /* Hmm, we have only a single path, then let's activate it */ + + pa_alsa_path_select(u->mixer_path, u->mixer_path->settings, u->mixer_handle, u->sink->muted); + } else + return 0; + } + + mixer_volume_init(u); + + /* Will we need to register callbacks? */ + if (u->mixer_path_set && u->mixer_path_set->paths) { + pa_alsa_path *p; + void *state; + + PA_HASHMAP_FOREACH(p, u->mixer_path_set->paths, state) { + if (p->has_volume || p->has_mute) + need_mixer_callback = true; + } + } + else if (u->mixer_path) + need_mixer_callback = u->mixer_path->has_volume || u->mixer_path->has_mute; + + if (need_mixer_callback) { + int (*mixer_callback)(snd_mixer_elem_t *, unsigned int); + if (u->sink->flags & PA_SINK_DEFERRED_VOLUME) { + u->mixer_pd = pa_alsa_mixer_pdata_new(); + mixer_callback = io_mixer_callback; + + if (pa_alsa_set_mixer_rtpoll(u->mixer_pd, u->mixer_handle, u->rtpoll) < 0) { + pa_log("Failed to initialize file descriptor monitoring"); + return -1; + } + } else { + u->mixer_fdl = pa_alsa_fdlist_new(); + mixer_callback = ctl_mixer_callback; + + if (pa_alsa_fdlist_set_handle(u->mixer_fdl, u->mixer_handle, NULL, u->core->mainloop) < 0) { + pa_log("Failed to initialize file descriptor monitoring"); + return -1; + } + } + + if (u->mixer_path_set) + pa_alsa_path_set_set_callback(u->mixer_path_set, u->mixer_handle, mixer_callback, u); + else + pa_alsa_path_set_callback(u->mixer_path, u->mixer_handle, mixer_callback, u); + } + + return 0; +} + +pa_sink *pa_alsa_sink_new(pa_module *m, pa_modargs *ma, const char*driver, pa_card *card, pa_alsa_mapping *mapping) { + + struct userdata *u = NULL; + const char *dev_id = NULL, *key, *mod_name; + pa_sample_spec ss; + char *thread_name = NULL; + uint32_t alternate_sample_rate; + pa_channel_map map; + uint32_t nfrags, frag_size, buffer_size, tsched_size, tsched_watermark, rewind_safeguard; + snd_pcm_uframes_t period_frames, buffer_frames, tsched_frames; + size_t frame_size; + bool use_mmap = true; + bool use_tsched = true; + bool ignore_dB = false; + bool namereg_fail = false; + bool deferred_volume = false; + bool set_formats = false; + bool fixed_latency_range = false; + bool b; + bool d; + bool avoid_resampling; + pa_sink_new_data data; + bool volume_is_set; + bool mute_is_set; + pa_alsa_profile_set *profile_set = NULL; + void *state; + + pa_assert(m); + pa_assert(ma); + + ss = m->core->default_sample_spec; + map = m->core->default_channel_map; + avoid_resampling = m->core->avoid_resampling; + + /* Pick sample spec overrides from the mapping, if any */ + if (mapping) { + if (mapping->sample_spec.format != PA_SAMPLE_INVALID) + ss.format = mapping->sample_spec.format; + if (mapping->sample_spec.rate != 0) + ss.rate = mapping->sample_spec.rate; + if (mapping->sample_spec.channels != 0) { + ss.channels = mapping->sample_spec.channels; + if (pa_channel_map_valid(&mapping->channel_map)) + pa_assert(pa_channel_map_compatible(&mapping->channel_map, &ss)); + } + } + + /* Override with modargs if provided */ + if (pa_modargs_get_sample_spec_and_channel_map(ma, &ss, &map, PA_CHANNEL_MAP_ALSA) < 0) { + pa_log("Failed to parse sample specification and channel map"); + goto fail; + } + + alternate_sample_rate = m->core->alternate_sample_rate; + if (pa_modargs_get_alternate_sample_rate(ma, &alternate_sample_rate) < 0) { + pa_log("Failed to parse alternate sample rate"); + goto fail; + } + + frame_size = pa_frame_size(&ss); + + nfrags = m->core->default_n_fragments; + frag_size = (uint32_t) pa_usec_to_bytes(m->core->default_fragment_size_msec*PA_USEC_PER_MSEC, &ss); + if (frag_size <= 0) + frag_size = (uint32_t) frame_size; + tsched_size = (uint32_t) pa_usec_to_bytes(DEFAULT_TSCHED_BUFFER_USEC, &ss); + tsched_watermark = (uint32_t) pa_usec_to_bytes(DEFAULT_TSCHED_WATERMARK_USEC, &ss); + + if (pa_modargs_get_value_u32(ma, "fragments", &nfrags) < 0 || + pa_modargs_get_value_u32(ma, "fragment_size", &frag_size) < 0 || + pa_modargs_get_value_u32(ma, "tsched_buffer_size", &tsched_size) < 0 || + pa_modargs_get_value_u32(ma, "tsched_buffer_watermark", &tsched_watermark) < 0) { + pa_log("Failed to parse buffer metrics"); + goto fail; + } + + buffer_size = nfrags * frag_size; + + period_frames = frag_size/frame_size; + buffer_frames = buffer_size/frame_size; + tsched_frames = tsched_size/frame_size; + + if (pa_modargs_get_value_boolean(ma, "mmap", &use_mmap) < 0) { + pa_log("Failed to parse mmap argument."); + goto fail; + } + + if (pa_modargs_get_value_boolean(ma, "tsched", &use_tsched) < 0) { + pa_log("Failed to parse tsched argument."); + goto fail; + } + + if (pa_modargs_get_value_boolean(ma, "ignore_dB", &ignore_dB) < 0) { + pa_log("Failed to parse ignore_dB argument."); + goto fail; + } + + rewind_safeguard = PA_MAX(DEFAULT_REWIND_SAFEGUARD_BYTES, pa_usec_to_bytes(DEFAULT_REWIND_SAFEGUARD_USEC, &ss)); + if (pa_modargs_get_value_u32(ma, "rewind_safeguard", &rewind_safeguard) < 0) { + pa_log("Failed to parse rewind_safeguard argument"); + goto fail; + } + + deferred_volume = m->core->deferred_volume; + if (pa_modargs_get_value_boolean(ma, "deferred_volume", &deferred_volume) < 0) { + pa_log("Failed to parse deferred_volume argument."); + goto fail; + } + + if (pa_modargs_get_value_boolean(ma, "fixed_latency_range", &fixed_latency_range) < 0) { + pa_log("Failed to parse fixed_latency_range argument."); + goto fail; + } + + use_tsched = pa_alsa_may_tsched(use_tsched); + + u = pa_xnew0(struct userdata, 1); + u->core = m->core; + u->module = m; + u->use_mmap = use_mmap; + u->use_tsched = use_tsched; + u->tsched_size = tsched_size; + u->initial_info.nfrags = (size_t) nfrags; + u->initial_info.fragment_size = (size_t) frag_size; + u->initial_info.tsched_size = (size_t) tsched_size; + u->initial_info.tsched_watermark = (size_t) tsched_watermark; + u->initial_info.rewind_safeguard = (size_t) rewind_safeguard; + u->deferred_volume = deferred_volume; + u->fixed_latency_range = fixed_latency_range; + u->first = true; + u->rewind_safeguard = rewind_safeguard; + u->rtpoll = pa_rtpoll_new(); + + if (pa_thread_mq_init(&u->thread_mq, m->core->mainloop, u->rtpoll) < 0) { + pa_log("pa_thread_mq_init() failed."); + goto fail; + } + + u->smoother = pa_smoother_new( + SMOOTHER_ADJUST_USEC, + SMOOTHER_WINDOW_USEC, + true, + true, + 5, + pa_rtclock_now(), + true); + u->smoother_interval = SMOOTHER_MIN_INTERVAL; + + /* use ucm */ + if (mapping && mapping->ucm_context.ucm) + u->ucm_context = &mapping->ucm_context; + + dev_id = pa_modargs_get_value( + ma, "device_id", + pa_modargs_get_value(ma, "device", DEFAULT_DEVICE)); + + u->paths_dir = pa_xstrdup(pa_modargs_get_value(ma, "paths_dir", NULL)); + + if (reserve_init(u, dev_id) < 0) + goto fail; + + if (reserve_monitor_init(u, dev_id) < 0) + goto fail; + + b = use_mmap; + d = use_tsched; + + /* Force ALSA to reread its configuration if module-alsa-card didn't + * do it for us. This matters if our device was hot-plugged after ALSA + * has already read its configuration - see + * https://bugs.freedesktop.org/show_bug.cgi?id=54029 + */ + + if (!card) + snd_config_update_free_global(); + + if (mapping) { + + if (!(dev_id = pa_modargs_get_value(ma, "device_id", NULL))) { + pa_log("device_id= not set"); + goto fail; + } + + if ((mod_name = pa_proplist_gets(mapping->proplist, PA_ALSA_PROP_UCM_MODIFIER))) { + if (snd_use_case_set(u->ucm_context->ucm->ucm_mgr, "_enamod", mod_name) < 0) + pa_log("Failed to enable ucm modifier %s", mod_name); + else + pa_log_debug("Enabled ucm modifier %s", mod_name); + } + + if (!(u->pcm_handle = pa_alsa_open_by_device_id_mapping( + dev_id, + &u->device_name, + &ss, &map, + SND_PCM_STREAM_PLAYBACK, + &period_frames, &buffer_frames, tsched_frames, + &b, &d, mapping))) + goto fail; + + } else if ((dev_id = pa_modargs_get_value(ma, "device_id", NULL))) { + + if (!(profile_set = pa_alsa_profile_set_new(NULL, &map))) + goto fail; + + if (!(u->pcm_handle = pa_alsa_open_by_device_id_auto( + dev_id, + &u->device_name, + &ss, &map, + SND_PCM_STREAM_PLAYBACK, + &period_frames, &buffer_frames, tsched_frames, + &b, &d, profile_set, &mapping))) + goto fail; + + } else { + + if (!(u->pcm_handle = pa_alsa_open_by_device_string( + pa_modargs_get_value(ma, "device", DEFAULT_DEVICE), + &u->device_name, + &ss, &map, + SND_PCM_STREAM_PLAYBACK, + &period_frames, &buffer_frames, tsched_frames, + &b, &d, false))) + goto fail; + } + + pa_assert(u->device_name); + pa_log_info("Successfully opened device %s.", u->device_name); + + if (pa_alsa_pcm_is_modem(u->pcm_handle)) { + pa_log_notice("Device %s is modem, refusing further initialization.", u->device_name); + goto fail; + } + + if (mapping) + pa_log_info("Selected mapping '%s' (%s).", mapping->description, mapping->name); + + if (use_mmap && !b) { + pa_log_info("Device doesn't support mmap(), falling back to UNIX read/write mode."); + u->use_mmap = use_mmap = false; + } + + if (use_tsched && (!b || !d)) { + pa_log_info("Cannot enable timer-based scheduling, falling back to sound IRQ scheduling."); + u->use_tsched = use_tsched = false; + } + + if (u->use_mmap) + pa_log_info("Successfully enabled mmap() mode."); + + if (u->use_tsched) { + pa_log_info("Successfully enabled timer-based scheduling mode."); + + if (u->fixed_latency_range) + pa_log_info("Disabling latency range changes on underrun"); + } + + /* All passthrough formats supported by PulseAudio require + * IEC61937 framing with two fake channels. So, passthrough + * clients will always send two channels. Multichannel sinks + * cannot accept that, because nobody implemented sink channel count + * switching so far. So just don't show known non-working settings + * to the user. */ + if ((is_iec958(u) || is_hdmi(u)) && ss.channels == 2) + set_formats = true; + + u->verified_sample_spec = ss; + + u->supported_formats = pa_alsa_get_supported_formats(u->pcm_handle, ss.format); + if (!u->supported_formats) { + pa_log_error("Failed to find any supported sample formats."); + goto fail; + } + + u->supported_rates = pa_alsa_get_supported_rates(u->pcm_handle, ss.rate); + if (!u->supported_rates) { + pa_log_error("Failed to find any supported sample rates."); + goto fail; + } + + /* ALSA might tweak the sample spec, so recalculate the frame size */ + frame_size = pa_frame_size(&ss); + + pa_sink_new_data_init(&data); + data.driver = driver; + data.module = m; + data.card = card; + set_sink_name(&data, ma, dev_id, u->device_name, mapping); + + /* We need to give pa_modargs_get_value_boolean() a pointer to a local + * variable instead of using &data.namereg_fail directly, because + * data.namereg_fail is a bitfield and taking the address of a bitfield + * variable is impossible. */ + namereg_fail = data.namereg_fail; + if (pa_modargs_get_value_boolean(ma, "namereg_fail", &namereg_fail) < 0) { + pa_log("Failed to parse namereg_fail argument."); + pa_sink_new_data_done(&data); + goto fail; + } + data.namereg_fail = namereg_fail; + + if (pa_modargs_get_value_boolean(ma, "avoid_resampling", &avoid_resampling) < 0) { + pa_log("Failed to parse avoid_resampling argument."); + pa_sink_new_data_done(&data); + goto fail; + } + pa_sink_new_data_set_avoid_resampling(&data, avoid_resampling); + + pa_sink_new_data_set_sample_spec(&data, &ss); + pa_sink_new_data_set_channel_map(&data, &map); + pa_sink_new_data_set_alternate_sample_rate(&data, alternate_sample_rate); + + pa_alsa_init_proplist_pcm(m->core, data.proplist, u->pcm_handle); + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_STRING, u->device_name); + pa_proplist_setf(data.proplist, PA_PROP_DEVICE_BUFFERING_BUFFER_SIZE, "%lu", (unsigned long) (buffer_frames * frame_size)); + pa_proplist_setf(data.proplist, PA_PROP_DEVICE_BUFFERING_FRAGMENT_SIZE, "%lu", (unsigned long) (period_frames * frame_size)); + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_ACCESS_MODE, u->use_tsched ? "mmap+timer" : (u->use_mmap ? "mmap" : "serial")); + + if (mapping) { + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_PROFILE_NAME, mapping->name); + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_PROFILE_DESCRIPTION, mapping->description); + + state = NULL; + while ((key = pa_proplist_iterate(mapping->proplist, &state))) + pa_proplist_sets(data.proplist, key, pa_proplist_gets(mapping->proplist, key)); + } + + pa_alsa_init_description(data.proplist, card); + + if (u->control_device) + pa_alsa_init_proplist_ctl(data.proplist, u->control_device); + + if (pa_modargs_get_proplist(ma, "sink_properties", data.proplist, PA_UPDATE_REPLACE) < 0) { + pa_log("Invalid properties"); + pa_sink_new_data_done(&data); + goto fail; + } + + if (u->ucm_context) { + pa_alsa_ucm_add_ports(&data.ports, data.proplist, u->ucm_context, true, card, u->pcm_handle, ignore_dB); + find_mixer(u, mapping, pa_modargs_get_value(ma, "control", NULL), ignore_dB); + } else { + find_mixer(u, mapping, pa_modargs_get_value(ma, "control", NULL), ignore_dB); + if (u->mixer_path_set) + pa_alsa_add_ports(&data, u->mixer_path_set, card); + } + + u->sink = pa_sink_new(m->core, &data, PA_SINK_HARDWARE | PA_SINK_LATENCY | (u->use_tsched ? PA_SINK_DYNAMIC_LATENCY : 0) | + (set_formats ? PA_SINK_SET_FORMATS : 0)); + volume_is_set = data.volume_is_set; + mute_is_set = data.muted_is_set; + pa_sink_new_data_done(&data); + + if (!u->sink) { + pa_log("Failed to create sink object"); + goto fail; + } + + if (u->ucm_context) { + pa_device_port *port; + unsigned h_prio = 0; + PA_HASHMAP_FOREACH(port, u->sink->ports, state) { + if (!h_prio || port->priority > h_prio) + h_prio = port->priority; + } + /* ucm ports prioriy is 100, 200, ..., 900, change it to units digit */ + h_prio = h_prio / 100; + u->sink->priority += h_prio; + } + + if (pa_modargs_get_value_u32(ma, "deferred_volume_safety_margin", + &u->sink->thread_info.volume_change_safety_margin) < 0) { + pa_log("Failed to parse deferred_volume_safety_margin parameter"); + goto fail; + } + + if (pa_modargs_get_value_s32(ma, "deferred_volume_extra_delay", + &u->sink->thread_info.volume_change_extra_delay) < 0) { + pa_log("Failed to parse deferred_volume_extra_delay parameter"); + goto fail; + } + + u->sink->parent.process_msg = sink_process_msg; + if (u->use_tsched) + u->sink->update_requested_latency = sink_update_requested_latency_cb; + u->sink->set_state_in_main_thread = sink_set_state_in_main_thread_cb; + u->sink->set_state_in_io_thread = sink_set_state_in_io_thread_cb; + if (u->ucm_context) + u->sink->set_port = sink_set_port_ucm_cb; + else + u->sink->set_port = sink_set_port_cb; + u->sink->reconfigure = sink_reconfigure_cb; + u->sink->userdata = u; + + pa_sink_set_asyncmsgq(u->sink, u->thread_mq.inq); + pa_sink_set_rtpoll(u->sink, u->rtpoll); + + u->frame_size = frame_size; + u->frames_per_block = pa_mempool_block_size_max(m->core->mempool) / frame_size; + u->fragment_size = frag_size = (size_t) (period_frames * frame_size); + u->hwbuf_size = buffer_size = (size_t) (buffer_frames * frame_size); + pa_cvolume_mute(&u->hardware_volume, u->sink->sample_spec.channels); + + pa_log_info("Using %0.1f fragments of size %lu bytes (%0.2fms), buffer size is %lu bytes (%0.2fms)", + (double) u->hwbuf_size / (double) u->fragment_size, + (long unsigned) u->fragment_size, + (double) pa_bytes_to_usec(u->fragment_size, &ss) / PA_USEC_PER_MSEC, + (long unsigned) u->hwbuf_size, + (double) pa_bytes_to_usec(u->hwbuf_size, &ss) / PA_USEC_PER_MSEC); + + pa_sink_set_max_request(u->sink, u->hwbuf_size); + if (pa_alsa_pcm_is_hw(u->pcm_handle)) + pa_sink_set_max_rewind(u->sink, u->hwbuf_size); + else { + pa_log_info("Disabling rewind for device %s", u->device_name); + pa_sink_set_max_rewind(u->sink, 0); + } + + if (u->use_tsched) { + u->tsched_watermark_ref = tsched_watermark; + reset_watermark(u, u->tsched_watermark_ref, &ss, false); + } else + pa_sink_set_fixed_latency(u->sink, pa_bytes_to_usec(u->hwbuf_size, &ss)); + + reserve_update(u); + + if (update_sw_params(u, false) < 0) + goto fail; + + if (setup_mixer(u, ignore_dB) < 0) + goto fail; + + pa_alsa_dump(PA_LOG_DEBUG, u->pcm_handle); + + thread_name = pa_sprintf_malloc("alsa-sink-%s", pa_strnull(pa_proplist_gets(u->sink->proplist, "alsa.id"))); + if (!(u->thread = pa_thread_new(thread_name, thread_func, u))) { + pa_log("Failed to create thread."); + goto fail; + } + pa_xfree(thread_name); + thread_name = NULL; + + /* Get initial mixer settings */ + if (volume_is_set) { + if (u->sink->set_volume) + u->sink->set_volume(u->sink); + } else { + if (u->sink->get_volume) + u->sink->get_volume(u->sink); + } + + if (mute_is_set) { + if (u->sink->set_mute) + u->sink->set_mute(u->sink); + } else { + if (u->sink->get_mute) { + bool mute; + + if (u->sink->get_mute(u->sink, &mute) >= 0) + pa_sink_set_mute(u->sink, mute, false); + } + } + + if ((volume_is_set || mute_is_set) && u->sink->write_volume) + u->sink->write_volume(u->sink); + + if (set_formats) { + /* For S/PDIF and HDMI, allow getting/setting custom formats */ + pa_format_info *format; + + /* To start with, we only support PCM formats. Other formats may be added + * with pa_sink_set_formats().*/ + format = pa_format_info_new(); + format->encoding = PA_ENCODING_PCM; + u->formats = pa_idxset_new(NULL, NULL); + pa_idxset_put(u->formats, format, NULL); + + u->sink->get_formats = sink_get_formats; + u->sink->set_formats = sink_set_formats; + } + + pa_sink_put(u->sink); + + if (profile_set) + pa_alsa_profile_set_free(profile_set); + + /* Suspend if necessary. FIXME: It would be better to start suspended, but + * that would require some core changes. It's possible to set + * pa_sink_new_data.suspend_cause, but that has to be done before the + * pa_sink_new() call, and we know if we need to suspend only after the + * pa_sink_new() call when the initial port has been chosen. Calling + * pa_sink_suspend() between pa_sink_new() and pa_sink_put() would + * otherwise work, but currently pa_sink_suspend() will crash if + * pa_sink_put() hasn't been called. */ + if (u->sink->active_port && !u->ucm_context) { + pa_alsa_port_data *port_data; + + port_data = PA_DEVICE_PORT_DATA(u->sink->active_port); + + if (port_data->suspend_when_unavailable && u->sink->active_port->available == PA_AVAILABLE_NO) + pa_sink_suspend(u->sink, true, PA_SUSPEND_UNAVAILABLE); + } + + return u->sink; + +fail: + pa_xfree(thread_name); + + if (u) + userdata_free(u); + + if (profile_set) + pa_alsa_profile_set_free(profile_set); + + return NULL; +} + +static void userdata_free(struct userdata *u) { + pa_assert(u); + + if (u->sink) + pa_sink_unlink(u->sink); + + if (u->thread) { + pa_asyncmsgq_send(u->thread_mq.inq, NULL, PA_MESSAGE_SHUTDOWN, NULL, 0, NULL); + pa_thread_free(u->thread); + } + + pa_thread_mq_done(&u->thread_mq); + + if (u->sink) + pa_sink_unref(u->sink); + + if (u->memchunk.memblock) + pa_memblock_unref(u->memchunk.memblock); + + if (u->mixer_pd) + pa_alsa_mixer_pdata_free(u->mixer_pd); + + if (u->alsa_rtpoll_item) + pa_rtpoll_item_free(u->alsa_rtpoll_item); + + if (u->rtpoll) + pa_rtpoll_free(u->rtpoll); + + if (u->pcm_handle) { + snd_pcm_drop(u->pcm_handle); + snd_pcm_close(u->pcm_handle); + } + + if (u->mixer_fdl) + pa_alsa_fdlist_free(u->mixer_fdl); + + /* Only free the mixer_path if the sink owns it */ + if (u->mixer_path && !u->mixer_path_set && !u->ucm_context) + pa_alsa_path_free(u->mixer_path); + + if (u->mixers) + pa_hashmap_free(u->mixers); + + if (u->smoother) + pa_smoother_free(u->smoother); + + if (u->formats) + pa_idxset_free(u->formats, (pa_free_cb_t) pa_format_info_free); + + if (u->supported_formats) + pa_xfree(u->supported_formats); + + if (u->supported_rates) + pa_xfree(u->supported_rates); + + reserve_done(u); + monitor_done(u); + + pa_xfree(u->device_name); + pa_xfree(u->control_device); + pa_xfree(u->paths_dir); + pa_xfree(u); +} + +void pa_alsa_sink_free(pa_sink *s) { + struct userdata *u; + + pa_sink_assert_ref(s); + pa_assert_se(u = s->userdata); + + userdata_free(u); +} diff --git a/src/modules/alsa/alsa-sink.h b/src/modules/alsa/alsa-sink.h new file mode 100644 index 0000000..78a2cb2 --- /dev/null +++ b/src/modules/alsa/alsa-sink.h @@ -0,0 +1,34 @@ +#ifndef fooalsasinkhfoo +#define fooalsasinkhfoo + +/*** + This file is part of PulseAudio. + + Copyright 2004-2006 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#include <pulsecore/module.h> +#include <pulsecore/modargs.h> +#include <pulsecore/sink.h> + +#include "alsa-util.h" + +pa_sink* pa_alsa_sink_new(pa_module *m, pa_modargs *ma, const char*driver, pa_card *card, pa_alsa_mapping *mapping); + +void pa_alsa_sink_free(pa_sink *s); + +#endif diff --git a/src/modules/alsa/alsa-source.c b/src/modules/alsa/alsa-source.c new file mode 100644 index 0000000..76370f8 --- /dev/null +++ b/src/modules/alsa/alsa-source.c @@ -0,0 +1,2473 @@ +/*** + This file is part of PulseAudio. + + Copyright 2004-2008 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <signal.h> +#include <stdio.h> + +#include <alsa/asoundlib.h> + +#include <pulse/rtclock.h> +#include <pulse/timeval.h> +#include <pulse/util.h> +#include <pulse/volume.h> +#include <pulse/xmalloc.h> + +#include <pulsecore/core.h> +#include <pulsecore/i18n.h> +#include <pulsecore/module.h> +#include <pulsecore/memchunk.h> +#include <pulsecore/sink.h> +#include <pulsecore/modargs.h> +#include <pulsecore/core-rtclock.h> +#include <pulsecore/core-util.h> +#include <pulsecore/sample-util.h> +#include <pulsecore/log.h> +#include <pulsecore/macro.h> +#include <pulsecore/thread.h> +#include <pulsecore/thread-mq.h> +#include <pulsecore/rtpoll.h> +#include <pulsecore/time-smoother.h> + +#include <modules/reserve-wrap.h> + +#include "alsa-util.h" +#include "alsa-source.h" + +/* #define DEBUG_TIMING */ + +#define DEFAULT_DEVICE "default" + +#define DEFAULT_TSCHED_BUFFER_USEC (2*PA_USEC_PER_SEC) /* 2s */ +#define DEFAULT_TSCHED_WATERMARK_USEC (20*PA_USEC_PER_MSEC) /* 20ms */ + +#define TSCHED_WATERMARK_INC_STEP_USEC (10*PA_USEC_PER_MSEC) /* 10ms */ +#define TSCHED_WATERMARK_DEC_STEP_USEC (5*PA_USEC_PER_MSEC) /* 5ms */ +#define TSCHED_WATERMARK_VERIFY_AFTER_USEC (20*PA_USEC_PER_SEC) /* 20s */ +#define TSCHED_WATERMARK_INC_THRESHOLD_USEC (0*PA_USEC_PER_MSEC) /* 0ms */ +#define TSCHED_WATERMARK_DEC_THRESHOLD_USEC (100*PA_USEC_PER_MSEC) /* 100ms */ +#define TSCHED_WATERMARK_STEP_USEC (10*PA_USEC_PER_MSEC) /* 10ms */ + +#define TSCHED_MIN_SLEEP_USEC (10*PA_USEC_PER_MSEC) /* 10ms */ +#define TSCHED_MIN_WAKEUP_USEC (4*PA_USEC_PER_MSEC) /* 4ms */ + +#define SMOOTHER_WINDOW_USEC (10*PA_USEC_PER_SEC) /* 10s */ +#define SMOOTHER_ADJUST_USEC (1*PA_USEC_PER_SEC) /* 1s */ + +#define SMOOTHER_MIN_INTERVAL (2*PA_USEC_PER_MSEC) /* 2ms */ +#define SMOOTHER_MAX_INTERVAL (200*PA_USEC_PER_MSEC) /* 200ms */ + +#define VOLUME_ACCURACY (PA_VOLUME_NORM/100) + +struct userdata { + pa_core *core; + pa_module *module; + pa_source *source; + + pa_thread *thread; + pa_thread_mq thread_mq; + pa_rtpoll *rtpoll; + + snd_pcm_t *pcm_handle; + + char *paths_dir; + pa_alsa_fdlist *mixer_fdl; + pa_alsa_mixer_pdata *mixer_pd; + pa_hashmap *mixers; + snd_mixer_t *mixer_handle; + pa_alsa_path_set *mixer_path_set; + pa_alsa_path *mixer_path; + + pa_cvolume hardware_volume; + + pa_sample_spec verified_sample_spec; + pa_sample_format_t *supported_formats; + unsigned int *supported_rates; + struct { + size_t fragment_size; + size_t nfrags; + size_t tsched_size; + size_t tsched_watermark; + } initial_info; + + size_t + frame_size, + fragment_size, + hwbuf_size, + tsched_size, + tsched_watermark, + tsched_watermark_ref, + hwbuf_unused, + min_sleep, + min_wakeup, + watermark_inc_step, + watermark_dec_step, + watermark_inc_threshold, + watermark_dec_threshold; + + snd_pcm_uframes_t frames_per_block; + + pa_usec_t watermark_dec_not_before; + pa_usec_t min_latency_ref; + pa_usec_t tsched_watermark_usec; + + char *device_name; /* name of the PCM device */ + char *control_device; /* name of the control device */ + + bool use_mmap:1, use_tsched:1, deferred_volume:1, fixed_latency_range:1; + + bool first; + + pa_rtpoll_item *alsa_rtpoll_item; + + pa_smoother *smoother; + uint64_t read_count; + pa_usec_t smoother_interval; + pa_usec_t last_smoother_update; + + pa_reserve_wrapper *reserve; + pa_hook_slot *reserve_slot; + pa_reserve_monitor_wrapper *monitor; + pa_hook_slot *monitor_slot; + + /* ucm context */ + pa_alsa_ucm_mapping_context *ucm_context; +}; + +enum { + SOURCE_MESSAGE_SYNC_MIXER = PA_SOURCE_MESSAGE_MAX +}; + +static void userdata_free(struct userdata *u); +static int unsuspend(struct userdata *u, bool recovering); + +static pa_hook_result_t reserve_cb(pa_reserve_wrapper *r, void *forced, struct userdata *u) { + pa_assert(r); + pa_assert(u); + + pa_log_debug("Suspending source %s, because another application requested us to release the device.", u->source->name); + + if (pa_source_suspend(u->source, true, PA_SUSPEND_APPLICATION) < 0) + return PA_HOOK_CANCEL; + + return PA_HOOK_OK; +} + +static void reserve_done(struct userdata *u) { + pa_assert(u); + + if (u->reserve_slot) { + pa_hook_slot_free(u->reserve_slot); + u->reserve_slot = NULL; + } + + if (u->reserve) { + pa_reserve_wrapper_unref(u->reserve); + u->reserve = NULL; + } +} + +static void reserve_update(struct userdata *u) { + const char *description; + pa_assert(u); + + if (!u->source || !u->reserve) + return; + + if ((description = pa_proplist_gets(u->source->proplist, PA_PROP_DEVICE_DESCRIPTION))) + pa_reserve_wrapper_set_application_device_name(u->reserve, description); +} + +static int reserve_init(struct userdata *u, const char *dname) { + char *rname; + + pa_assert(u); + pa_assert(dname); + + if (u->reserve) + return 0; + + if (pa_in_system_mode()) + return 0; + + if (!(rname = pa_alsa_get_reserve_name(dname))) + return 0; + + /* We are resuming, try to lock the device */ + u->reserve = pa_reserve_wrapper_get(u->core, rname); + pa_xfree(rname); + + if (!(u->reserve)) + return -1; + + reserve_update(u); + + pa_assert(!u->reserve_slot); + u->reserve_slot = pa_hook_connect(pa_reserve_wrapper_hook(u->reserve), PA_HOOK_NORMAL, (pa_hook_cb_t) reserve_cb, u); + + return 0; +} + +static pa_hook_result_t monitor_cb(pa_reserve_monitor_wrapper *w, void* busy, struct userdata *u) { + pa_assert(w); + pa_assert(u); + + if (PA_PTR_TO_UINT(busy) && !u->reserve) { + pa_log_debug("Suspending source %s, because another application is blocking the access to the device.", u->source->name); + pa_source_suspend(u->source, true, PA_SUSPEND_APPLICATION); + } else { + pa_log_debug("Resuming source %s, because other applications aren't blocking access to the device any more.", u->source->name); + pa_source_suspend(u->source, false, PA_SUSPEND_APPLICATION); + } + + return PA_HOOK_OK; +} + +static void monitor_done(struct userdata *u) { + pa_assert(u); + + if (u->monitor_slot) { + pa_hook_slot_free(u->monitor_slot); + u->monitor_slot = NULL; + } + + if (u->monitor) { + pa_reserve_monitor_wrapper_unref(u->monitor); + u->monitor = NULL; + } +} + +static int reserve_monitor_init(struct userdata *u, const char *dname) { + char *rname; + + pa_assert(u); + pa_assert(dname); + + if (pa_in_system_mode()) + return 0; + + if (!(rname = pa_alsa_get_reserve_name(dname))) + return 0; + + /* We are resuming, try to lock the device */ + u->monitor = pa_reserve_monitor_wrapper_get(u->core, rname); + pa_xfree(rname); + + if (!(u->monitor)) + return -1; + + pa_assert(!u->monitor_slot); + u->monitor_slot = pa_hook_connect(pa_reserve_monitor_wrapper_hook(u->monitor), PA_HOOK_NORMAL, (pa_hook_cb_t) monitor_cb, u); + + return 0; +} + +static void fix_min_sleep_wakeup(struct userdata *u) { + size_t max_use, max_use_2; + + pa_assert(u); + pa_assert(u->use_tsched); + + max_use = u->hwbuf_size - u->hwbuf_unused; + max_use_2 = pa_frame_align(max_use/2, &u->source->sample_spec); + + u->min_sleep = pa_usec_to_bytes(TSCHED_MIN_SLEEP_USEC, &u->source->sample_spec); + u->min_sleep = PA_CLAMP(u->min_sleep, u->frame_size, max_use_2); + + u->min_wakeup = pa_usec_to_bytes(TSCHED_MIN_WAKEUP_USEC, &u->source->sample_spec); + u->min_wakeup = PA_CLAMP(u->min_wakeup, u->frame_size, max_use_2); +} + +static void fix_tsched_watermark(struct userdata *u) { + size_t max_use; + pa_assert(u); + pa_assert(u->use_tsched); + + max_use = u->hwbuf_size - u->hwbuf_unused; + + if (u->tsched_watermark > max_use - u->min_sleep) + u->tsched_watermark = max_use - u->min_sleep; + + if (u->tsched_watermark < u->min_wakeup) + u->tsched_watermark = u->min_wakeup; + + u->tsched_watermark_usec = pa_bytes_to_usec(u->tsched_watermark, &u->source->sample_spec); +} + +static void increase_watermark(struct userdata *u) { + size_t old_watermark; + pa_usec_t old_min_latency, new_min_latency; + + pa_assert(u); + pa_assert(u->use_tsched); + + /* First, just try to increase the watermark */ + old_watermark = u->tsched_watermark; + u->tsched_watermark = PA_MIN(u->tsched_watermark * 2, u->tsched_watermark + u->watermark_inc_step); + fix_tsched_watermark(u); + + if (old_watermark != u->tsched_watermark) { + pa_log_info("Increasing wakeup watermark to %0.2f ms", + (double) u->tsched_watermark_usec / PA_USEC_PER_MSEC); + return; + } + + /* Hmm, we cannot increase the watermark any further, hence let's + raise the latency unless doing so was disabled in + configuration */ + if (u->fixed_latency_range) + return; + + old_min_latency = u->source->thread_info.min_latency; + new_min_latency = PA_MIN(old_min_latency * 2, old_min_latency + TSCHED_WATERMARK_INC_STEP_USEC); + new_min_latency = PA_MIN(new_min_latency, u->source->thread_info.max_latency); + + if (old_min_latency != new_min_latency) { + pa_log_info("Increasing minimal latency to %0.2f ms", + (double) new_min_latency / PA_USEC_PER_MSEC); + + pa_source_set_latency_range_within_thread(u->source, new_min_latency, u->source->thread_info.max_latency); + } + + /* When we reach this we're officially fucked! */ +} + +static void decrease_watermark(struct userdata *u) { + size_t old_watermark; + pa_usec_t now; + + pa_assert(u); + pa_assert(u->use_tsched); + + now = pa_rtclock_now(); + + if (u->watermark_dec_not_before <= 0) + goto restart; + + if (u->watermark_dec_not_before > now) + return; + + old_watermark = u->tsched_watermark; + + if (u->tsched_watermark < u->watermark_dec_step) + u->tsched_watermark = u->tsched_watermark / 2; + else + u->tsched_watermark = PA_MAX(u->tsched_watermark / 2, u->tsched_watermark - u->watermark_dec_step); + + fix_tsched_watermark(u); + + if (old_watermark != u->tsched_watermark) + pa_log_info("Decreasing wakeup watermark to %0.2f ms", + (double) u->tsched_watermark_usec / PA_USEC_PER_MSEC); + + /* We don't change the latency range*/ + +restart: + u->watermark_dec_not_before = now + TSCHED_WATERMARK_VERIFY_AFTER_USEC; +} + +/* Called from IO Context on unsuspend or from main thread when creating source */ +static void reset_watermark(struct userdata *u, size_t tsched_watermark, pa_sample_spec *ss, + bool in_thread) { + u->tsched_watermark = pa_convert_size(tsched_watermark, ss, &u->source->sample_spec); + + u->watermark_inc_step = pa_usec_to_bytes(TSCHED_WATERMARK_INC_STEP_USEC, &u->source->sample_spec); + u->watermark_dec_step = pa_usec_to_bytes(TSCHED_WATERMARK_DEC_STEP_USEC, &u->source->sample_spec); + + u->watermark_inc_threshold = pa_usec_to_bytes_round_up(TSCHED_WATERMARK_INC_THRESHOLD_USEC, &u->source->sample_spec); + u->watermark_dec_threshold = pa_usec_to_bytes_round_up(TSCHED_WATERMARK_DEC_THRESHOLD_USEC, &u->source->sample_spec); + + fix_min_sleep_wakeup(u); + fix_tsched_watermark(u); + + if (in_thread) + pa_source_set_latency_range_within_thread(u->source, + u->min_latency_ref, + pa_bytes_to_usec(u->hwbuf_size, ss)); + else { + pa_source_set_latency_range(u->source, + 0, + pa_bytes_to_usec(u->hwbuf_size, ss)); + + /* work-around assert in pa_source_set_latency_within_thead, + keep track of min_latency and reuse it when + this routine is called from IO context */ + u->min_latency_ref = u->source->thread_info.min_latency; + } + + pa_log_info("Time scheduling watermark is %0.2fms", + (double) u->tsched_watermark_usec / PA_USEC_PER_MSEC); +} + +static void hw_sleep_time(struct userdata *u, pa_usec_t *sleep_usec, pa_usec_t*process_usec) { + pa_usec_t wm, usec; + + pa_assert(sleep_usec); + pa_assert(process_usec); + + pa_assert(u); + pa_assert(u->use_tsched); + + usec = pa_source_get_requested_latency_within_thread(u->source); + + if (usec == (pa_usec_t) -1) + usec = pa_bytes_to_usec(u->hwbuf_size, &u->source->sample_spec); + + wm = u->tsched_watermark_usec; + + if (wm > usec) + wm = usec/2; + + *sleep_usec = usec - wm; + *process_usec = wm; + +#ifdef DEBUG_TIMING + pa_log_debug("Buffer time: %lu ms; Sleep time: %lu ms; Process time: %lu ms", + (unsigned long) (usec / PA_USEC_PER_MSEC), + (unsigned long) (*sleep_usec / PA_USEC_PER_MSEC), + (unsigned long) (*process_usec / PA_USEC_PER_MSEC)); +#endif +} + +/* Reset smoother and counters */ +static void reset_vars(struct userdata *u) { + + pa_smoother_reset(u->smoother, pa_rtclock_now(), true); + u->smoother_interval = SMOOTHER_MIN_INTERVAL; + u->last_smoother_update = 0; + + u->read_count = 0; + u->first = true; +} + +/* Called from IO context */ +static void close_pcm(struct userdata *u) { + pa_smoother_pause(u->smoother, pa_rtclock_now()); + + /* Let's suspend */ + snd_pcm_close(u->pcm_handle); + u->pcm_handle = NULL; + + if (u->alsa_rtpoll_item) { + pa_rtpoll_item_free(u->alsa_rtpoll_item); + u->alsa_rtpoll_item = NULL; + } +} + +static int try_recover(struct userdata *u, const char *call, int err) { + pa_assert(u); + pa_assert(call); + pa_assert(err < 0); + + pa_log_debug("%s: %s", call, pa_alsa_strerror(err)); + + pa_assert(err != -EAGAIN); + + if (err == -EPIPE) + pa_log_debug("%s: Buffer overrun!", call); + + if (err == -ESTRPIPE) + pa_log_debug("%s: System suspended!", call); + + if ((err = snd_pcm_recover(u->pcm_handle, err, 1)) < 0) { + pa_log("%s: %s, trying to restart PCM", call, pa_alsa_strerror(err)); + + /* As a last measure, restart the PCM and inform the caller about it. */ + close_pcm(u); + if (unsuspend(u, true) < 0) + return -1; + + return 1; + } + + reset_vars(u); + return 0; +} + +static size_t check_left_to_record(struct userdata *u, size_t n_bytes, bool on_timeout) { + size_t left_to_record; + size_t rec_space = u->hwbuf_size - u->hwbuf_unused; + bool overrun = false; + + /* We use <= instead of < for this check here because an overrun + * only happens after the last sample was processed, not already when + * it is removed from the buffer. This is particularly important + * when block transfer is used. */ + + if (n_bytes <= rec_space) + left_to_record = rec_space - n_bytes; + else { + + /* We got a dropout. What a mess! */ + left_to_record = 0; + overrun = true; + +#ifdef DEBUG_TIMING + PA_DEBUG_TRAP; +#endif + + if (pa_log_ratelimit(PA_LOG_INFO)) + pa_log_info("Overrun!"); + } + +#ifdef DEBUG_TIMING + pa_log_debug("%0.2f ms left to record", (double) pa_bytes_to_usec(left_to_record, &u->source->sample_spec) / PA_USEC_PER_MSEC); +#endif + + if (u->use_tsched) { + bool reset_not_before = true; + + if (overrun || left_to_record < u->watermark_inc_threshold) + increase_watermark(u); + else if (left_to_record > u->watermark_dec_threshold) { + reset_not_before = false; + + /* We decrease the watermark only if have actually + * been woken up by a timeout. If something else woke + * us up it's too easy to fulfill the deadlines... */ + + if (on_timeout) + decrease_watermark(u); + } + + if (reset_not_before) + u->watermark_dec_not_before = 0; + } + + return left_to_record; +} + +static int mmap_read(struct userdata *u, pa_usec_t *sleep_usec, bool polled, bool on_timeout) { + bool work_done = false; + bool recovery_done = false; + pa_usec_t max_sleep_usec = 0, process_usec = 0; + size_t left_to_record; + unsigned j = 0; + + pa_assert(u); + pa_source_assert_ref(u->source); + + if (u->use_tsched) + hw_sleep_time(u, &max_sleep_usec, &process_usec); + + for (;;) { + snd_pcm_sframes_t n; + size_t n_bytes; + int r; + bool after_avail = true; + + if (PA_UNLIKELY((n = pa_alsa_safe_avail(u->pcm_handle, u->hwbuf_size, &u->source->sample_spec)) < 0)) { + + recovery_done = true; + if ((r = try_recover(u, "snd_pcm_avail", (int) n)) >= 0) + continue; + + return r; + } + + n_bytes = (size_t) n * u->frame_size; + +#ifdef DEBUG_TIMING + pa_log_debug("avail: %lu", (unsigned long) n_bytes); +#endif + + left_to_record = check_left_to_record(u, n_bytes, on_timeout); + on_timeout = false; + + if (u->use_tsched) + if (!polled && + pa_bytes_to_usec(left_to_record, &u->source->sample_spec) > process_usec+max_sleep_usec/2) { +#ifdef DEBUG_TIMING + pa_log_debug("Not reading, because too early."); +#endif + break; + } + + if (PA_UNLIKELY(n_bytes <= 0)) { + + if (polled) + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(u->pcm_handle); + pa_log(_("ALSA woke us up to read new data from the device, but there was actually nothing to read.\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.\n" + "We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail."), + pa_strnull(dn)); + pa_xfree(dn); + } PA_ONCE_END; + +#ifdef DEBUG_TIMING + pa_log_debug("Not reading, because not necessary."); +#endif + break; + } + + if (++j > 10) { +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because already too many iterations."); +#endif + + break; + } + + polled = false; + +#ifdef DEBUG_TIMING + pa_log_debug("Reading"); +#endif + + for (;;) { + pa_memchunk chunk; + void *p; + int err; + const snd_pcm_channel_area_t *areas; + snd_pcm_uframes_t offset, frames; + snd_pcm_sframes_t sframes; + + frames = (snd_pcm_uframes_t) (n_bytes / u->frame_size); +/* pa_log_debug("%lu frames to read", (unsigned long) frames); */ + + if (PA_UNLIKELY((err = pa_alsa_safe_mmap_begin(u->pcm_handle, &areas, &offset, &frames, u->hwbuf_size, &u->source->sample_spec)) < 0)) { + + if (!after_avail && err == -EAGAIN) + break; + + recovery_done = true; + if ((r = try_recover(u, "snd_pcm_mmap_begin", err)) == 0) + continue; + + if (r == 1) + break; + + return r; + } + + /* Make sure that if these memblocks need to be copied they will fit into one slot */ + frames = PA_MIN(frames, u->frames_per_block); + + if (!after_avail && frames == 0) + break; + + pa_assert(frames > 0); + after_avail = false; + + /* Check these are multiples of 8 bit */ + pa_assert((areas[0].first & 7) == 0); + pa_assert((areas[0].step & 7) == 0); + + /* We assume a single interleaved memory buffer */ + pa_assert((areas[0].first >> 3) == 0); + pa_assert((areas[0].step >> 3) == u->frame_size); + + p = (uint8_t*) areas[0].addr + (offset * u->frame_size); + + chunk.memblock = pa_memblock_new_fixed(u->core->mempool, p, frames * u->frame_size, true); + chunk.length = pa_memblock_get_length(chunk.memblock); + chunk.index = 0; + + pa_source_post(u->source, &chunk); + pa_memblock_unref_fixed(chunk.memblock); + + if (PA_UNLIKELY((sframes = snd_pcm_mmap_commit(u->pcm_handle, offset, frames)) < 0)) { + + recovery_done = true; + if ((r = try_recover(u, "snd_pcm_mmap_commit", (int) sframes)) == 0) + continue; + + if (r == 1) + break; + + return r; + } + + work_done = true; + + u->read_count += frames * u->frame_size; + +#ifdef DEBUG_TIMING + pa_log_debug("Read %lu bytes (of possible %lu bytes)", (unsigned long) (frames * u->frame_size), (unsigned long) n_bytes); +#endif + + if ((size_t) frames * u->frame_size >= n_bytes) + break; + + n_bytes -= (size_t) frames * u->frame_size; + } + } + + if (u->use_tsched) { + *sleep_usec = pa_bytes_to_usec(left_to_record, &u->source->sample_spec); + process_usec = u->tsched_watermark_usec; + + if (*sleep_usec > process_usec) + *sleep_usec -= process_usec; + else + *sleep_usec = 0; + + /* If the PCM was recovered, it may need restarting. Reduce the sleep time + * to 0 to ensure immediate restart. */ + if (recovery_done) + *sleep_usec = 0; + } + + return work_done ? 1 : 0; +} + +static int unix_read(struct userdata *u, pa_usec_t *sleep_usec, bool polled, bool on_timeout) { + int work_done = false; + bool recovery_done = false; + pa_usec_t max_sleep_usec = 0, process_usec = 0; + size_t left_to_record; + unsigned j = 0; + + pa_assert(u); + pa_source_assert_ref(u->source); + + if (u->use_tsched) + hw_sleep_time(u, &max_sleep_usec, &process_usec); + + for (;;) { + snd_pcm_sframes_t n; + size_t n_bytes; + int r; + bool after_avail = true; + + if (PA_UNLIKELY((n = pa_alsa_safe_avail(u->pcm_handle, u->hwbuf_size, &u->source->sample_spec)) < 0)) { + + recovery_done = true; + if ((r = try_recover(u, "snd_pcm_avail", (int) n)) >= 0) + continue; + + return r; + } + + n_bytes = (size_t) n * u->frame_size; + left_to_record = check_left_to_record(u, n_bytes, on_timeout); + on_timeout = false; + + if (u->use_tsched) + if (!polled && + pa_bytes_to_usec(left_to_record, &u->source->sample_spec) > process_usec+max_sleep_usec/2) + break; + + if (PA_UNLIKELY(n_bytes <= 0)) { + + if (polled) + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(u->pcm_handle); + pa_log(_("ALSA woke us up to read new data from the device, but there was actually nothing to read.\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.\n" + "We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail."), + pa_strnull(dn)); + pa_xfree(dn); + } PA_ONCE_END; + + break; + } + + if (++j > 10) { +#ifdef DEBUG_TIMING + pa_log_debug("Not filling up, because already too many iterations."); +#endif + + break; + } + + polled = false; + + for (;;) { + void *p; + snd_pcm_sframes_t frames; + pa_memchunk chunk; + + chunk.memblock = pa_memblock_new(u->core->mempool, (size_t) -1); + + frames = (snd_pcm_sframes_t) (pa_memblock_get_length(chunk.memblock) / u->frame_size); + + if (frames > (snd_pcm_sframes_t) (n_bytes/u->frame_size)) + frames = (snd_pcm_sframes_t) (n_bytes/u->frame_size); + +/* pa_log_debug("%lu frames to read", (unsigned long) n); */ + + p = pa_memblock_acquire(chunk.memblock); + frames = snd_pcm_readi(u->pcm_handle, (uint8_t*) p, (snd_pcm_uframes_t) frames); + pa_memblock_release(chunk.memblock); + + if (PA_UNLIKELY(frames < 0)) { + pa_memblock_unref(chunk.memblock); + + if (!after_avail && (int) frames == -EAGAIN) + break; + + recovery_done = true; + if ((r = try_recover(u, "snd_pcm_readi", (int) frames)) == 0) + continue; + + if (r == 1) + break; + + return r; + } + + if (!after_avail && frames == 0) { + pa_memblock_unref(chunk.memblock); + break; + } + + pa_assert(frames > 0); + after_avail = false; + + chunk.index = 0; + chunk.length = (size_t) frames * u->frame_size; + + pa_source_post(u->source, &chunk); + pa_memblock_unref(chunk.memblock); + + work_done = true; + + u->read_count += frames * u->frame_size; + +/* pa_log_debug("read %lu frames", (unsigned long) frames); */ + + if ((size_t) frames * u->frame_size >= n_bytes) + break; + + n_bytes -= (size_t) frames * u->frame_size; + } + } + + if (u->use_tsched) { + *sleep_usec = pa_bytes_to_usec(left_to_record, &u->source->sample_spec); + process_usec = u->tsched_watermark_usec; + + if (*sleep_usec > process_usec) + *sleep_usec -= process_usec; + else + *sleep_usec = 0; + + /* If the PCM was recovered, it may need restarting. Reduce the sleep time + * to 0 to ensure immediate restart. */ + if (recovery_done) + *sleep_usec = 0; + } + + return work_done ? 1 : 0; +} + +static void update_smoother(struct userdata *u) { + snd_pcm_sframes_t delay = 0; + uint64_t position; + int err; + pa_usec_t now1 = 0, now2; + snd_pcm_status_t *status; + snd_htimestamp_t htstamp = { 0, 0 }; + + snd_pcm_status_alloca(&status); + + pa_assert(u); + pa_assert(u->pcm_handle); + + /* Let's update the time smoother */ + + if (PA_UNLIKELY((err = pa_alsa_safe_delay(u->pcm_handle, status, &delay, u->hwbuf_size, &u->source->sample_spec, true)) < 0)) { + pa_log_warn("Failed to get delay: %s", pa_alsa_strerror(err)); + return; + } + + snd_pcm_status_get_htstamp(status, &htstamp); + now1 = pa_timespec_load(&htstamp); + + /* Hmm, if the timestamp is 0, then it wasn't set and we take the current time */ + if (now1 <= 0) + now1 = pa_rtclock_now(); + + /* check if the time since the last update is bigger than the interval */ + if (u->last_smoother_update > 0) + if (u->last_smoother_update + u->smoother_interval > now1) + return; + + position = u->read_count + ((uint64_t) delay * (uint64_t) u->frame_size); + now2 = pa_bytes_to_usec(position, &u->source->sample_spec); + + pa_smoother_put(u->smoother, now1, now2); + + u->last_smoother_update = now1; + /* exponentially increase the update interval up to the MAX limit */ + u->smoother_interval = PA_MIN (u->smoother_interval * 2, SMOOTHER_MAX_INTERVAL); +} + +static int64_t source_get_latency(struct userdata *u) { + int64_t delay; + pa_usec_t now1, now2; + + pa_assert(u); + + now1 = pa_rtclock_now(); + now2 = pa_smoother_get(u->smoother, now1); + + delay = (int64_t) now2 - (int64_t) pa_bytes_to_usec(u->read_count, &u->source->sample_spec); + + return delay; +} + +static int build_pollfd(struct userdata *u) { + pa_assert(u); + pa_assert(u->pcm_handle); + + if (u->alsa_rtpoll_item) + pa_rtpoll_item_free(u->alsa_rtpoll_item); + + if (!(u->alsa_rtpoll_item = pa_alsa_build_pollfd(u->pcm_handle, u->rtpoll))) + return -1; + + return 0; +} + +/* Called from IO context */ +static void suspend(struct userdata *u) { + pa_assert(u); + + /* PCM may have been invalidated due to device failure. + * In that case, there is nothing to do. */ + if (!u->pcm_handle) + return; + + /* Close PCM device */ + close_pcm(u); + + pa_log_info("Device suspended..."); +} + +/* Called from IO context */ +static int update_sw_params(struct userdata *u) { + snd_pcm_uframes_t avail_min; + int err; + + pa_assert(u); + + /* Use the full buffer if no one asked us for anything specific */ + u->hwbuf_unused = 0; + + if (u->use_tsched) { + pa_usec_t latency; + + if ((latency = pa_source_get_requested_latency_within_thread(u->source)) != (pa_usec_t) -1) { + size_t b; + + pa_log_debug("latency set to %0.2fms", (double) latency / PA_USEC_PER_MSEC); + + b = pa_usec_to_bytes(latency, &u->source->sample_spec); + + /* We need at least one sample in our buffer */ + + if (PA_UNLIKELY(b < u->frame_size)) + b = u->frame_size; + + u->hwbuf_unused = PA_LIKELY(b < u->hwbuf_size) ? (u->hwbuf_size - b) : 0; + } + + fix_min_sleep_wakeup(u); + fix_tsched_watermark(u); + } + + pa_log_debug("hwbuf_unused=%lu", (unsigned long) u->hwbuf_unused); + + avail_min = 1; + + if (u->use_tsched) { + pa_usec_t sleep_usec, process_usec; + + hw_sleep_time(u, &sleep_usec, &process_usec); + avail_min += pa_usec_to_bytes(sleep_usec, &u->source->sample_spec) / u->frame_size; + } + + pa_log_debug("setting avail_min=%lu", (unsigned long) avail_min); + + if ((err = pa_alsa_set_sw_params(u->pcm_handle, avail_min, !u->use_tsched)) < 0) { + pa_log("Failed to set software parameters: %s", pa_alsa_strerror(err)); + return err; + } + + return 0; +} + +/* Called from IO Context on unsuspend */ +static void update_size(struct userdata *u, pa_sample_spec *ss) { + pa_assert(u); + pa_assert(ss); + + u->frame_size = pa_frame_size(ss); + u->frames_per_block = pa_mempool_block_size_max(u->core->mempool) / u->frame_size; + + /* use initial values including module arguments */ + u->fragment_size = u->initial_info.fragment_size; + u->hwbuf_size = u->initial_info.nfrags * u->fragment_size; + u->tsched_size = u->initial_info.tsched_size; + u->tsched_watermark = u->initial_info.tsched_watermark; + + u->tsched_watermark_ref = u->tsched_watermark; + + pa_log_info("Updated frame_size %zu, frames_per_block %lu, fragment_size %zu, hwbuf_size %zu, tsched(size %zu, watermark %zu)", + u->frame_size, (unsigned long) u->frames_per_block, u->fragment_size, u->hwbuf_size, u->tsched_size, u->tsched_watermark); +} + +/* Called from IO context */ +static int unsuspend(struct userdata *u, bool recovering) { + pa_sample_spec ss; + int err, i; + bool b, d; + snd_pcm_uframes_t period_frames, buffer_frames; + snd_pcm_uframes_t tsched_frames = 0; + bool frame_size_changed = false; + + pa_assert(u); + pa_assert(!u->pcm_handle); + + pa_log_info("Trying resume..."); + + /* + * On some machines, during the system suspend and resume, the thread_func could receive + * POLLERR events before the dev nodes in /dev/snd/ are accessible, and thread_func calls + * the unsuspend() to try to recover the PCM, this will make the snd_pcm_open() fail, here + * we add msleep and retry to make sure those nodes are accessible. + */ + for (i = 0; i < 4; i++) { + if ((err = snd_pcm_open(&u->pcm_handle, u->device_name, SND_PCM_STREAM_CAPTURE, + SND_PCM_NONBLOCK| + SND_PCM_NO_AUTO_RESAMPLE| + SND_PCM_NO_AUTO_CHANNELS| + SND_PCM_NO_AUTO_FORMAT)) < 0 && recovering) + pa_msleep(25); + else + break; + } + + if (err < 0) { + pa_log("Error opening PCM device %s: %s", u->device_name, pa_alsa_strerror(err)); + goto fail; + } + + if (pa_frame_size(&u->source->sample_spec) != u->frame_size) { + update_size(u, &u->source->sample_spec); + tsched_frames = u->tsched_size / u->frame_size; + frame_size_changed = true; + } + + ss = u->source->sample_spec; + period_frames = u->fragment_size / u->frame_size; + buffer_frames = u->hwbuf_size / u->frame_size; + b = u->use_mmap; + d = u->use_tsched; + + if ((err = pa_alsa_set_hw_params(u->pcm_handle, &ss, &period_frames, &buffer_frames, tsched_frames, &b, &d, true)) < 0) { + pa_log("Failed to set hardware parameters: %s", pa_alsa_strerror(err)); + goto fail; + } + + if (b != u->use_mmap || d != u->use_tsched) { + pa_log_warn("Resume failed, couldn't get original access mode."); + goto fail; + } + + if (!pa_sample_spec_equal(&ss, &u->source->sample_spec)) { + pa_log_warn("Resume failed, couldn't restore original sample settings."); + goto fail; + } + + if (frame_size_changed) { + u->fragment_size = (size_t)(period_frames * u->frame_size); + u->hwbuf_size = (size_t)(buffer_frames * u->frame_size); + pa_proplist_setf(u->source->proplist, PA_PROP_DEVICE_BUFFERING_BUFFER_SIZE, "%zu", u->hwbuf_size); + pa_proplist_setf(u->source->proplist, PA_PROP_DEVICE_BUFFERING_FRAGMENT_SIZE, "%zu", u->fragment_size); + + } else if (period_frames * u->frame_size != u->fragment_size || + buffer_frames * u->frame_size != u->hwbuf_size) { + pa_log_warn("Resume failed, couldn't restore original fragment settings. (Old: %zu/%zu, New %lu/%lu)", + u->hwbuf_size, u->fragment_size, + (unsigned long) buffer_frames * u->frame_size, (unsigned long) period_frames * u->frame_size); + goto fail; + } + + if (update_sw_params(u) < 0) + goto fail; + + if (build_pollfd(u) < 0) + goto fail; + + /* FIXME: We need to reload the volume somehow */ + + reset_vars(u); + + /* reset the watermark to the value defined when source was created */ + if (u->use_tsched && !recovering) + reset_watermark(u, u->tsched_watermark_ref, &u->source->sample_spec, true); + + pa_log_info("Resumed successfully..."); + + return 0; + +fail: + if (u->pcm_handle) { + snd_pcm_close(u->pcm_handle); + u->pcm_handle = NULL; + } + + return -PA_ERR_IO; +} + +/* Called from the IO thread or the main thread depending on whether deferred + * volume is enabled or not (with deferred volume all mixer handling is done + * from the IO thread). + * + * Sets the mixer settings to match the current source and port state (the port + * is given as an argument, because active_port may still point to the old + * port, if we're switching ports). */ +static void sync_mixer(struct userdata *u, pa_device_port *port) { + pa_alsa_setting *setting = NULL; + + pa_assert(u); + + if (!u->mixer_path) + return; + + /* port may be NULL, because if we use a synthesized mixer path, then the + * source has no ports. */ + if (port && !u->ucm_context) { + pa_alsa_port_data *data; + + data = PA_DEVICE_PORT_DATA(port); + setting = data->setting; + } + + pa_alsa_path_select(u->mixer_path, setting, u->mixer_handle, u->source->muted); + + if (u->source->set_mute) + u->source->set_mute(u->source); + if (u->source->flags & PA_SOURCE_DEFERRED_VOLUME) { + if (u->source->write_volume) + u->source->write_volume(u->source); + } else { + if (u->source->set_volume) + u->source->set_volume(u->source); + } +} + +/* Called from IO context */ +static int source_process_msg(pa_msgobject *o, int code, void *data, int64_t offset, pa_memchunk *chunk) { + struct userdata *u = PA_SOURCE(o)->userdata; + + switch (code) { + + case PA_SOURCE_MESSAGE_GET_LATENCY: { + int64_t r = 0; + + if (u->pcm_handle) + r = source_get_latency(u); + + *((int64_t*) data) = r; + + return 0; + } + + case SOURCE_MESSAGE_SYNC_MIXER: { + pa_device_port *port = data; + + sync_mixer(u, port); + return 0; + } + } + + return pa_source_process_msg(o, code, data, offset, chunk); +} + +/* Called from main context */ +static int source_set_state_in_main_thread_cb(pa_source *s, pa_source_state_t new_state, pa_suspend_cause_t new_suspend_cause) { + pa_source_state_t old_state; + struct userdata *u; + + pa_source_assert_ref(s); + pa_assert_se(u = s->userdata); + + /* When our session becomes active, we need to sync the mixer, because + * another user may have changed the mixer settings. + * + * If deferred volume is enabled, the syncing is done in the + * set_state_in_io_thread() callback instead. */ + if (!(s->flags & PA_SOURCE_DEFERRED_VOLUME) + && (s->suspend_cause & PA_SUSPEND_SESSION) + && !(new_suspend_cause & PA_SUSPEND_SESSION)) + sync_mixer(u, s->active_port); + + old_state = u->source->state; + + if (PA_SOURCE_IS_OPENED(old_state) && new_state == PA_SOURCE_SUSPENDED) + reserve_done(u); + else if (old_state == PA_SOURCE_SUSPENDED && PA_SOURCE_IS_OPENED(new_state)) + if (reserve_init(u, u->device_name) < 0) + return -PA_ERR_BUSY; + + return 0; +} + +/* Called from the IO thread. */ +static int source_set_state_in_io_thread_cb(pa_source *s, pa_source_state_t new_state, pa_suspend_cause_t new_suspend_cause) { + struct userdata *u; + + pa_assert(s); + pa_assert_se(u = s->userdata); + + /* When our session becomes active, we need to sync the mixer, because + * another user may have changed the mixer settings. + * + * If deferred volume is disabled, the syncing is done in the + * set_state_in_main_thread() callback instead. */ + if ((s->flags & PA_SOURCE_DEFERRED_VOLUME) + && (s->suspend_cause & PA_SUSPEND_SESSION) + && !(new_suspend_cause & PA_SUSPEND_SESSION)) + sync_mixer(u, s->active_port); + + /* It may be that only the suspend cause is changing, in which case there's + * nothing more to do. */ + if (new_state == s->thread_info.state) + return 0; + + switch (new_state) { + + case PA_SOURCE_SUSPENDED: { + pa_assert(PA_SOURCE_IS_OPENED(s->thread_info.state)); + + suspend(u); + + break; + } + + case PA_SOURCE_IDLE: + case PA_SOURCE_RUNNING: { + int r; + + if (s->thread_info.state == PA_SOURCE_INIT) { + if (build_pollfd(u) < 0) + /* FIXME: This will cause an assertion failure, because + * with the current design pa_source_put() is not allowed + * to fail and pa_source_put() has no fallback code that + * would start the source suspended if opening the device + * fails. */ + return -PA_ERR_IO; + } + + if (s->thread_info.state == PA_SOURCE_SUSPENDED) { + if ((r = unsuspend(u, false)) < 0) + return r; + } + + break; + } + + case PA_SOURCE_UNLINKED: + case PA_SOURCE_INIT: + case PA_SOURCE_INVALID_STATE: + ; + } + + return 0; +} + +static int ctl_mixer_callback(snd_mixer_elem_t *elem, unsigned int mask) { + struct userdata *u = snd_mixer_elem_get_callback_private(elem); + + pa_assert(u); + pa_assert(u->mixer_handle); + + if (mask == SND_CTL_EVENT_MASK_REMOVE) + return 0; + + if (!PA_SOURCE_IS_LINKED(u->source->state)) + return 0; + + if (u->source->suspend_cause & PA_SUSPEND_SESSION) + return 0; + + if (mask & SND_CTL_EVENT_MASK_VALUE) { + pa_source_get_volume(u->source, true); + pa_source_get_mute(u->source, true); + } + + return 0; +} + +static int io_mixer_callback(snd_mixer_elem_t *elem, unsigned int mask) { + struct userdata *u = snd_mixer_elem_get_callback_private(elem); + + pa_assert(u); + pa_assert(u->mixer_handle); + + if (mask == SND_CTL_EVENT_MASK_REMOVE) + return 0; + + if (u->source->suspend_cause & PA_SUSPEND_SESSION) + return 0; + + if (mask & SND_CTL_EVENT_MASK_VALUE) + pa_source_update_volume_and_mute(u->source); + + return 0; +} + +static void source_get_volume_cb(pa_source *s) { + struct userdata *u = s->userdata; + pa_cvolume r; + char volume_buf[PA_CVOLUME_SNPRINT_VERBOSE_MAX]; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + if (pa_alsa_path_get_volume(u->mixer_path, u->mixer_handle, &s->channel_map, &r) < 0) + return; + + /* Shift down by the base volume, so that 0dB becomes maximum volume */ + pa_sw_cvolume_multiply_scalar(&r, &r, s->base_volume); + + pa_log_debug("Read hardware volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &r, &s->channel_map, u->mixer_path->has_dB)); + + if (pa_cvolume_equal(&u->hardware_volume, &r)) + return; + + s->real_volume = u->hardware_volume = r; + + /* Hmm, so the hardware volume changed, let's reset our software volume */ + if (u->mixer_path->has_dB) + pa_source_set_soft_volume(s, NULL); +} + +static void source_set_volume_cb(pa_source *s) { + struct userdata *u = s->userdata; + pa_cvolume r; + char volume_buf[PA_CVOLUME_SNPRINT_VERBOSE_MAX]; + bool deferred_volume = !!(s->flags & PA_SOURCE_DEFERRED_VOLUME); + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + /* Shift up by the base volume */ + pa_sw_cvolume_divide_scalar(&r, &s->real_volume, s->base_volume); + + if (pa_alsa_path_set_volume(u->mixer_path, u->mixer_handle, &s->channel_map, &r, deferred_volume, !deferred_volume) < 0) + return; + + /* Shift down by the base volume, so that 0dB becomes maximum volume */ + pa_sw_cvolume_multiply_scalar(&r, &r, s->base_volume); + + u->hardware_volume = r; + + if (u->mixer_path->has_dB) { + pa_cvolume new_soft_volume; + bool accurate_enough; + + /* Match exactly what the user requested by software */ + pa_sw_cvolume_divide(&new_soft_volume, &s->real_volume, &u->hardware_volume); + + /* If the adjustment to do in software is only minimal we + * can skip it. That saves us CPU at the expense of a bit of + * accuracy */ + accurate_enough = + (pa_cvolume_min(&new_soft_volume) >= (PA_VOLUME_NORM - VOLUME_ACCURACY)) && + (pa_cvolume_max(&new_soft_volume) <= (PA_VOLUME_NORM + VOLUME_ACCURACY)); + + pa_log_debug("Requested volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &s->real_volume, &s->channel_map, true)); + pa_log_debug("Got hardware volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &u->hardware_volume, &s->channel_map, true)); + pa_log_debug("Calculated software volume: %s (accurate-enough=%s)", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &new_soft_volume, &s->channel_map, true), + pa_yes_no(accurate_enough)); + + if (!accurate_enough) + s->soft_volume = new_soft_volume; + + } else { + pa_log_debug("Wrote hardware volume: %s", + pa_cvolume_snprint_verbose(volume_buf, sizeof(volume_buf), &r, &s->channel_map, false)); + + /* We can't match exactly what the user requested, hence let's + * at least tell the user about it */ + + s->real_volume = r; + } +} + +static void source_write_volume_cb(pa_source *s) { + struct userdata *u = s->userdata; + pa_cvolume hw_vol = s->thread_info.current_hw_volume; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + pa_assert(s->flags & PA_SOURCE_DEFERRED_VOLUME); + + /* Shift up by the base volume */ + pa_sw_cvolume_divide_scalar(&hw_vol, &hw_vol, s->base_volume); + + if (pa_alsa_path_set_volume(u->mixer_path, u->mixer_handle, &s->channel_map, &hw_vol, true, true) < 0) + pa_log_error("Writing HW volume failed"); + else { + pa_cvolume tmp_vol; + bool accurate_enough; + + /* Shift down by the base volume, so that 0dB becomes maximum volume */ + pa_sw_cvolume_multiply_scalar(&hw_vol, &hw_vol, s->base_volume); + + pa_sw_cvolume_divide(&tmp_vol, &hw_vol, &s->thread_info.current_hw_volume); + accurate_enough = + (pa_cvolume_min(&tmp_vol) >= (PA_VOLUME_NORM - VOLUME_ACCURACY)) && + (pa_cvolume_max(&tmp_vol) <= (PA_VOLUME_NORM + VOLUME_ACCURACY)); + + if (!accurate_enough) { + char volume_buf[2][PA_CVOLUME_SNPRINT_VERBOSE_MAX]; + + pa_log_debug("Written HW volume did not match with the request: %s (request) != %s", + pa_cvolume_snprint_verbose(volume_buf[0], + sizeof(volume_buf[0]), + &s->thread_info.current_hw_volume, + &s->channel_map, + true), + pa_cvolume_snprint_verbose(volume_buf[1], sizeof(volume_buf[1]), &hw_vol, &s->channel_map, true)); + } + } +} + +static int source_get_mute_cb(pa_source *s, bool *mute) { + struct userdata *u = s->userdata; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + if (pa_alsa_path_get_mute(u->mixer_path, u->mixer_handle, mute) < 0) + return -1; + + return 0; +} + +static void source_set_mute_cb(pa_source *s) { + struct userdata *u = s->userdata; + + pa_assert(u); + pa_assert(u->mixer_path); + pa_assert(u->mixer_handle); + + pa_alsa_path_set_mute(u->mixer_path, u->mixer_handle, s->muted); +} + +static void mixer_volume_init(struct userdata *u) { + pa_assert(u); + + if (!u->mixer_path || !u->mixer_path->has_volume) { + pa_source_set_write_volume_callback(u->source, NULL); + pa_source_set_get_volume_callback(u->source, NULL); + pa_source_set_set_volume_callback(u->source, NULL); + + pa_log_info("Driver does not support hardware volume control, falling back to software volume control."); + } else { + pa_source_set_get_volume_callback(u->source, source_get_volume_cb); + pa_source_set_set_volume_callback(u->source, source_set_volume_cb); + + if (u->mixer_path->has_dB && u->deferred_volume) { + pa_source_set_write_volume_callback(u->source, source_write_volume_cb); + pa_log_info("Successfully enabled deferred volume."); + } else + pa_source_set_write_volume_callback(u->source, NULL); + + if (u->mixer_path->has_dB) { + pa_source_enable_decibel_volume(u->source, true); + pa_log_info("Hardware volume ranges from %0.2f dB to %0.2f dB.", u->mixer_path->min_dB, u->mixer_path->max_dB); + + u->source->base_volume = pa_sw_volume_from_dB(-u->mixer_path->max_dB); + u->source->n_volume_steps = PA_VOLUME_NORM+1; + + pa_log_info("Fixing base volume to %0.2f dB", pa_sw_volume_to_dB(u->source->base_volume)); + } else { + pa_source_enable_decibel_volume(u->source, false); + pa_log_info("Hardware volume ranges from %li to %li.", u->mixer_path->min_volume, u->mixer_path->max_volume); + + u->source->base_volume = PA_VOLUME_NORM; + u->source->n_volume_steps = u->mixer_path->max_volume - u->mixer_path->min_volume + 1; + } + + pa_log_info("Using hardware volume control. Hardware dB scale %s.", u->mixer_path->has_dB ? "supported" : "not supported"); + } + + if (!u->mixer_path || !u->mixer_path->has_mute) { + pa_source_set_get_mute_callback(u->source, NULL); + pa_source_set_set_mute_callback(u->source, NULL); + pa_log_info("Driver does not support hardware mute control, falling back to software mute control."); + } else { + pa_source_set_get_mute_callback(u->source, source_get_mute_cb); + pa_source_set_set_mute_callback(u->source, source_set_mute_cb); + pa_log_info("Using hardware mute control."); + } +} + +static int source_set_port_ucm_cb(pa_source *s, pa_device_port *p) { + struct userdata *u = s->userdata; + pa_alsa_ucm_port_data *data; + + pa_assert(u); + pa_assert(p); + pa_assert(u->ucm_context); + + data = PA_DEVICE_PORT_DATA(p); + u->mixer_path = data->path; + mixer_volume_init(u); + + if (s->flags & PA_SOURCE_DEFERRED_VOLUME) + pa_asyncmsgq_send(u->source->asyncmsgq, PA_MSGOBJECT(u->source), SOURCE_MESSAGE_SYNC_MIXER, p, 0, NULL); + else + sync_mixer(u, p); + + return pa_alsa_ucm_set_port(u->ucm_context, p, false); +} + +static int source_set_port_cb(pa_source *s, pa_device_port *p) { + struct userdata *u = s->userdata; + pa_alsa_port_data *data; + + pa_assert(u); + pa_assert(p); + pa_assert(u->mixer_handle); + pa_assert(!u->ucm_context); + + data = PA_DEVICE_PORT_DATA(p); + pa_assert_se(u->mixer_path = data->path); + mixer_volume_init(u); + + if (s->flags & PA_SOURCE_DEFERRED_VOLUME) + pa_asyncmsgq_send(u->source->asyncmsgq, PA_MSGOBJECT(u->source), SOURCE_MESSAGE_SYNC_MIXER, p, 0, NULL); + else + sync_mixer(u, p); + + return 0; +} + +static void source_update_requested_latency_cb(pa_source *s) { + struct userdata *u = s->userdata; + pa_assert(u); + pa_assert(u->use_tsched); /* only when timer scheduling is used + * we can dynamically adjust the + * latency */ + + if (!u->pcm_handle) + return; + + update_sw_params(u); +} + +static void source_reconfigure_cb(pa_source *s, pa_sample_spec *spec, bool passthrough) { + struct userdata *u = s->userdata; + int i; + bool format_supported = false; + bool rate_supported = false; + + pa_assert(u); + + for (i = 0; u->supported_formats[i] != PA_SAMPLE_MAX; i++) { + if (u->supported_formats[i] == spec->format) { + pa_source_set_sample_format(u->source, spec->format); + format_supported = true; + break; + } + } + + if (!format_supported) { + pa_log_info("Source does not support sample format of %s, set it to a verified value", + pa_sample_format_to_string(spec->format)); + pa_source_set_sample_format(u->source, u->verified_sample_spec.format); + } + + for (i = 0; u->supported_rates[i]; i++) { + if (u->supported_rates[i] == spec->rate) { + pa_source_set_sample_rate(u->source, spec->rate); + rate_supported = true; + break; + } + } + + if (!rate_supported) { + pa_log_info("Source does not support sample rate of %u, set it to a verfied value", spec->rate); + pa_source_set_sample_rate(u->source, u->verified_sample_spec.rate); + } +} + +static void thread_func(void *userdata) { + struct userdata *u = userdata; + unsigned short revents = 0; + + pa_assert(u); + + pa_log_debug("Thread starting up"); + + if (u->core->realtime_scheduling) + pa_thread_make_realtime(u->core->realtime_priority); + + pa_thread_mq_install(&u->thread_mq); + + for (;;) { + int ret; + pa_usec_t rtpoll_sleep = 0, real_sleep; + +#ifdef DEBUG_TIMING + pa_log_debug("Loop"); +#endif + + /* Read some data and pass it to the sources */ + if (PA_SOURCE_IS_OPENED(u->source->thread_info.state)) { + int work_done; + pa_usec_t sleep_usec = 0; + bool on_timeout = pa_rtpoll_timer_elapsed(u->rtpoll); + + if (u->first) { + pa_log_info("Starting capture."); + snd_pcm_start(u->pcm_handle); + + pa_smoother_resume(u->smoother, pa_rtclock_now(), true); + + u->first = false; + } + + if (u->use_mmap) + work_done = mmap_read(u, &sleep_usec, revents & POLLIN, on_timeout); + else + work_done = unix_read(u, &sleep_usec, revents & POLLIN, on_timeout); + + if (work_done < 0) + goto fail; + +/* pa_log_debug("work_done = %i", work_done); */ + + if (work_done) + update_smoother(u); + + if (u->use_tsched) { + pa_usec_t cusec; + + /* OK, the capture buffer is now empty, let's + * calculate when to wake up next */ + +/* pa_log_debug("Waking up in %0.2fms (sound card clock).", (double) sleep_usec / PA_USEC_PER_MSEC); */ + + /* Convert from the sound card time domain to the + * system time domain */ + cusec = pa_smoother_translate(u->smoother, pa_rtclock_now(), sleep_usec); + +/* pa_log_debug("Waking up in %0.2fms (system clock).", (double) cusec / PA_USEC_PER_MSEC); */ + + /* We don't trust the conversion, so we wake up whatever comes first */ + rtpoll_sleep = PA_MIN(sleep_usec, cusec); + } + } + + if (u->source->flags & PA_SOURCE_DEFERRED_VOLUME) { + pa_usec_t volume_sleep; + pa_source_volume_change_apply(u->source, &volume_sleep); + if (volume_sleep > 0) { + if (rtpoll_sleep > 0) + rtpoll_sleep = PA_MIN(volume_sleep, rtpoll_sleep); + else + rtpoll_sleep = volume_sleep; + } + } + + if (rtpoll_sleep > 0) { + pa_rtpoll_set_timer_relative(u->rtpoll, rtpoll_sleep); + real_sleep = pa_rtclock_now(); + } + else + pa_rtpoll_set_timer_disabled(u->rtpoll); + + /* Hmm, nothing to do. Let's sleep */ + if ((ret = pa_rtpoll_run(u->rtpoll)) < 0) + goto fail; + + if (rtpoll_sleep > 0) { + real_sleep = pa_rtclock_now() - real_sleep; +#ifdef DEBUG_TIMING + pa_log_debug("Expected sleep: %0.2fms, real sleep: %0.2fms (diff %0.2f ms)", + (double) rtpoll_sleep / PA_USEC_PER_MSEC, (double) real_sleep / PA_USEC_PER_MSEC, + (double) ((int64_t) real_sleep - (int64_t) rtpoll_sleep) / PA_USEC_PER_MSEC); +#endif + if (u->use_tsched && real_sleep > rtpoll_sleep + u->tsched_watermark_usec) + pa_log_info("Scheduling delay of %0.2f ms > %0.2f ms, you might want to investigate this to improve latency...", + (double) (real_sleep - rtpoll_sleep) / PA_USEC_PER_MSEC, + (double) (u->tsched_watermark_usec) / PA_USEC_PER_MSEC); + } + + if (u->source->flags & PA_SOURCE_DEFERRED_VOLUME) + pa_source_volume_change_apply(u->source, NULL); + + if (ret == 0) + goto finish; + + /* Tell ALSA about this and process its response */ + if (PA_SOURCE_IS_OPENED(u->source->thread_info.state)) { + struct pollfd *pollfd; + int err; + unsigned n; + + pollfd = pa_rtpoll_item_get_pollfd(u->alsa_rtpoll_item, &n); + + if ((err = snd_pcm_poll_descriptors_revents(u->pcm_handle, pollfd, n, &revents)) < 0) { + pa_log("snd_pcm_poll_descriptors_revents() failed: %s", pa_alsa_strerror(err)); + goto fail; + } + + if (revents & ~POLLIN) { + if ((err = pa_alsa_recover_from_poll(u->pcm_handle, revents)) < 0) + goto fail; + + /* Stream needs to be restarted */ + if (err == 1) { + close_pcm(u); + if (unsuspend(u, true) < 0) + goto fail; + } else + reset_vars(u); + + revents = 0; + } else if (revents && u->use_tsched && pa_log_ratelimit(PA_LOG_DEBUG)) + pa_log_debug("Wakeup from ALSA!"); + + } else + revents = 0; + } + +fail: + /* If this was no regular exit from the loop we have to continue + * processing messages until we received PA_MESSAGE_SHUTDOWN */ + pa_asyncmsgq_post(u->thread_mq.outq, PA_MSGOBJECT(u->core), PA_CORE_MESSAGE_UNLOAD_MODULE, u->module, 0, NULL, NULL); + pa_asyncmsgq_wait_for(u->thread_mq.inq, PA_MESSAGE_SHUTDOWN); + +finish: + pa_log_debug("Thread shutting down"); +} + +static void set_source_name(pa_source_new_data *data, pa_modargs *ma, const char *device_id, const char *device_name, pa_alsa_mapping *mapping) { + const char *n; + char *t; + + pa_assert(data); + pa_assert(ma); + pa_assert(device_name); + + if ((n = pa_modargs_get_value(ma, "source_name", NULL))) { + pa_source_new_data_set_name(data, n); + data->namereg_fail = true; + return; + } + + if ((n = pa_modargs_get_value(ma, "name", NULL))) + data->namereg_fail = true; + else { + n = device_id ? device_id : device_name; + data->namereg_fail = false; + } + + if (mapping) + t = pa_sprintf_malloc("alsa_input.%s.%s", n, mapping->name); + else + t = pa_sprintf_malloc("alsa_input.%s", n); + + pa_source_new_data_set_name(data, t); + pa_xfree(t); +} + +static void find_mixer(struct userdata *u, pa_alsa_mapping *mapping, const char *element, bool ignore_dB) { + const char *mdev; + + if (!mapping && !element) + return; + + if (!element && mapping && pa_alsa_path_set_is_empty(mapping->input_path_set)) + return; + + u->mixers = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, + NULL, (pa_free_cb_t) pa_alsa_mixer_free); + + mdev = pa_proplist_gets(mapping->proplist, "alsa.mixer_device"); + if (mdev) { + u->mixer_handle = pa_alsa_open_mixer_by_name(u->mixers, mdev, false); + } else { + u->mixer_handle = pa_alsa_open_mixer_for_pcm(u->mixers, u->pcm_handle, false); + } + if (!u->mixer_handle) { + pa_log_info("Failed to find a working mixer device."); + return; + } + + if (element) { + + if (!(u->mixer_path = pa_alsa_path_synthesize(element, PA_ALSA_DIRECTION_INPUT))) + goto fail; + + if (pa_alsa_path_probe(u->mixer_path, NULL, u->mixer_handle, ignore_dB) < 0) + goto fail; + + pa_log_debug("Probed mixer path %s:", u->mixer_path->name); + pa_alsa_path_dump(u->mixer_path); + } else { + u->mixer_path_set = mapping->input_path_set; + } + + return; + +fail: + + if (u->mixer_path) { + pa_alsa_path_free(u->mixer_path); + u->mixer_path = NULL; + } + + u->mixer_handle = NULL; + pa_hashmap_free(u->mixers); + u->mixers = NULL; +} + +static int setup_mixer(struct userdata *u, bool ignore_dB) { + bool need_mixer_callback = false; + + pa_assert(u); + + /* This code is before the u->mixer_handle check, because if the UCM + * configuration doesn't specify volume or mute controls, u->mixer_handle + * will be NULL, but the UCM device enable sequence will still need to be + * executed. */ + if (u->source->active_port && u->ucm_context) { + if (pa_alsa_ucm_set_port(u->ucm_context, u->source->active_port, false) < 0) + return -1; + } + + if (!u->mixer_handle) + return 0; + + if (u->source->active_port) { + if (!u->ucm_context) { + pa_alsa_port_data *data; + + /* We have a list of supported paths, so let's activate the + * one that has been chosen as active */ + + data = PA_DEVICE_PORT_DATA(u->source->active_port); + u->mixer_path = data->path; + + pa_alsa_path_select(data->path, data->setting, u->mixer_handle, u->source->muted); + } else { + pa_alsa_ucm_port_data *data; + + data = PA_DEVICE_PORT_DATA(u->source->active_port); + + /* Now activate volume controls, if any */ + if (data->path) { + u->mixer_path = data->path; + pa_alsa_path_select(u->mixer_path, NULL, u->mixer_handle, u->source->muted); + } + } + } else { + + if (!u->mixer_path && u->mixer_path_set) + u->mixer_path = pa_hashmap_first(u->mixer_path_set->paths); + + if (u->mixer_path) { + /* Hmm, we have only a single path, then let's activate it */ + + pa_alsa_path_select(u->mixer_path, u->mixer_path->settings, u->mixer_handle, u->source->muted); + } else + return 0; + } + + mixer_volume_init(u); + + /* Will we need to register callbacks? */ + if (u->mixer_path_set && u->mixer_path_set->paths) { + pa_alsa_path *p; + void *state; + + PA_HASHMAP_FOREACH(p, u->mixer_path_set->paths, state) { + if (p->has_volume || p->has_mute) + need_mixer_callback = true; + } + } + else if (u->mixer_path) + need_mixer_callback = u->mixer_path->has_volume || u->mixer_path->has_mute; + + if (need_mixer_callback) { + int (*mixer_callback)(snd_mixer_elem_t *, unsigned int); + if (u->source->flags & PA_SOURCE_DEFERRED_VOLUME) { + u->mixer_pd = pa_alsa_mixer_pdata_new(); + mixer_callback = io_mixer_callback; + + if (pa_alsa_set_mixer_rtpoll(u->mixer_pd, u->mixer_handle, u->rtpoll) < 0) { + pa_log("Failed to initialize file descriptor monitoring"); + return -1; + } + } else { + u->mixer_fdl = pa_alsa_fdlist_new(); + mixer_callback = ctl_mixer_callback; + + if (pa_alsa_fdlist_set_handle(u->mixer_fdl, u->mixer_handle, NULL, u->core->mainloop) < 0) { + pa_log("Failed to initialize file descriptor monitoring"); + return -1; + } + } + + if (u->mixer_path_set) + pa_alsa_path_set_set_callback(u->mixer_path_set, u->mixer_handle, mixer_callback, u); + else + pa_alsa_path_set_callback(u->mixer_path, u->mixer_handle, mixer_callback, u); + } + + return 0; +} + +pa_source *pa_alsa_source_new(pa_module *m, pa_modargs *ma, const char*driver, pa_card *card, pa_alsa_mapping *mapping) { + + struct userdata *u = NULL; + const char *dev_id = NULL, *key, *mod_name; + pa_sample_spec ss; + char *thread_name = NULL; + uint32_t alternate_sample_rate; + pa_channel_map map; + uint32_t nfrags, frag_size, buffer_size, tsched_size, tsched_watermark; + snd_pcm_uframes_t period_frames, buffer_frames, tsched_frames; + size_t frame_size; + bool use_mmap = true; + bool use_tsched = true; + bool ignore_dB = false; + bool namereg_fail = false; + bool deferred_volume = false; + bool fixed_latency_range = false; + bool b; + bool d; + bool avoid_resampling; + pa_source_new_data data; + bool volume_is_set; + bool mute_is_set; + pa_alsa_profile_set *profile_set = NULL; + void *state; + + pa_assert(m); + pa_assert(ma); + + ss = m->core->default_sample_spec; + map = m->core->default_channel_map; + avoid_resampling = m->core->avoid_resampling; + + /* Pick sample spec overrides from the mapping, if any */ + if (mapping) { + if (mapping->sample_spec.format != PA_SAMPLE_INVALID) + ss.format = mapping->sample_spec.format; + if (mapping->sample_spec.rate != 0) + ss.rate = mapping->sample_spec.rate; + if (mapping->sample_spec.channels != 0) { + ss.channels = mapping->sample_spec.channels; + if (pa_channel_map_valid(&mapping->channel_map)) + pa_assert(pa_channel_map_compatible(&mapping->channel_map, &ss)); + } + } + + /* Override with modargs if provided */ + if (pa_modargs_get_sample_spec_and_channel_map(ma, &ss, &map, PA_CHANNEL_MAP_ALSA) < 0) { + pa_log("Failed to parse sample specification and channel map"); + goto fail; + } + + alternate_sample_rate = m->core->alternate_sample_rate; + if (pa_modargs_get_alternate_sample_rate(ma, &alternate_sample_rate) < 0) { + pa_log("Failed to parse alternate sample rate"); + goto fail; + } + + frame_size = pa_frame_size(&ss); + + nfrags = m->core->default_n_fragments; + frag_size = (uint32_t) pa_usec_to_bytes(m->core->default_fragment_size_msec*PA_USEC_PER_MSEC, &ss); + if (frag_size <= 0) + frag_size = (uint32_t) frame_size; + tsched_size = (uint32_t) pa_usec_to_bytes(DEFAULT_TSCHED_BUFFER_USEC, &ss); + tsched_watermark = (uint32_t) pa_usec_to_bytes(DEFAULT_TSCHED_WATERMARK_USEC, &ss); + + if (pa_modargs_get_value_u32(ma, "fragments", &nfrags) < 0 || + pa_modargs_get_value_u32(ma, "fragment_size", &frag_size) < 0 || + pa_modargs_get_value_u32(ma, "tsched_buffer_size", &tsched_size) < 0 || + pa_modargs_get_value_u32(ma, "tsched_buffer_watermark", &tsched_watermark) < 0) { + pa_log("Failed to parse buffer metrics"); + goto fail; + } + + buffer_size = nfrags * frag_size; + + period_frames = frag_size/frame_size; + buffer_frames = buffer_size/frame_size; + tsched_frames = tsched_size/frame_size; + + if (pa_modargs_get_value_boolean(ma, "mmap", &use_mmap) < 0) { + pa_log("Failed to parse mmap argument."); + goto fail; + } + + if (pa_modargs_get_value_boolean(ma, "tsched", &use_tsched) < 0) { + pa_log("Failed to parse tsched argument."); + goto fail; + } + + if (pa_modargs_get_value_boolean(ma, "ignore_dB", &ignore_dB) < 0) { + pa_log("Failed to parse ignore_dB argument."); + goto fail; + } + + deferred_volume = m->core->deferred_volume; + if (pa_modargs_get_value_boolean(ma, "deferred_volume", &deferred_volume) < 0) { + pa_log("Failed to parse deferred_volume argument."); + goto fail; + } + + if (pa_modargs_get_value_boolean(ma, "fixed_latency_range", &fixed_latency_range) < 0) { + pa_log("Failed to parse fixed_latency_range argument."); + goto fail; + } + + use_tsched = pa_alsa_may_tsched(use_tsched); + + u = pa_xnew0(struct userdata, 1); + u->core = m->core; + u->module = m; + u->use_mmap = use_mmap; + u->use_tsched = use_tsched; + u->tsched_size = tsched_size; + u->initial_info.nfrags = (size_t) nfrags; + u->initial_info.fragment_size = (size_t) frag_size; + u->initial_info.tsched_size = (size_t) tsched_size; + u->initial_info.tsched_watermark = (size_t) tsched_watermark; + u->deferred_volume = deferred_volume; + u->fixed_latency_range = fixed_latency_range; + u->first = true; + u->rtpoll = pa_rtpoll_new(); + + if (pa_thread_mq_init(&u->thread_mq, m->core->mainloop, u->rtpoll) < 0) { + pa_log("pa_thread_mq_init() failed."); + goto fail; + } + + u->smoother = pa_smoother_new( + SMOOTHER_ADJUST_USEC, + SMOOTHER_WINDOW_USEC, + true, + true, + 5, + pa_rtclock_now(), + true); + u->smoother_interval = SMOOTHER_MIN_INTERVAL; + + /* use ucm */ + if (mapping && mapping->ucm_context.ucm) + u->ucm_context = &mapping->ucm_context; + + dev_id = pa_modargs_get_value( + ma, "device_id", + pa_modargs_get_value(ma, "device", DEFAULT_DEVICE)); + + u->paths_dir = pa_xstrdup(pa_modargs_get_value(ma, "paths_dir", NULL)); + + if (reserve_init(u, dev_id) < 0) + goto fail; + + if (reserve_monitor_init(u, dev_id) < 0) + goto fail; + + b = use_mmap; + d = use_tsched; + + /* Force ALSA to reread its configuration if module-alsa-card didn't + * do it for us. This matters if our device was hot-plugged after ALSA + * has already read its configuration - see + * https://bugs.freedesktop.org/show_bug.cgi?id=54029 + */ + + if (!card) + snd_config_update_free_global(); + + if (mapping) { + + if (!(dev_id = pa_modargs_get_value(ma, "device_id", NULL))) { + pa_log("device_id= not set"); + goto fail; + } + + if ((mod_name = pa_proplist_gets(mapping->proplist, PA_ALSA_PROP_UCM_MODIFIER))) { + if (snd_use_case_set(u->ucm_context->ucm->ucm_mgr, "_enamod", mod_name) < 0) + pa_log("Failed to enable ucm modifier %s", mod_name); + else + pa_log_debug("Enabled ucm modifier %s", mod_name); + } + + if (!(u->pcm_handle = pa_alsa_open_by_device_id_mapping( + dev_id, + &u->device_name, + &ss, &map, + SND_PCM_STREAM_CAPTURE, + &period_frames, &buffer_frames, tsched_frames, + &b, &d, mapping))) + goto fail; + + } else if ((dev_id = pa_modargs_get_value(ma, "device_id", NULL))) { + + if (!(profile_set = pa_alsa_profile_set_new(NULL, &map))) + goto fail; + + if (!(u->pcm_handle = pa_alsa_open_by_device_id_auto( + dev_id, + &u->device_name, + &ss, &map, + SND_PCM_STREAM_CAPTURE, + &period_frames, &buffer_frames, tsched_frames, + &b, &d, profile_set, &mapping))) + goto fail; + + } else { + + if (!(u->pcm_handle = pa_alsa_open_by_device_string( + pa_modargs_get_value(ma, "device", DEFAULT_DEVICE), + &u->device_name, + &ss, &map, + SND_PCM_STREAM_CAPTURE, + &period_frames, &buffer_frames, tsched_frames, + &b, &d, false))) + goto fail; + } + + pa_assert(u->device_name); + pa_log_info("Successfully opened device %s.", u->device_name); + + if (pa_alsa_pcm_is_modem(u->pcm_handle)) { + pa_log_notice("Device %s is modem, refusing further initialization.", u->device_name); + goto fail; + } + + if (mapping) + pa_log_info("Selected mapping '%s' (%s).", mapping->description, mapping->name); + + if (use_mmap && !b) { + pa_log_info("Device doesn't support mmap(), falling back to UNIX read/write mode."); + u->use_mmap = use_mmap = false; + } + + if (use_tsched && (!b || !d)) { + pa_log_info("Cannot enable timer-based scheduling, falling back to sound IRQ scheduling."); + u->use_tsched = use_tsched = false; + } + + if (u->use_mmap) + pa_log_info("Successfully enabled mmap() mode."); + + if (u->use_tsched) { + pa_log_info("Successfully enabled timer-based scheduling mode."); + if (u->fixed_latency_range) + pa_log_info("Disabling latency range changes on overrun"); + } + + u->verified_sample_spec = ss; + + u->supported_formats = pa_alsa_get_supported_formats(u->pcm_handle, ss.format); + if (!u->supported_formats) { + pa_log_error("Failed to find any supported sample formats."); + goto fail; + } + + u->supported_rates = pa_alsa_get_supported_rates(u->pcm_handle, ss.rate); + if (!u->supported_rates) { + pa_log_error("Failed to find any supported sample rates."); + goto fail; + } + + /* ALSA might tweak the sample spec, so recalculate the frame size */ + frame_size = pa_frame_size(&ss); + + pa_source_new_data_init(&data); + data.driver = driver; + data.module = m; + data.card = card; + set_source_name(&data, ma, dev_id, u->device_name, mapping); + + /* We need to give pa_modargs_get_value_boolean() a pointer to a local + * variable instead of using &data.namereg_fail directly, because + * data.namereg_fail is a bitfield and taking the address of a bitfield + * variable is impossible. */ + namereg_fail = data.namereg_fail; + if (pa_modargs_get_value_boolean(ma, "namereg_fail", &namereg_fail) < 0) { + pa_log("Failed to parse namereg_fail argument."); + pa_source_new_data_done(&data); + goto fail; + } + data.namereg_fail = namereg_fail; + + if (pa_modargs_get_value_boolean(ma, "avoid_resampling", &avoid_resampling) < 0) { + pa_log("Failed to parse avoid_resampling argument."); + pa_source_new_data_done(&data); + goto fail; + } + pa_source_new_data_set_avoid_resampling(&data, avoid_resampling); + + pa_source_new_data_set_sample_spec(&data, &ss); + pa_source_new_data_set_channel_map(&data, &map); + pa_source_new_data_set_alternate_sample_rate(&data, alternate_sample_rate); + + pa_alsa_init_proplist_pcm(m->core, data.proplist, u->pcm_handle); + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_STRING, u->device_name); + pa_proplist_setf(data.proplist, PA_PROP_DEVICE_BUFFERING_BUFFER_SIZE, "%lu", (unsigned long) (buffer_frames * frame_size)); + pa_proplist_setf(data.proplist, PA_PROP_DEVICE_BUFFERING_FRAGMENT_SIZE, "%lu", (unsigned long) (period_frames * frame_size)); + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_ACCESS_MODE, u->use_tsched ? "mmap+timer" : (u->use_mmap ? "mmap" : "serial")); + + if (mapping) { + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_PROFILE_NAME, mapping->name); + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_PROFILE_DESCRIPTION, mapping->description); + + state = NULL; + while ((key = pa_proplist_iterate(mapping->proplist, &state))) + pa_proplist_sets(data.proplist, key, pa_proplist_gets(mapping->proplist, key)); + } + + pa_alsa_init_description(data.proplist, card); + + if (u->control_device) + pa_alsa_init_proplist_ctl(data.proplist, u->control_device); + + if (pa_modargs_get_proplist(ma, "source_properties", data.proplist, PA_UPDATE_REPLACE) < 0) { + pa_log("Invalid properties"); + pa_source_new_data_done(&data); + goto fail; + } + + if (u->ucm_context) { + pa_alsa_ucm_add_ports(&data.ports, data.proplist, u->ucm_context, false, card, u->pcm_handle, ignore_dB); + find_mixer(u, mapping, pa_modargs_get_value(ma, "control", NULL), ignore_dB); + } else { + find_mixer(u, mapping, pa_modargs_get_value(ma, "control", NULL), ignore_dB); + if (u->mixer_path_set) + pa_alsa_add_ports(&data, u->mixer_path_set, card); + } + + u->source = pa_source_new(m->core, &data, PA_SOURCE_HARDWARE|PA_SOURCE_LATENCY|(u->use_tsched ? PA_SOURCE_DYNAMIC_LATENCY : 0)); + volume_is_set = data.volume_is_set; + mute_is_set = data.muted_is_set; + pa_source_new_data_done(&data); + + if (!u->source) { + pa_log("Failed to create source object"); + goto fail; + } + + if (u->ucm_context) { + pa_device_port *port; + unsigned h_prio = 0; + PA_HASHMAP_FOREACH(port, u->source->ports, state) { + if (!h_prio || port->priority > h_prio) + h_prio = port->priority; + } + /* ucm ports prioriy is 100, 200, ..., 900, change it to units digit */ + h_prio = h_prio / 100; + u->source->priority += h_prio; + } + + if (pa_modargs_get_value_u32(ma, "deferred_volume_safety_margin", + &u->source->thread_info.volume_change_safety_margin) < 0) { + pa_log("Failed to parse deferred_volume_safety_margin parameter"); + goto fail; + } + + if (pa_modargs_get_value_s32(ma, "deferred_volume_extra_delay", + &u->source->thread_info.volume_change_extra_delay) < 0) { + pa_log("Failed to parse deferred_volume_extra_delay parameter"); + goto fail; + } + + u->source->parent.process_msg = source_process_msg; + if (u->use_tsched) + u->source->update_requested_latency = source_update_requested_latency_cb; + u->source->set_state_in_main_thread = source_set_state_in_main_thread_cb; + u->source->set_state_in_io_thread = source_set_state_in_io_thread_cb; + if (u->ucm_context) + u->source->set_port = source_set_port_ucm_cb; + else + u->source->set_port = source_set_port_cb; + u->source->reconfigure = source_reconfigure_cb; + u->source->userdata = u; + + pa_source_set_asyncmsgq(u->source, u->thread_mq.inq); + pa_source_set_rtpoll(u->source, u->rtpoll); + + u->frame_size = frame_size; + u->frames_per_block = pa_mempool_block_size_max(m->core->mempool) / frame_size; + u->fragment_size = frag_size = (size_t) (period_frames * frame_size); + u->hwbuf_size = buffer_size = (size_t) (buffer_frames * frame_size); + pa_cvolume_mute(&u->hardware_volume, u->source->sample_spec.channels); + + pa_log_info("Using %0.1f fragments of size %lu bytes (%0.2fms), buffer size is %lu bytes (%0.2fms)", + (double) u->hwbuf_size / (double) u->fragment_size, + (long unsigned) u->fragment_size, + (double) pa_bytes_to_usec(u->fragment_size, &ss) / PA_USEC_PER_MSEC, + (long unsigned) u->hwbuf_size, + (double) pa_bytes_to_usec(u->hwbuf_size, &ss) / PA_USEC_PER_MSEC); + + if (u->use_tsched) { + u->tsched_watermark_ref = tsched_watermark; + reset_watermark(u, u->tsched_watermark_ref, &ss, false); + } + else + pa_source_set_fixed_latency(u->source, pa_bytes_to_usec(u->hwbuf_size, &ss)); + + reserve_update(u); + + if (update_sw_params(u) < 0) + goto fail; + + if (setup_mixer(u, ignore_dB) < 0) + goto fail; + + pa_alsa_dump(PA_LOG_DEBUG, u->pcm_handle); + + thread_name = pa_sprintf_malloc("alsa-source-%s", pa_strnull(pa_proplist_gets(u->source->proplist, "alsa.id"))); + if (!(u->thread = pa_thread_new(thread_name, thread_func, u))) { + pa_log("Failed to create thread."); + goto fail; + } + pa_xfree(thread_name); + thread_name = NULL; + + /* Get initial mixer settings */ + if (volume_is_set) { + if (u->source->set_volume) + u->source->set_volume(u->source); + } else { + if (u->source->get_volume) + u->source->get_volume(u->source); + } + + if (mute_is_set) { + if (u->source->set_mute) + u->source->set_mute(u->source); + } else { + if (u->source->get_mute) { + bool mute; + + if (u->source->get_mute(u->source, &mute) >= 0) + pa_source_set_mute(u->source, mute, false); + } + } + + if ((volume_is_set || mute_is_set) && u->source->write_volume) + u->source->write_volume(u->source); + + pa_source_put(u->source); + + if (profile_set) + pa_alsa_profile_set_free(profile_set); + + return u->source; + +fail: + pa_xfree(thread_name); + + if (u) + userdata_free(u); + + if (profile_set) + pa_alsa_profile_set_free(profile_set); + + return NULL; +} + +static void userdata_free(struct userdata *u) { + pa_assert(u); + + if (u->source) + pa_source_unlink(u->source); + + if (u->thread) { + pa_asyncmsgq_send(u->thread_mq.inq, NULL, PA_MESSAGE_SHUTDOWN, NULL, 0, NULL); + pa_thread_free(u->thread); + } + + pa_thread_mq_done(&u->thread_mq); + + if (u->source) + pa_source_unref(u->source); + + if (u->mixer_pd) + pa_alsa_mixer_pdata_free(u->mixer_pd); + + if (u->alsa_rtpoll_item) + pa_rtpoll_item_free(u->alsa_rtpoll_item); + + if (u->rtpoll) + pa_rtpoll_free(u->rtpoll); + + if (u->pcm_handle) { + snd_pcm_drop(u->pcm_handle); + snd_pcm_close(u->pcm_handle); + } + + if (u->mixer_fdl) + pa_alsa_fdlist_free(u->mixer_fdl); + + /* Only free the mixer_path if the sink owns it */ + if (u->mixer_path && !u->mixer_path_set && !u->ucm_context) + pa_alsa_path_free(u->mixer_path); + + if (u->mixers) + pa_hashmap_free(u->mixers); + + if (u->smoother) + pa_smoother_free(u->smoother); + + if (u->supported_formats) + pa_xfree(u->supported_formats); + + if (u->supported_rates) + pa_xfree(u->supported_rates); + + reserve_done(u); + monitor_done(u); + + pa_xfree(u->device_name); + pa_xfree(u->control_device); + pa_xfree(u->paths_dir); + pa_xfree(u); +} + +void pa_alsa_source_free(pa_source *s) { + struct userdata *u; + + pa_source_assert_ref(s); + pa_assert_se(u = s->userdata); + + userdata_free(u); +} diff --git a/src/modules/alsa/alsa-source.h b/src/modules/alsa/alsa-source.h new file mode 100644 index 0000000..ecbdfcd --- /dev/null +++ b/src/modules/alsa/alsa-source.h @@ -0,0 +1,34 @@ +#ifndef fooalsasourcehfoo +#define fooalsasourcehfoo + +/*** + This file is part of PulseAudio. + + Copyright 2004-2006 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#include <pulsecore/module.h> +#include <pulsecore/modargs.h> +#include <pulsecore/source.h> + +#include "alsa-util.h" + +pa_source* pa_alsa_source_new(pa_module *m, pa_modargs *ma, const char*driver, pa_card *card, pa_alsa_mapping *mapping); + +void pa_alsa_source_free(pa_source *s); + +#endif diff --git a/src/modules/alsa/alsa-ucm.c b/src/modules/alsa/alsa-ucm.c new file mode 100644 index 0000000..d9cea61 --- /dev/null +++ b/src/modules/alsa/alsa-ucm.c @@ -0,0 +1,2396 @@ +/*** + This file is part of PulseAudio. + + Copyright 2011 Wolfson Microelectronics PLC + Author Margarita Olaya <magi@slimlogic.co.uk> + Copyright 2012 Feng Wei <wei.feng@freescale.com>, Freescale Ltd. + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <ctype.h> +#include <sys/types.h> +#include <limits.h> +#include <alsa/asoundlib.h> + +#ifdef HAVE_VALGRIND_MEMCHECK_H +#include <valgrind/memcheck.h> +#endif + +#include <pulse/sample.h> +#include <pulse/xmalloc.h> +#include <pulse/timeval.h> +#include <pulse/util.h> + +#include <pulsecore/log.h> +#include <pulsecore/macro.h> +#include <pulsecore/core-util.h> +#include <pulsecore/atomic.h> +#include <pulsecore/core-error.h> +#include <pulsecore/once.h> +#include <pulsecore/thread.h> +#include <pulsecore/conf-parser.h> +#include <pulsecore/strbuf.h> + +#include "alsa-mixer.h" +#include "alsa-util.h" +#include "alsa-ucm.h" + +#define PA_UCM_PRE_TAG_OUTPUT "[Out] " +#define PA_UCM_PRE_TAG_INPUT "[In] " + +#define PA_UCM_PLAYBACK_PRIORITY_UNSET(device) ((device)->playback_channels && !(device)->playback_priority) +#define PA_UCM_CAPTURE_PRIORITY_UNSET(device) ((device)->capture_channels && !(device)->capture_priority) +#define PA_UCM_DEVICE_PRIORITY_SET(device, priority) \ + do { \ + if (PA_UCM_PLAYBACK_PRIORITY_UNSET(device)) (device)->playback_priority = (priority); \ + if (PA_UCM_CAPTURE_PRIORITY_UNSET(device)) (device)->capture_priority = (priority); \ + } while (0) +#define PA_UCM_IS_MODIFIER_MAPPING(m) ((pa_proplist_gets((m)->proplist, PA_ALSA_PROP_UCM_MODIFIER)) != NULL) + +#ifdef HAVE_ALSA_UCM + +struct ucm_type { + const char *prefix; + pa_device_port_type_t type; +}; + +struct ucm_items { + const char *id; + const char *property; +}; + +struct ucm_info { + const char *id; + unsigned priority; +}; + +static pa_alsa_jack* ucm_get_jack(pa_alsa_ucm_config *ucm, pa_alsa_ucm_device *device); +static void device_set_jack(pa_alsa_ucm_device *device, pa_alsa_jack *jack); +static void device_add_hw_mute_jack(pa_alsa_ucm_device *device, pa_alsa_jack *jack); + +static pa_alsa_ucm_device *verb_find_device(pa_alsa_ucm_verb *verb, const char *device_name); + + +static void ucm_port_data_init(pa_alsa_ucm_port_data *port, pa_alsa_ucm_config *ucm, pa_device_port *core_port, + pa_alsa_ucm_device **devices, unsigned n_devices); +static void ucm_port_data_free(pa_device_port *port); +static void ucm_port_update_available(pa_alsa_ucm_port_data *port); + +static struct ucm_type types[] = { + {"None", PA_DEVICE_PORT_TYPE_UNKNOWN}, + {"Speaker", PA_DEVICE_PORT_TYPE_SPEAKER}, + {"Line", PA_DEVICE_PORT_TYPE_LINE}, + {"Mic", PA_DEVICE_PORT_TYPE_MIC}, + {"Headphones", PA_DEVICE_PORT_TYPE_HEADPHONES}, + {"Headset", PA_DEVICE_PORT_TYPE_HEADSET}, + {"Handset", PA_DEVICE_PORT_TYPE_HANDSET}, + {"Bluetooth", PA_DEVICE_PORT_TYPE_BLUETOOTH}, + {"Earpiece", PA_DEVICE_PORT_TYPE_EARPIECE}, + {"SPDIF", PA_DEVICE_PORT_TYPE_SPDIF}, + {"HDMI", PA_DEVICE_PORT_TYPE_HDMI}, + {NULL, 0} +}; + +static struct ucm_items item[] = { + {"PlaybackPCM", PA_ALSA_PROP_UCM_SINK}, + {"CapturePCM", PA_ALSA_PROP_UCM_SOURCE}, + {"PlaybackCTL", PA_ALSA_PROP_UCM_PLAYBACK_CTL_DEVICE}, + {"PlaybackVolume", PA_ALSA_PROP_UCM_PLAYBACK_VOLUME}, + {"PlaybackSwitch", PA_ALSA_PROP_UCM_PLAYBACK_SWITCH}, + {"PlaybackMixer", PA_ALSA_PROP_UCM_PLAYBACK_MIXER_DEVICE}, + {"PlaybackMixerElem", PA_ALSA_PROP_UCM_PLAYBACK_MIXER_ELEM}, + {"PlaybackMasterElem", PA_ALSA_PROP_UCM_PLAYBACK_MASTER_ELEM}, + {"PlaybackMasterType", PA_ALSA_PROP_UCM_PLAYBACK_MASTER_TYPE}, + {"PlaybackPriority", PA_ALSA_PROP_UCM_PLAYBACK_PRIORITY}, + {"PlaybackRate", PA_ALSA_PROP_UCM_PLAYBACK_RATE}, + {"PlaybackChannels", PA_ALSA_PROP_UCM_PLAYBACK_CHANNELS}, + {"CaptureCTL", PA_ALSA_PROP_UCM_CAPTURE_CTL_DEVICE}, + {"CaptureVolume", PA_ALSA_PROP_UCM_CAPTURE_VOLUME}, + {"CaptureSwitch", PA_ALSA_PROP_UCM_CAPTURE_SWITCH}, + {"CaptureMixer", PA_ALSA_PROP_UCM_CAPTURE_MIXER_DEVICE}, + {"CaptureMixerElem", PA_ALSA_PROP_UCM_CAPTURE_MIXER_ELEM}, + {"CaptureMasterElem", PA_ALSA_PROP_UCM_CAPTURE_MASTER_ELEM}, + {"CaptureMasterType", PA_ALSA_PROP_UCM_CAPTURE_MASTER_TYPE}, + {"CapturePriority", PA_ALSA_PROP_UCM_CAPTURE_PRIORITY}, + {"CaptureRate", PA_ALSA_PROP_UCM_CAPTURE_RATE}, + {"CaptureChannels", PA_ALSA_PROP_UCM_CAPTURE_CHANNELS}, + {"TQ", PA_ALSA_PROP_UCM_QOS}, + {"JackCTL", PA_ALSA_PROP_UCM_JACK_DEVICE}, + {"JackControl", PA_ALSA_PROP_UCM_JACK_CONTROL}, + {"JackHWMute", PA_ALSA_PROP_UCM_JACK_HW_MUTE}, + {NULL, NULL}, +}; + +/* UCM verb info - this should eventually be part of policy manangement */ +static struct ucm_info verb_info[] = { + {SND_USE_CASE_VERB_INACTIVE, 0}, + {SND_USE_CASE_VERB_HIFI, 8000}, + {SND_USE_CASE_VERB_HIFI_LOW_POWER, 7000}, + {SND_USE_CASE_VERB_VOICE, 6000}, + {SND_USE_CASE_VERB_VOICE_LOW_POWER, 5000}, + {SND_USE_CASE_VERB_VOICECALL, 4000}, + {SND_USE_CASE_VERB_IP_VOICECALL, 4000}, + {SND_USE_CASE_VERB_ANALOG_RADIO, 3000}, + {SND_USE_CASE_VERB_DIGITAL_RADIO, 3000}, + {NULL, 0} +}; + +/* UCM device info - should be overwritten by ucm property */ +static struct ucm_info dev_info[] = { + {SND_USE_CASE_DEV_SPEAKER, 100}, + {SND_USE_CASE_DEV_LINE, 100}, + {SND_USE_CASE_DEV_HEADPHONES, 100}, + {SND_USE_CASE_DEV_HEADSET, 300}, + {SND_USE_CASE_DEV_HANDSET, 200}, + {SND_USE_CASE_DEV_BLUETOOTH, 400}, + {SND_USE_CASE_DEV_EARPIECE, 100}, + {SND_USE_CASE_DEV_SPDIF, 100}, + {SND_USE_CASE_DEV_HDMI, 100}, + {SND_USE_CASE_DEV_NONE, 100}, + {NULL, 0} +}; + + +static char *ucm_verb_value( + snd_use_case_mgr_t *uc_mgr, + const char *verb_name, + const char *id) { + + const char *value; + char *_id = pa_sprintf_malloc("=%s//%s", id, verb_name); + int err = snd_use_case_get(uc_mgr, _id, &value); + pa_xfree(_id); + if (err < 0) + return NULL; + pa_log_debug("Got %s for verb %s: %s", id, verb_name, value); + /* Use the cast here to allow free() call without casting for callers. + * The snd_use_case_get() returns mallocated string. + * See the Note: in use-case.h for snd_use_case_get(). + */ + return (char *)value; +} + +static int ucm_device_exists(pa_idxset *idxset, pa_alsa_ucm_device *dev) { + pa_alsa_ucm_device *d; + uint32_t idx; + + PA_IDXSET_FOREACH(d, idxset, idx) + if (d == dev) + return 1; + + return 0; +} + +static void ucm_add_devices_to_idxset( + pa_idxset *idxset, + pa_alsa_ucm_device *me, + pa_alsa_ucm_device *devices, + const char **dev_names, + int n) { + + pa_alsa_ucm_device *d; + + PA_LLIST_FOREACH(d, devices) { + const char *name; + int i; + + if (d == me) + continue; + + name = pa_proplist_gets(d->proplist, PA_ALSA_PROP_UCM_NAME); + + for (i = 0; i < n; i++) + if (pa_streq(dev_names[i], name)) + pa_idxset_put(idxset, d, NULL); + } +} + +/* Split a string into words. Like pa_split_spaces() but handle '' and "". */ +static char *ucm_split_devnames(const char *c, const char **state) { + const char *current = *state ? *state : c; + char h; + size_t l; + + if (!*current || *c == 0) + return NULL; + + current += strspn(current, "\n\r \t"); + h = *current; + if (h == '\'' || h =='"') { + c = ++current; + for (l = 0; *c && *c != h; l++) c++; + if (*c != h) + return NULL; + *state = c + 1; + } else { + l = strcspn(current, "\n\r \t"); + *state = current+l; + } + + return pa_xstrndup(current, l); +} + + +static void ucm_volume_free(pa_alsa_ucm_volume *vol) { + pa_assert(vol); + pa_xfree(vol->mixer_elem); + pa_xfree(vol->master_elem); + pa_xfree(vol->master_type); + pa_xfree(vol); +} + +/* Get the volume identifier */ +static char *ucm_get_mixer_id( + pa_alsa_ucm_device *device, + const char *mprop, + const char *cprop, + const char *cid) +{ +#if SND_LIB_VERSION >= 0x10201 /* alsa-lib-1.2.1+ check */ + snd_ctl_elem_id_t *ctl; + int err; +#endif + const char *value; + char *value2; + int index; + + /* mixer element as first, if it's found, return it without modifications */ + value = pa_proplist_gets(device->proplist, mprop); + if (value) + return pa_xstrdup(value); + /* fallback, get the control element identifier */ + /* and try to do some heuristic to determine the mixer element name */ + value = pa_proplist_gets(device->proplist, cprop); + if (value == NULL) + return NULL; +#if SND_LIB_VERSION >= 0x10201 /* alsa-lib-1.2.1+ check */ + /* The new parser may return also element index. */ + snd_ctl_elem_id_alloca(&ctl); + err = snd_use_case_parse_ctl_elem_id(ctl, cid, value); + if (err < 0) + return NULL; + value = snd_ctl_elem_id_get_name(ctl); + index = snd_ctl_elem_id_get_index(ctl); +#else +#warning "Upgrade to alsa-lib 1.2.1!" + index = 0; +#endif + if (!(value2 = pa_str_strip_suffix(value, " Playback Volume"))) + if (!(value2 = pa_str_strip_suffix(value, " Capture Volume"))) + if (!(value2 = pa_str_strip_suffix(value, " Volume"))) + value2 = pa_xstrdup(value); + if (index > 0) { + char *mix = pa_sprintf_malloc("'%s',%d", value2, index); + pa_xfree(value2); + return mix; + } + return value2; +} + +/* Get the volume identifier */ +static pa_alsa_ucm_volume *ucm_get_mixer_volume( + pa_alsa_ucm_device *device, + const char *mprop, + const char *cprop, + const char *cid, + const char *masterid, + const char *mastertype) +{ + pa_alsa_ucm_volume *vol; + char *mixer_elem; + + mixer_elem = ucm_get_mixer_id(device, mprop, cprop, cid); + if (mixer_elem == NULL) + return NULL; + vol = pa_xnew0(pa_alsa_ucm_volume, 1); + if (vol == NULL) { + pa_xfree(mixer_elem); + return NULL; + } + vol->mixer_elem = mixer_elem; + vol->master_elem = pa_xstrdup(pa_proplist_gets(device->proplist, masterid)); + vol->master_type = pa_xstrdup(pa_proplist_gets(device->proplist, mastertype)); + return vol; +} + +/* Get the ALSA mixer device for the UCM device */ +static const char *get_mixer_device(pa_alsa_ucm_device *dev, bool is_sink) +{ + const char *dev_name; + + if (is_sink) { + dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_PLAYBACK_MIXER_DEVICE); + if (!dev_name) + dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_PLAYBACK_CTL_DEVICE); + } else { + dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_CAPTURE_MIXER_DEVICE); + if (!dev_name) + dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_CAPTURE_CTL_DEVICE); + } + return dev_name; +} + +/* Get the ALSA mixer device for the UCM jack */ +static const char *get_jack_mixer_device(pa_alsa_ucm_device *dev, bool is_sink) { + const char *dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_JACK_DEVICE); + if (!dev_name) + return get_mixer_device(dev, is_sink); + return dev_name; +} + +/* Create a property list for this ucm device */ +static int ucm_get_device_property( + pa_alsa_ucm_device *device, + snd_use_case_mgr_t *uc_mgr, + pa_alsa_ucm_verb *verb, + const char *device_name) { + + const char *value; + const char **devices; + char *id, *s; + int i; + int err; + uint32_t ui; + int n_confdev, n_suppdev; + pa_alsa_ucm_volume *vol; + + /* determine the device type */ + device->type = PA_DEVICE_PORT_TYPE_UNKNOWN; + id = s = pa_xstrdup(device_name); + while (s && *s && isalpha(*s)) s++; + if (s) + *s = '\0'; + for (i = 0; types[i].prefix; i++) + if (pa_streq(id, types[i].prefix)) { + device->type = types[i].type; + break; + } + pa_xfree(id); + + /* set properties */ + for (i = 0; item[i].id; i++) { + id = pa_sprintf_malloc("%s/%s", item[i].id, device_name); + err = snd_use_case_get(uc_mgr, id, &value); + pa_xfree(id); + if (err < 0) + continue; + + pa_log_debug("Got %s for device %s: %s", item[i].id, device_name, value); + pa_proplist_sets(device->proplist, item[i].property, value); + free((void*)value); + } + + /* get direction and channels */ + value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_PLAYBACK_CHANNELS); + if (value) { /* output */ + /* get channels */ + if (pa_atou(value, &ui) == 0 && pa_channels_valid(ui)) + device->playback_channels = ui; + else + pa_log("UCM playback channels %s for device %s out of range", value, device_name); + + /* get pcm */ + value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_SINK); + if (!value) /* take pcm from verb playback default */ + pa_log("UCM playback device %s fetch pcm failed", device_name); + } + + if (pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_SINK) && + device->playback_channels == 0) { + pa_log_info("UCM file does not specify 'PlaybackChannels' " + "for device %s, assuming stereo.", device_name); + device->playback_channels = 2; + } + + value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_CAPTURE_CHANNELS); + if (value) { /* input */ + /* get channels */ + if (pa_atou(value, &ui) == 0 && pa_channels_valid(ui)) + device->capture_channels = ui; + else + pa_log("UCM capture channels %s for device %s out of range", value, device_name); + + /* get pcm */ + value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_SOURCE); + if (!value) /* take pcm from verb capture default */ + pa_log("UCM capture device %s fetch pcm failed", device_name); + } + + if (pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_SOURCE) && + device->capture_channels == 0) { + pa_log_info("UCM file does not specify 'CaptureChannels' " + "for device %s, assuming stereo.", device_name); + device->capture_channels = 2; + } + + /* get rate and priority of device */ + if (device->playback_channels) { /* sink device */ + /* get rate */ + if ((value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_PLAYBACK_RATE))) { + if (pa_atou(value, &ui) == 0 && pa_sample_rate_valid(ui)) { + pa_log_debug("UCM playback device %s rate %d", device_name, ui); + device->playback_rate = ui; + } else + pa_log_debug("UCM playback device %s has bad rate %s", device_name, value); + } + + value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_PLAYBACK_PRIORITY); + if (value) { + /* get priority from ucm config */ + if (pa_atou(value, &ui) == 0) + device->playback_priority = ui; + else + pa_log_debug("UCM playback priority %s for device %s error", value, device_name); + } + + vol = ucm_get_mixer_volume(device, + PA_ALSA_PROP_UCM_PLAYBACK_MIXER_ELEM, + PA_ALSA_PROP_UCM_PLAYBACK_VOLUME, + "PlaybackVolume", + PA_ALSA_PROP_UCM_PLAYBACK_MASTER_ELEM, + PA_ALSA_PROP_UCM_PLAYBACK_MASTER_TYPE); + if (vol) + pa_hashmap_put(device->playback_volumes, pa_xstrdup(pa_proplist_gets(verb->proplist, PA_ALSA_PROP_UCM_NAME)), vol); + } + + if (device->capture_channels) { /* source device */ + /* get rate */ + if ((value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_CAPTURE_RATE))) { + if (pa_atou(value, &ui) == 0 && pa_sample_rate_valid(ui)) { + pa_log_debug("UCM capture device %s rate %d", device_name, ui); + device->capture_rate = ui; + } else + pa_log_debug("UCM capture device %s has bad rate %s", device_name, value); + } + + value = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_CAPTURE_PRIORITY); + if (value) { + /* get priority from ucm config */ + if (pa_atou(value, &ui) == 0) + device->capture_priority = ui; + else + pa_log_debug("UCM capture priority %s for device %s error", value, device_name); + } + + vol = ucm_get_mixer_volume(device, + PA_ALSA_PROP_UCM_CAPTURE_MIXER_ELEM, + PA_ALSA_PROP_UCM_CAPTURE_VOLUME, + "CaptureVolume", + PA_ALSA_PROP_UCM_CAPTURE_MASTER_ELEM, + PA_ALSA_PROP_UCM_CAPTURE_MASTER_TYPE); + if (vol) + pa_hashmap_put(device->capture_volumes, pa_xstrdup(pa_proplist_gets(verb->proplist, PA_ALSA_PROP_UCM_NAME)), vol); + } + + if (PA_UCM_PLAYBACK_PRIORITY_UNSET(device) || PA_UCM_CAPTURE_PRIORITY_UNSET(device)) { + /* get priority from static table */ + for (i = 0; dev_info[i].id; i++) { + if (strcasecmp(dev_info[i].id, device_name) == 0) { + PA_UCM_DEVICE_PRIORITY_SET(device, dev_info[i].priority); + break; + } + } + } + + if (PA_UCM_PLAYBACK_PRIORITY_UNSET(device)) { + /* fall through to default priority */ + device->playback_priority = 100; + } + + if (PA_UCM_CAPTURE_PRIORITY_UNSET(device)) { + /* fall through to default priority */ + device->capture_priority = 100; + } + + id = pa_sprintf_malloc("%s/%s", "_conflictingdevs", device_name); + n_confdev = snd_use_case_get_list(uc_mgr, id, &devices); + pa_xfree(id); + + if (n_confdev <= 0) + pa_log_debug("No %s for device %s", "_conflictingdevs", device_name); + else { + device->conflicting_devices = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + ucm_add_devices_to_idxset(device->conflicting_devices, device, verb->devices, devices, n_confdev); + snd_use_case_free_list(devices, n_confdev); + } + + id = pa_sprintf_malloc("%s/%s", "_supporteddevs", device_name); + n_suppdev = snd_use_case_get_list(uc_mgr, id, &devices); + pa_xfree(id); + + if (n_suppdev <= 0) + pa_log_debug("No %s for device %s", "_supporteddevs", device_name); + else { + device->supported_devices = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + ucm_add_devices_to_idxset(device->supported_devices, device, verb->devices, devices, n_suppdev); + snd_use_case_free_list(devices, n_suppdev); + } + + return 0; +}; + +/* Create a property list for this ucm modifier */ +static int ucm_get_modifier_property(pa_alsa_ucm_modifier *modifier, snd_use_case_mgr_t *uc_mgr, const char *modifier_name) { + const char *value; + char *id; + int i; + + for (i = 0; item[i].id; i++) { + int err; + + id = pa_sprintf_malloc("=%s/%s", item[i].id, modifier_name); + err = snd_use_case_get(uc_mgr, id, &value); + pa_xfree(id); + if (err < 0) + continue; + + pa_log_debug("Got %s for modifier %s: %s", item[i].id, modifier_name, value); + pa_proplist_sets(modifier->proplist, item[i].property, value); + free((void*)value); + } + + id = pa_sprintf_malloc("%s/%s", "_conflictingdevs", modifier_name); + modifier->n_confdev = snd_use_case_get_list(uc_mgr, id, &modifier->conflicting_devices); + pa_xfree(id); + if (modifier->n_confdev < 0) + pa_log_debug("No %s for modifier %s", "_conflictingdevs", modifier_name); + + id = pa_sprintf_malloc("%s/%s", "_supporteddevs", modifier_name); + modifier->n_suppdev = snd_use_case_get_list(uc_mgr, id, &modifier->supported_devices); + pa_xfree(id); + if (modifier->n_suppdev < 0) + pa_log_debug("No %s for modifier %s", "_supporteddevs", modifier_name); + + return 0; +}; + +/* Create a list of devices for this verb */ +static int ucm_get_devices(pa_alsa_ucm_verb *verb, snd_use_case_mgr_t *uc_mgr) { + const char **dev_list; + int num_dev, i; + + num_dev = snd_use_case_get_list(uc_mgr, "_devices", &dev_list); + if (num_dev < 0) + return num_dev; + + for (i = 0; i < num_dev; i += 2) { + pa_alsa_ucm_device *d = pa_xnew0(pa_alsa_ucm_device, 1); + + d->proplist = pa_proplist_new(); + pa_proplist_sets(d->proplist, PA_ALSA_PROP_UCM_NAME, pa_strnull(dev_list[i])); + pa_proplist_sets(d->proplist, PA_ALSA_PROP_UCM_DESCRIPTION, pa_strna(dev_list[i + 1])); + d->ucm_ports = pa_dynarray_new(NULL); + d->hw_mute_jacks = pa_dynarray_new(NULL); + d->available = PA_AVAILABLE_UNKNOWN; + + d->playback_volumes = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, pa_xfree, + (pa_free_cb_t) ucm_volume_free); + d->capture_volumes = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, pa_xfree, + (pa_free_cb_t) ucm_volume_free); + + PA_LLIST_PREPEND(pa_alsa_ucm_device, verb->devices, d); + } + + snd_use_case_free_list(dev_list, num_dev); + + return 0; +}; + +static int ucm_get_modifiers(pa_alsa_ucm_verb *verb, snd_use_case_mgr_t *uc_mgr) { + const char **mod_list; + int num_mod, i; + + num_mod = snd_use_case_get_list(uc_mgr, "_modifiers", &mod_list); + if (num_mod < 0) + return num_mod; + + for (i = 0; i < num_mod; i += 2) { + pa_alsa_ucm_modifier *m; + + if (!mod_list[i]) { + pa_log_warn("Got a modifier with a null name. Skipping."); + continue; + } + + m = pa_xnew0(pa_alsa_ucm_modifier, 1); + m->proplist = pa_proplist_new(); + + pa_proplist_sets(m->proplist, PA_ALSA_PROP_UCM_NAME, mod_list[i]); + pa_proplist_sets(m->proplist, PA_ALSA_PROP_UCM_DESCRIPTION, pa_strna(mod_list[i + 1])); + + PA_LLIST_PREPEND(pa_alsa_ucm_modifier, verb->modifiers, m); + } + + snd_use_case_free_list(mod_list, num_mod); + + return 0; +}; + +static void add_role_to_device(pa_alsa_ucm_device *dev, const char *dev_name, const char *role_name, const char *role) { + const char *cur = pa_proplist_gets(dev->proplist, role_name); + + if (!cur) + pa_proplist_sets(dev->proplist, role_name, role); + else if (!pa_str_in_list_spaces(cur, role)) { /* does not exist */ + char *value = pa_sprintf_malloc("%s %s", cur, role); + + pa_proplist_sets(dev->proplist, role_name, value); + pa_xfree(value); + } + + pa_log_info("Add role %s to device %s(%s), result %s", role, dev_name, role_name, pa_proplist_gets(dev->proplist, + role_name)); +} + +static void add_media_role(const char *name, pa_alsa_ucm_device *list, const char *role_name, const char *role, bool is_sink) { + pa_alsa_ucm_device *d; + + PA_LLIST_FOREACH(d, list) { + const char *dev_name = pa_proplist_gets(d->proplist, PA_ALSA_PROP_UCM_NAME); + + if (pa_streq(dev_name, name)) { + const char *sink = pa_proplist_gets(d->proplist, PA_ALSA_PROP_UCM_SINK); + const char *source = pa_proplist_gets(d->proplist, PA_ALSA_PROP_UCM_SOURCE); + + if (is_sink && sink) + add_role_to_device(d, dev_name, role_name, role); + else if (!is_sink && source) + add_role_to_device(d, dev_name, role_name, role); + break; + } + } +} + +static char *modifier_name_to_role(const char *mod_name, bool *is_sink) { + char *sub = NULL, *tmp; + + *is_sink = false; + + if (pa_startswith(mod_name, "Play")) { + *is_sink = true; + sub = pa_xstrdup(mod_name + 4); + } else if (pa_startswith(mod_name, "Capture")) + sub = pa_xstrdup(mod_name + 7); + + if (!sub || !*sub) { + pa_xfree(sub); + pa_log_warn("Can't match media roles for modifer %s", mod_name); + return NULL; + } + + tmp = sub; + + do { + *tmp = tolower(*tmp); + } while (*(++tmp)); + + return sub; +} + +static void ucm_set_media_roles(pa_alsa_ucm_modifier *modifier, pa_alsa_ucm_device *list, const char *mod_name) { + int i; + bool is_sink = false; + char *sub = NULL; + const char *role_name; + + sub = modifier_name_to_role(mod_name, &is_sink); + if (!sub) + return; + + modifier->action_direction = is_sink ? PA_DIRECTION_OUTPUT : PA_DIRECTION_INPUT; + modifier->media_role = sub; + + role_name = is_sink ? PA_ALSA_PROP_UCM_PLAYBACK_ROLES : PA_ALSA_PROP_UCM_CAPTURE_ROLES; + for (i = 0; i < modifier->n_suppdev; i++) { + /* if modifier has no specific pcm, we add role intent to its supported devices */ + if (!pa_proplist_gets(modifier->proplist, PA_ALSA_PROP_UCM_SINK) && + !pa_proplist_gets(modifier->proplist, PA_ALSA_PROP_UCM_SOURCE)) + add_media_role(modifier->supported_devices[i], list, role_name, sub, is_sink); + } +} + +static void append_lost_relationship(pa_alsa_ucm_device *dev) { + uint32_t idx; + pa_alsa_ucm_device *d; + + if (dev->conflicting_devices) { + PA_IDXSET_FOREACH(d, dev->conflicting_devices, idx) { + if (!d->conflicting_devices) + d->conflicting_devices = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + if (pa_idxset_put(d->conflicting_devices, dev, NULL) == 0) + pa_log_warn("Add lost conflicting device %s to %s", + pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_NAME), + pa_proplist_gets(d->proplist, PA_ALSA_PROP_UCM_NAME)); + } + } + + if (dev->supported_devices) { + PA_IDXSET_FOREACH(d, dev->supported_devices, idx) { + if (!d->supported_devices) + d->supported_devices = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + if (pa_idxset_put(d->supported_devices, dev, NULL) == 0) + pa_log_warn("Add lost supported device %s to %s", + pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_NAME), + pa_proplist_gets(d->proplist, PA_ALSA_PROP_UCM_NAME)); + } + } +} + +int pa_alsa_ucm_query_profiles(pa_alsa_ucm_config *ucm, int card_index) { + char *card_name; + const char **verb_list, *value; + int num_verbs, i, err = 0; + + /* support multiple card instances, address card directly by index */ + card_name = pa_sprintf_malloc("hw:%i", card_index); + err = snd_use_case_mgr_open(&ucm->ucm_mgr, card_name); + if (err < 0) { + /* fallback longname: is UCM available for this card ? */ + pa_xfree(card_name); + err = snd_card_get_name(card_index, &card_name); + if (err < 0) { + pa_log("Card can't get card_name from card_index %d", card_index); + err = -PA_ALSA_ERR_UNSPECIFIED; + goto name_fail; + } + + err = snd_use_case_mgr_open(&ucm->ucm_mgr, card_name); + if (err < 0) { + pa_log_info("UCM not available for card %s", card_name); + err = -PA_ALSA_ERR_UCM_OPEN; + goto ucm_mgr_fail; + } + } + + err = snd_use_case_get(ucm->ucm_mgr, "=Linked", &value); + if (err >= 0) { + if (strcasecmp(value, "true") == 0 || strcasecmp(value, "1") == 0) { + free((void *)value); + pa_log_info("Empty (linked) UCM for card %s", card_name); + err = -PA_ALSA_ERR_UCM_LINKED; + goto ucm_verb_fail; + } + free((void *)value); + } + + pa_log_info("UCM available for card %s", card_name); + + /* get a list of all UCM verbs (profiles) for this card */ + num_verbs = snd_use_case_verb_list(ucm->ucm_mgr, &verb_list); + if (num_verbs < 0) { + pa_log("UCM verb list not found for %s", card_name); + err = -PA_ALSA_ERR_UNSPECIFIED; + goto ucm_verb_fail; + } + + /* get the properties of each UCM verb */ + for (i = 0; i < num_verbs; i += 2) { + pa_alsa_ucm_verb *verb; + + /* Get devices and modifiers for each verb */ + err = pa_alsa_ucm_get_verb(ucm->ucm_mgr, verb_list[i], verb_list[i+1], &verb); + if (err < 0) { + pa_log("Failed to get the verb %s", verb_list[i]); + continue; + } + + PA_LLIST_PREPEND(pa_alsa_ucm_verb, ucm->verbs, verb); + } + + if (!ucm->verbs) { + pa_log("No UCM verb is valid for %s", card_name); + err = -PA_ALSA_ERR_UCM_NO_VERB; + } + + snd_use_case_free_list(verb_list, num_verbs); + +ucm_verb_fail: + if (err < 0) { + snd_use_case_mgr_close(ucm->ucm_mgr); + ucm->ucm_mgr = NULL; + } + +ucm_mgr_fail: + pa_xfree(card_name); + +name_fail: + return err; +} + +int pa_alsa_ucm_get_verb(snd_use_case_mgr_t *uc_mgr, const char *verb_name, const char *verb_desc, pa_alsa_ucm_verb **p_verb) { + pa_alsa_ucm_device *d; + pa_alsa_ucm_modifier *mod; + pa_alsa_ucm_verb *verb; + char *value; + unsigned ui; + int err = 0; + + *p_verb = NULL; + pa_log_info("Set UCM verb to %s", verb_name); + err = snd_use_case_set(uc_mgr, "_verb", verb_name); + if (err < 0) + return err; + + verb = pa_xnew0(pa_alsa_ucm_verb, 1); + verb->proplist = pa_proplist_new(); + + pa_proplist_sets(verb->proplist, PA_ALSA_PROP_UCM_NAME, pa_strnull(verb_name)); + pa_proplist_sets(verb->proplist, PA_ALSA_PROP_UCM_DESCRIPTION, pa_strna(verb_desc)); + + value = ucm_verb_value(uc_mgr, verb_name, "Priority"); + if (value && !pa_atou(value, &ui)) + verb->priority = ui > 10000 ? 10000 : ui; + free(value); + + err = ucm_get_devices(verb, uc_mgr); + if (err < 0) + pa_log("No UCM devices for verb %s", verb_name); + + err = ucm_get_modifiers(verb, uc_mgr); + if (err < 0) + pa_log("No UCM modifiers for verb %s", verb_name); + + PA_LLIST_FOREACH(d, verb->devices) { + const char *dev_name = pa_proplist_gets(d->proplist, PA_ALSA_PROP_UCM_NAME); + + /* Devices properties */ + ucm_get_device_property(d, uc_mgr, verb, dev_name); + } + /* make conflicting or supported device mutual */ + PA_LLIST_FOREACH(d, verb->devices) + append_lost_relationship(d); + + PA_LLIST_FOREACH(mod, verb->modifiers) { + const char *mod_name = pa_proplist_gets(mod->proplist, PA_ALSA_PROP_UCM_NAME); + + /* Modifier properties */ + ucm_get_modifier_property(mod, uc_mgr, mod_name); + + /* Set PA_PROP_DEVICE_INTENDED_ROLES property to devices */ + pa_log_debug("Set media roles for verb %s, modifier %s", verb_name, mod_name); + ucm_set_media_roles(mod, verb->devices, mod_name); + } + + *p_verb = verb; + return 0; +} + +static int pa_alsa_ucm_device_cmp(const void *a, const void *b) { + const pa_alsa_ucm_device *d1 = *(pa_alsa_ucm_device **)a; + const pa_alsa_ucm_device *d2 = *(pa_alsa_ucm_device **)b; + + return strcmp(pa_proplist_gets(d1->proplist, PA_ALSA_PROP_UCM_NAME), pa_proplist_gets(d2->proplist, PA_ALSA_PROP_UCM_NAME)); +} + +static void set_eld_devices(pa_hashmap *hash) +{ + pa_device_port *port; + pa_alsa_ucm_port_data *data; + pa_alsa_ucm_device *dev; + const char *eld_mixer_device_name; + void *state; + int idx, eld_device; + + PA_HASHMAP_FOREACH(port, hash, state) { + data = PA_DEVICE_PORT_DATA(port); + eld_mixer_device_name = NULL; + eld_device = -1; + PA_DYNARRAY_FOREACH(dev, data->devices, idx) { + if (dev->eld_device >= 0 && dev->eld_mixer_device_name) { + if (eld_device >= 0 && eld_device != dev->eld_device) { + pa_log_error("The ELD device is already set!"); + } else if (eld_mixer_device_name && pa_streq(dev->eld_mixer_device_name, eld_mixer_device_name)) { + pa_log_error("The ELD mixer device is already set (%s, %s)!", dev->eld_mixer_device_name, dev->eld_mixer_device_name); + } else { + eld_mixer_device_name = dev->eld_mixer_device_name; + eld_device = dev->eld_device; + } + } + } + data->eld_device = eld_device; + data->eld_mixer_device_name = pa_xstrdup(eld_mixer_device_name); + } +} + +static void probe_volumes(pa_hashmap *hash, bool is_sink, snd_pcm_t *pcm_handle, pa_hashmap *mixers, bool ignore_dB) { + pa_device_port *port; + pa_alsa_path *path; + pa_alsa_ucm_port_data *data; + pa_alsa_ucm_device *dev; + snd_mixer_t *mixer_handle; + const char *profile, *mdev, *mdev2; + void *state, *state2; + int idx; + + PA_HASHMAP_FOREACH(port, hash, state) { + data = PA_DEVICE_PORT_DATA(port); + + mdev = NULL; + PA_DYNARRAY_FOREACH(dev, data->devices, idx) { + mdev2 = get_mixer_device(dev, is_sink); + if (mdev && mdev2 && !pa_streq(mdev, mdev2)) { + pa_log_error("Two mixer device names found ('%s', '%s'), using s/w volume", mdev, mdev2); + goto fail; + } + if (mdev2) + mdev = mdev2; + } + + if (mdev == NULL || !(mixer_handle = pa_alsa_open_mixer_by_name(mixers, mdev, true))) { + pa_log_error("Failed to find a working mixer device (%s).", mdev); + goto fail; + } + + PA_HASHMAP_FOREACH_KV(profile, path, data->paths, state2) { + if (pa_alsa_path_probe(path, NULL, mixer_handle, ignore_dB) < 0) { + pa_log_warn("Could not probe path: %s, using s/w volume", data->path->name); + pa_hashmap_remove(data->paths, profile); + } else if (!path->has_volume) { + pa_log_warn("Path %s is not a volume control", data->path->name); + pa_hashmap_remove(data->paths, profile); + } else + pa_log_debug("Set up h/w volume using '%s' for %s:%s", path->name, profile, port->name); + } + } + + return; + +fail: + /* We could not probe the paths we created. Free them and revert to software volumes. */ + PA_HASHMAP_FOREACH(port, hash, state) { + data = PA_DEVICE_PORT_DATA(port); + pa_hashmap_remove_all(data->paths); + } +} + +static void ucm_add_port_combination( + pa_hashmap *hash, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_alsa_ucm_device **pdevices, + int num, + pa_hashmap *ports, + pa_card_profile *cp, + pa_core *core) { + + pa_device_port *port; + int i; + unsigned priority; + double prio2; + char *name, *desc; + const char *dev_name; + const char *direction; + const char *profile; + pa_alsa_ucm_device *sorted[num], *dev; + pa_alsa_ucm_port_data *data; + pa_alsa_ucm_volume *vol; + pa_alsa_jack *jack, *jack2; + pa_device_port_type_t type, type2; + void *state; + + for (i = 0; i < num; i++) + sorted[i] = pdevices[i]; + + /* Sort by alphabetical order so as to have a deterministic naming scheme + * for combination ports */ + qsort(&sorted[0], num, sizeof(pa_alsa_ucm_device *), pa_alsa_ucm_device_cmp); + + dev = sorted[0]; + dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_NAME); + + name = pa_sprintf_malloc("%s%s", is_sink ? PA_UCM_PRE_TAG_OUTPUT : PA_UCM_PRE_TAG_INPUT, dev_name); + desc = num == 1 ? pa_xstrdup(pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_DESCRIPTION)) + : pa_sprintf_malloc("Combination port for %s", dev_name); + + priority = is_sink ? dev->playback_priority : dev->capture_priority; + prio2 = (priority == 0 ? 0 : 1.0/priority); + jack = ucm_get_jack(context->ucm, dev); + type = dev->type; + + for (i = 1; i < num; i++) { + char *tmp; + + dev = sorted[i]; + dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_NAME); + + tmp = pa_sprintf_malloc("%s+%s", name, dev_name); + pa_xfree(name); + name = tmp; + + tmp = pa_sprintf_malloc("%s,%s", desc, dev_name); + pa_xfree(desc); + desc = tmp; + + priority = is_sink ? dev->playback_priority : dev->capture_priority; + if (priority != 0 && prio2 > 0) + prio2 += 1.0/priority; + + jack2 = ucm_get_jack(context->ucm, dev); + if (jack2) { + if (jack && jack != jack2) + pa_log_warn("Multiple jacks per combined device '%s': '%s' '%s'", name, jack->name, jack2->name); + jack = jack2; + } + + type2 = dev->type; + if (type2 != PA_DEVICE_PORT_TYPE_UNKNOWN) { + if (type != PA_DEVICE_PORT_TYPE_UNKNOWN && type != type2) + pa_log_warn("Multiple device types per combined device '%s': %d %d", name, type, type2); + type = type2; + } + } + + /* Make combination ports always have lower priority, and use the formula + 1/p = 1/p1 + 1/p2 + ... 1/pn. + This way, the result will always be less than the individual components, + yet higher components will lead to higher result. */ + + if (num > 1) + priority = prio2 > 0 ? 1.0/prio2 : 0; + + port = pa_hashmap_get(ports, name); + if (!port) { + pa_device_port_new_data port_data; + + pa_device_port_new_data_init(&port_data); + pa_device_port_new_data_set_name(&port_data, name); + pa_device_port_new_data_set_description(&port_data, desc); + pa_device_port_new_data_set_type(&port_data, type); + pa_device_port_new_data_set_direction(&port_data, is_sink ? PA_DIRECTION_OUTPUT : PA_DIRECTION_INPUT); + if (jack) + pa_device_port_new_data_set_availability_group(&port_data, jack->name); + + port = pa_device_port_new(core, &port_data, sizeof(pa_alsa_ucm_port_data)); + pa_device_port_new_data_done(&port_data); + + data = PA_DEVICE_PORT_DATA(port); + ucm_port_data_init(data, context->ucm, port, pdevices, num); + port->impl_free = ucm_port_data_free; + + pa_hashmap_put(ports, port->name, port); + pa_log_debug("Add port %s: %s", port->name, port->description); + + if (num == 1) { + /* To keep things simple and not worry about stacking controls, we only support hardware volumes on non-combination + * ports. */ + data = PA_DEVICE_PORT_DATA(port); + + PA_HASHMAP_FOREACH_KV(profile, vol, is_sink ? dev->playback_volumes : dev->capture_volumes, state) { + pa_alsa_path *path = pa_alsa_path_synthesize(vol->mixer_elem, + is_sink ? PA_ALSA_DIRECTION_OUTPUT : PA_ALSA_DIRECTION_INPUT); + + if (!path) + pa_log_warn("Failed to set up volume control: %s", vol->mixer_elem); + else { + if (vol->master_elem) { + pa_alsa_element *e = pa_alsa_element_get(path, vol->master_elem, false); + e->switch_use = PA_ALSA_SWITCH_MUTE; + e->volume_use = PA_ALSA_VOLUME_MERGE; + } + + pa_hashmap_put(data->paths, pa_xstrdup(profile), path); + + /* Add path also to already created empty path set */ + dev = sorted[0]; + if (is_sink) + pa_hashmap_put(dev->playback_mapping->output_path_set->paths, pa_xstrdup(vol->mixer_elem), path); + else + pa_hashmap_put(dev->capture_mapping->input_path_set->paths, pa_xstrdup(vol->mixer_elem), path); + } + } + } + } + + port->priority = priority; + + pa_xfree(name); + pa_xfree(desc); + + direction = is_sink ? "output" : "input"; + pa_log_debug("Port %s direction %s, priority %d", port->name, direction, priority); + + if (cp) { + pa_log_debug("Adding profile %s to port %s.", cp->name, port->name); + pa_hashmap_put(port->profiles, cp->name, cp); + } + + if (hash) { + pa_hashmap_put(hash, port->name, port); + pa_device_port_ref(port); + } +} + +static int ucm_port_contains(const char *port_name, const char *dev_name, bool is_sink) { + int ret = 0; + const char *r; + const char *state = NULL; + size_t len; + + if (!port_name || !dev_name) + return false; + + port_name += is_sink ? strlen(PA_UCM_PRE_TAG_OUTPUT) : strlen(PA_UCM_PRE_TAG_INPUT); + + while ((r = pa_split_in_place(port_name, "+", &len, &state))) { + if (strlen(dev_name) == len && !strncmp(r, dev_name, len)) { + ret = 1; + break; + } + } + + return ret; +} + +static int ucm_check_conformance( + pa_alsa_ucm_mapping_context *context, + pa_alsa_ucm_device **pdevices, + int dev_num, + pa_alsa_ucm_device *dev) { + + uint32_t idx; + pa_alsa_ucm_device *d; + int i; + + pa_assert(dev); + + pa_log_debug("Check device %s conformance with %d other devices", + pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_NAME), dev_num); + if (dev_num == 0) { + pa_log_debug("First device in combination, number 1"); + return 1; + } + + if (dev->conflicting_devices) { /* the device defines conflicting devices */ + PA_IDXSET_FOREACH(d, dev->conflicting_devices, idx) { + for (i = 0; i < dev_num; i++) { + if (pdevices[i] == d) { + pa_log_debug("Conflicting device found"); + return 0; + } + } + } + } else if (dev->supported_devices) { /* the device defines supported devices */ + for (i = 0; i < dev_num; i++) { + if (!ucm_device_exists(dev->supported_devices, pdevices[i])) { + pa_log_debug("Supported device not found"); + return 0; + } + } + } else { /* not support any other devices */ + pa_log_debug("Not support any other devices"); + return 0; + } + + pa_log_debug("Device added to combination, number %d", dev_num + 1); + return 1; +} + +static inline pa_alsa_ucm_device *get_next_device(pa_idxset *idxset, uint32_t *idx) { + pa_alsa_ucm_device *dev; + + if (*idx == PA_IDXSET_INVALID) + dev = pa_idxset_first(idxset, idx); + else + dev = pa_idxset_next(idxset, idx); + + return dev; +} + +static void ucm_add_ports_combination( + pa_hashmap *hash, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_alsa_ucm_device **pdevices, + int dev_num, + uint32_t map_index, + pa_hashmap *ports, + pa_card_profile *cp, + pa_core *core) { + + pa_alsa_ucm_device *dev; + uint32_t idx = map_index; + + if ((dev = get_next_device(context->ucm_devices, &idx)) == NULL) + return; + + /* check if device at map_index can combine with existing devices combination */ + if (ucm_check_conformance(context, pdevices, dev_num, dev)) { + /* add device at map_index to devices combination */ + pdevices[dev_num] = dev; + /* add current devices combination as a new port */ + ucm_add_port_combination(hash, context, is_sink, pdevices, dev_num + 1, ports, cp, core); + /* try more elements combination */ + ucm_add_ports_combination(hash, context, is_sink, pdevices, dev_num + 1, idx, ports, cp, core); + } + + /* try other device with current elements number */ + ucm_add_ports_combination(hash, context, is_sink, pdevices, dev_num, idx, ports, cp, core); +} + +static char* merge_roles(const char *cur, const char *add) { + char *r, *ret; + const char *state = NULL; + + if (add == NULL) + return pa_xstrdup(cur); + else if (cur == NULL) + return pa_xstrdup(add); + + ret = pa_xstrdup(cur); + + while ((r = pa_split_spaces(add, &state))) { + char *value; + + if (!pa_str_in_list_spaces(ret, r)) + value = pa_sprintf_malloc("%s %s", ret, r); + else { + pa_xfree(r); + continue; + } + + pa_xfree(ret); + ret = value; + pa_xfree(r); + } + + return ret; +} + +void pa_alsa_ucm_add_ports_combination( + pa_hashmap *p, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_hashmap *ports, + pa_card_profile *cp, + pa_core *core) { + + pa_alsa_ucm_device **pdevices; + + pa_assert(context->ucm_devices); + + if (pa_idxset_size(context->ucm_devices) > 0) { + pdevices = pa_xnew(pa_alsa_ucm_device *, pa_idxset_size(context->ucm_devices)); + ucm_add_ports_combination(p, context, is_sink, pdevices, 0, PA_IDXSET_INVALID, ports, cp, core); + pa_xfree(pdevices); + } + + /* ELD devices */ + set_eld_devices(ports); +} + +void pa_alsa_ucm_add_ports( + pa_hashmap **p, + pa_proplist *proplist, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_card *card, + snd_pcm_t *pcm_handle, + bool ignore_dB) { + + uint32_t idx; + char *merged_roles; + const char *role_name = is_sink ? PA_ALSA_PROP_UCM_PLAYBACK_ROLES : PA_ALSA_PROP_UCM_CAPTURE_ROLES; + pa_alsa_ucm_device *dev; + pa_alsa_ucm_modifier *mod; + char *tmp; + + pa_assert(p); + pa_assert(*p); + + /* add ports first */ + pa_alsa_ucm_add_ports_combination(*p, context, is_sink, card->ports, NULL, card->core); + + /* now set up volume paths if any */ + probe_volumes(*p, is_sink, pcm_handle, context->ucm->mixers, ignore_dB); + + /* then set property PA_PROP_DEVICE_INTENDED_ROLES */ + merged_roles = pa_xstrdup(pa_proplist_gets(proplist, PA_PROP_DEVICE_INTENDED_ROLES)); + PA_IDXSET_FOREACH(dev, context->ucm_devices, idx) { + const char *roles = pa_proplist_gets(dev->proplist, role_name); + tmp = merge_roles(merged_roles, roles); + pa_xfree(merged_roles); + merged_roles = tmp; + } + + if (context->ucm_modifiers) + PA_IDXSET_FOREACH(mod, context->ucm_modifiers, idx) { + tmp = merge_roles(merged_roles, mod->media_role); + pa_xfree(merged_roles); + merged_roles = tmp; + } + + if (merged_roles) + pa_proplist_sets(proplist, PA_PROP_DEVICE_INTENDED_ROLES, merged_roles); + + pa_log_info("ALSA device %s roles: %s", pa_proplist_gets(proplist, PA_PROP_DEVICE_STRING), pa_strnull(merged_roles)); + pa_xfree(merged_roles); +} + +/* Change UCM verb and device to match selected card profile */ +int pa_alsa_ucm_set_profile(pa_alsa_ucm_config *ucm, pa_card *card, const char *new_profile, const char *old_profile) { + int ret = 0; + const char *profile; + pa_alsa_ucm_verb *verb; + pa_device_port *port; + pa_alsa_ucm_port_data *data; + void *state; + + if (new_profile == old_profile) + return ret; + else if (new_profile == NULL || old_profile == NULL) + profile = new_profile ? new_profile : SND_USE_CASE_VERB_INACTIVE; + else if (!pa_streq(new_profile, old_profile)) + profile = new_profile; + else + return ret; + + /* change verb */ + pa_log_info("Set UCM verb to %s", profile); + if ((snd_use_case_set(ucm->ucm_mgr, "_verb", profile)) < 0) { + pa_log("Failed to set verb %s", profile); + ret = -1; + } + + /* find active verb */ + ucm->active_verb = NULL; + PA_LLIST_FOREACH(verb, ucm->verbs) { + const char *verb_name; + verb_name = pa_proplist_gets(verb->proplist, PA_ALSA_PROP_UCM_NAME); + if (pa_streq(verb_name, profile)) { + ucm->active_verb = verb; + break; + } + } + + /* select volume controls on ports */ + PA_HASHMAP_FOREACH(port, card->ports, state) { + data = PA_DEVICE_PORT_DATA(port); + data->path = pa_hashmap_get(data->paths, profile); + } + + return ret; +} + +int pa_alsa_ucm_set_port(pa_alsa_ucm_mapping_context *context, pa_device_port *port, bool is_sink) { + int i; + int ret = 0; + pa_alsa_ucm_config *ucm; + const char **enable_devs; + int enable_num = 0; + uint32_t idx; + pa_alsa_ucm_device *dev; + + pa_assert(context && context->ucm); + + ucm = context->ucm; + pa_assert(ucm->ucm_mgr); + + enable_devs = pa_xnew(const char *, pa_idxset_size(context->ucm_devices)); + + /* first disable then enable */ + PA_IDXSET_FOREACH(dev, context->ucm_devices, idx) { + const char *dev_name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_NAME); + + if (ucm_port_contains(port->name, dev_name, is_sink)) + enable_devs[enable_num++] = dev_name; + else { + pa_log_debug("Disable ucm device %s", dev_name); + if (snd_use_case_set(ucm->ucm_mgr, "_disdev", dev_name) > 0) { + pa_log("Failed to disable ucm device %s", dev_name); + ret = -1; + break; + } + } + } + + for (i = 0; i < enable_num; i++) { + pa_log_debug("Enable ucm device %s", enable_devs[i]); + if (snd_use_case_set(ucm->ucm_mgr, "_enadev", enable_devs[i]) < 0) { + pa_log("Failed to enable ucm device %s", enable_devs[i]); + ret = -1; + break; + } + } + + pa_xfree(enable_devs); + + return ret; +} + +static void ucm_add_mapping(pa_alsa_profile *p, pa_alsa_mapping *m) { + + pa_alsa_path_set *ps; + + /* create empty path set for the future path additions */ + ps = pa_xnew0(pa_alsa_path_set, 1); + ps->direction = m->direction; + ps->paths = pa_hashmap_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + switch (m->direction) { + case PA_ALSA_DIRECTION_ANY: + pa_idxset_put(p->output_mappings, m, NULL); + pa_idxset_put(p->input_mappings, m, NULL); + m->output_path_set = ps; + m->input_path_set = ps; + break; + case PA_ALSA_DIRECTION_OUTPUT: + pa_idxset_put(p->output_mappings, m, NULL); + m->output_path_set = ps; + break; + case PA_ALSA_DIRECTION_INPUT: + pa_idxset_put(p->input_mappings, m, NULL); + m->input_path_set = ps; + break; + } +} + +static void alsa_mapping_add_ucm_device(pa_alsa_mapping *m, pa_alsa_ucm_device *device) { + char *cur_desc; + const char *new_desc, *mdev; + bool is_sink = m->direction == PA_ALSA_DIRECTION_OUTPUT; + + pa_idxset_put(m->ucm_context.ucm_devices, device, NULL); + + new_desc = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_DESCRIPTION); + cur_desc = m->description; + if (cur_desc) + m->description = pa_sprintf_malloc("%s + %s", cur_desc, new_desc); + else + m->description = pa_xstrdup(new_desc); + pa_xfree(cur_desc); + + /* walk around null case */ + m->description = m->description ? m->description : pa_xstrdup(""); + + /* save mapping to ucm device */ + if (is_sink) + device->playback_mapping = m; + else + device->capture_mapping = m; + + mdev = get_mixer_device(device, is_sink); + if (mdev) + pa_proplist_sets(m->proplist, "alsa.mixer_device", mdev); +} + +static void alsa_mapping_add_ucm_modifier(pa_alsa_mapping *m, pa_alsa_ucm_modifier *modifier) { + char *cur_desc; + const char *new_desc, *mod_name, *channel_str; + uint32_t channels = 0; + + pa_idxset_put(m->ucm_context.ucm_modifiers, modifier, NULL); + + new_desc = pa_proplist_gets(modifier->proplist, PA_ALSA_PROP_UCM_DESCRIPTION); + cur_desc = m->description; + if (cur_desc) + m->description = pa_sprintf_malloc("%s + %s", cur_desc, new_desc); + else + m->description = pa_xstrdup(new_desc); + pa_xfree(cur_desc); + + m->description = m->description ? m->description : pa_xstrdup(""); + + /* Modifier sinks should not be routed to by default */ + m->priority = 0; + + mod_name = pa_proplist_gets(modifier->proplist, PA_ALSA_PROP_UCM_NAME); + pa_proplist_sets(m->proplist, PA_ALSA_PROP_UCM_MODIFIER, mod_name); + + /* save mapping to ucm modifier */ + if (m->direction == PA_ALSA_DIRECTION_OUTPUT) { + modifier->playback_mapping = m; + channel_str = pa_proplist_gets(modifier->proplist, PA_ALSA_PROP_UCM_PLAYBACK_CHANNELS); + } else { + modifier->capture_mapping = m; + channel_str = pa_proplist_gets(modifier->proplist, PA_ALSA_PROP_UCM_CAPTURE_CHANNELS); + } + + if (channel_str) { + /* FIXME: channel_str is unsanitized input from the UCM configuration, + * we should do proper error handling instead of asserting. + * https://bugs.freedesktop.org/show_bug.cgi?id=71823 */ + pa_assert_se(pa_atou(channel_str, &channels) == 0 && pa_channels_valid(channels)); + pa_log_debug("Got channel count %" PRIu32 " for modifier", channels); + } + + if (channels) + pa_channel_map_init_extend(&m->channel_map, channels, PA_CHANNEL_MAP_ALSA); + else + pa_channel_map_init(&m->channel_map); +} + +static int ucm_create_mapping_direction( + pa_alsa_ucm_config *ucm, + pa_alsa_profile_set *ps, + pa_alsa_profile *p, + pa_alsa_ucm_device *device, + const char *verb_name, + const char *device_name, + const char *device_str, + bool is_sink) { + + pa_alsa_mapping *m; + char *mapping_name; + unsigned priority, rate, channels; + + mapping_name = pa_sprintf_malloc("Mapping %s: %s: %s", verb_name, device_str, is_sink ? "sink" : "source"); + + m = pa_alsa_mapping_get(ps, mapping_name); + if (!m) { + pa_log("No mapping for %s", mapping_name); + pa_xfree(mapping_name); + return -1; + } + pa_log_debug("UCM mapping: %s dev %s", mapping_name, device_name); + pa_xfree(mapping_name); + + priority = is_sink ? device->playback_priority : device->capture_priority; + rate = is_sink ? device->playback_rate : device->capture_rate; + channels = is_sink ? device->playback_channels : device->capture_channels; + + if (!m->ucm_context.ucm_devices) { /* new mapping */ + m->ucm_context.ucm_devices = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + m->ucm_context.ucm = ucm; + m->ucm_context.direction = is_sink ? PA_DIRECTION_OUTPUT : PA_DIRECTION_INPUT; + + m->device_strings = pa_xnew0(char*, 2); + m->device_strings[0] = pa_xstrdup(device_str); + m->direction = is_sink ? PA_ALSA_DIRECTION_OUTPUT : PA_ALSA_DIRECTION_INPUT; + + ucm_add_mapping(p, m); + if (rate) + m->sample_spec.rate = rate; + pa_channel_map_init_extend(&m->channel_map, channels, PA_CHANNEL_MAP_ALSA); + } + + /* mapping priority is the highest one of ucm devices */ + if (priority > m->priority) + m->priority = priority; + + /* mapping channels is the lowest one of ucm devices */ + if (channels < m->channel_map.channels) + pa_channel_map_init_extend(&m->channel_map, channels, PA_CHANNEL_MAP_ALSA); + + alsa_mapping_add_ucm_device(m, device); + + return 0; +} + +static int ucm_create_mapping_for_modifier( + pa_alsa_ucm_config *ucm, + pa_alsa_profile_set *ps, + pa_alsa_profile *p, + pa_alsa_ucm_modifier *modifier, + const char *verb_name, + const char *mod_name, + const char *device_str, + bool is_sink) { + + pa_alsa_mapping *m; + char *mapping_name; + + mapping_name = pa_sprintf_malloc("Mapping %s: %s: %s", verb_name, device_str, is_sink ? "sink" : "source"); + + m = pa_alsa_mapping_get(ps, mapping_name); + if (!m) { + pa_log("no mapping for %s", mapping_name); + pa_xfree(mapping_name); + return -1; + } + pa_log_info("ucm mapping: %s modifier %s", mapping_name, mod_name); + pa_xfree(mapping_name); + + if (!m->ucm_context.ucm_devices && !m->ucm_context.ucm_modifiers) { /* new mapping */ + m->ucm_context.ucm_devices = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + m->ucm_context.ucm_modifiers = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + m->ucm_context.ucm = ucm; + m->ucm_context.direction = is_sink ? PA_DIRECTION_OUTPUT : PA_DIRECTION_INPUT; + + m->device_strings = pa_xnew0(char*, 2); + m->device_strings[0] = pa_xstrdup(device_str); + m->direction = is_sink ? PA_ALSA_DIRECTION_OUTPUT : PA_ALSA_DIRECTION_INPUT; + /* Modifier sinks should not be routed to by default */ + m->priority = 0; + + ucm_add_mapping(p, m); + } else if (!m->ucm_context.ucm_modifiers) /* share pcm with device */ + m->ucm_context.ucm_modifiers = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + alsa_mapping_add_ucm_modifier(m, modifier); + + return 0; +} + +static int ucm_create_mapping( + pa_alsa_ucm_config *ucm, + pa_alsa_profile_set *ps, + pa_alsa_profile *p, + pa_alsa_ucm_device *device, + const char *verb_name, + const char *device_name, + const char *sink, + const char *source) { + + int ret = 0; + + if (!sink && !source) { + pa_log("No sink and source at %s: %s", verb_name, device_name); + return -1; + } + + if (sink) + ret = ucm_create_mapping_direction(ucm, ps, p, device, verb_name, device_name, sink, true); + if (ret == 0 && source) + ret = ucm_create_mapping_direction(ucm, ps, p, device, verb_name, device_name, source, false); + + return ret; +} + +static pa_alsa_jack* ucm_get_jack(pa_alsa_ucm_config *ucm, pa_alsa_ucm_device *device) { + pa_alsa_jack *j; + const char *device_name; + const char *jack_control; + const char *mixer_device_name; + char *name; + + pa_assert(ucm); + pa_assert(device); + + device_name = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_NAME); + + jack_control = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_JACK_CONTROL); + if (jack_control) { +#if SND_LIB_VERSION >= 0x10201 + snd_ctl_elem_id_t *ctl; + int err, index; + snd_ctl_elem_id_alloca(&ctl); + err = snd_use_case_parse_ctl_elem_id(ctl, "JackControl", jack_control); + if (err < 0) + return NULL; + jack_control = snd_ctl_elem_id_get_name(ctl); + index = snd_ctl_elem_id_get_index(ctl); + if (index > 0) { + pa_log("[%s] Invalid JackControl index value: \"%s\",%d", device_name, jack_control, index); + return NULL; + } +#else +#warning "Upgrade to alsa-lib 1.2.1!" +#endif + if (!pa_endswith(jack_control, " Jack")) { + pa_log("[%s] Invalid JackControl value: \"%s\"", device_name, jack_control); + return NULL; + } + + /* pa_alsa_jack_new() expects a jack name without " Jack" at the + * end, so drop the trailing " Jack". */ + name = pa_xstrndup(jack_control, strlen(jack_control) - 5); + } else { + /* The jack control hasn't been explicitly configured, fail. */ + return NULL; + } + + PA_LLIST_FOREACH(j, ucm->jacks) + if (pa_streq(j->name, name)) + goto finish; + + mixer_device_name = get_jack_mixer_device(device, true); + if (!mixer_device_name) + mixer_device_name = get_jack_mixer_device(device, false); + if (!mixer_device_name) { + pa_log("[%s] No mixer device name for JackControl \"%s\"", device_name, jack_control); + return NULL; + } + j = pa_alsa_jack_new(NULL, mixer_device_name, name, 0); + PA_LLIST_PREPEND(pa_alsa_jack, ucm->jacks, j); + +finish: + pa_xfree(name); + + return j; +} + +static int ucm_create_profile( + pa_alsa_ucm_config *ucm, + pa_alsa_profile_set *ps, + pa_alsa_ucm_verb *verb, + const char *verb_name, + const char *verb_desc) { + + pa_alsa_profile *p; + pa_alsa_ucm_device *dev; + pa_alsa_ucm_modifier *mod; + int i = 0; + const char *name, *sink, *source; + unsigned int priority; + + pa_assert(ps); + + if (pa_hashmap_get(ps->profiles, verb_name)) { + pa_log("Verb %s already exists", verb_name); + return -1; + } + + p = pa_xnew0(pa_alsa_profile, 1); + p->profile_set = ps; + p->name = pa_xstrdup(verb_name); + p->description = pa_xstrdup(verb_desc); + + p->output_mappings = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + p->input_mappings = pa_idxset_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + p->supported = true; + pa_hashmap_put(ps->profiles, p->name, p); + + /* TODO: get profile priority from policy management */ + priority = verb->priority; + + if (priority == 0) { + char *verb_cmp, *c; + c = verb_cmp = pa_xstrdup(verb_name); + while (*c) { + if (*c == '_') *c = ' '; + c++; + } + for (i = 0; verb_info[i].id; i++) { + if (strcasecmp(verb_info[i].id, verb_cmp) == 0) { + priority = verb_info[i].priority; + break; + } + } + pa_xfree(verb_cmp); + } + + p->priority = priority; + + PA_LLIST_FOREACH(dev, verb->devices) { + pa_alsa_jack *jack; + const char *jack_hw_mute; + + name = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_NAME); + + sink = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_SINK); + source = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_SOURCE); + + ucm_create_mapping(ucm, ps, p, dev, verb_name, name, sink, source); + + jack = ucm_get_jack(ucm, dev); + if (jack) + device_set_jack(dev, jack); + + /* JackHWMute contains a list of device names. Each listed device must + * be associated with the jack object that we just created. */ + jack_hw_mute = pa_proplist_gets(dev->proplist, PA_ALSA_PROP_UCM_JACK_HW_MUTE); + if (jack_hw_mute && !jack) { + pa_log("[%s] JackHWMute set, but JackControl is missing", name); + jack_hw_mute = NULL; + } + if (jack_hw_mute) { + char *hw_mute_device_name; + const char *state = NULL; + + while ((hw_mute_device_name = ucm_split_devnames(jack_hw_mute, &state))) { + pa_alsa_ucm_verb *verb2; + bool device_found = false; + + /* Search the referenced device from all verbs. If there are + * multiple verbs that have a device with this name, we add the + * hw mute association to each of those devices. */ + PA_LLIST_FOREACH(verb2, ucm->verbs) { + pa_alsa_ucm_device *hw_mute_device; + + hw_mute_device = verb_find_device(verb2, hw_mute_device_name); + if (hw_mute_device) { + device_found = true; + device_add_hw_mute_jack(hw_mute_device, jack); + } + } + + if (!device_found) + pa_log("[%s] JackHWMute references an unknown device: %s", name, hw_mute_device_name); + + pa_xfree(hw_mute_device_name); + } + } + } + + /* Now find modifiers that have their own PlaybackPCM and create + * separate sinks for them. */ + PA_LLIST_FOREACH(mod, verb->modifiers) { + name = pa_proplist_gets(mod->proplist, PA_ALSA_PROP_UCM_NAME); + + sink = pa_proplist_gets(mod->proplist, PA_ALSA_PROP_UCM_SINK); + source = pa_proplist_gets(mod->proplist, PA_ALSA_PROP_UCM_SOURCE); + + if (sink) + ucm_create_mapping_for_modifier(ucm, ps, p, mod, verb_name, name, sink, true); + else if (source) + ucm_create_mapping_for_modifier(ucm, ps, p, mod, verb_name, name, source, false); + } + + pa_alsa_profile_dump(p); + + return 0; +} + +static void mapping_init_eld(pa_alsa_mapping *m, snd_pcm_t *pcm) +{ + pa_alsa_ucm_mapping_context *context = &m->ucm_context; + pa_alsa_ucm_device *dev; + uint32_t idx; + char *mdev; + snd_pcm_info_t *info; + int pcm_card, pcm_device; + + snd_pcm_info_alloca(&info); + if (snd_pcm_info(pcm, info) < 0) + return; + + if ((pcm_card = snd_pcm_info_get_card(info)) < 0) + return; + if ((pcm_device = snd_pcm_info_get_device(info)) < 0) + return; + + PA_IDXSET_FOREACH(dev, context->ucm_devices, idx) { + mdev = pa_sprintf_malloc("hw:%i", pcm_card); + if (mdev == NULL) + continue; + dev->eld_mixer_device_name = mdev; + dev->eld_device = pcm_device; + } +} + +static snd_pcm_t* mapping_open_pcm(pa_alsa_ucm_config *ucm, pa_alsa_mapping *m, int mode) { + snd_pcm_t* pcm; + pa_sample_spec try_ss = ucm->core->default_sample_spec; + pa_channel_map try_map; + snd_pcm_uframes_t try_period_size, try_buffer_size; + bool exact_channels = m->channel_map.channels > 0; + + if (exact_channels) { + try_map = m->channel_map; + try_ss.channels = try_map.channels; + } else + pa_channel_map_init_extend(&try_map, try_ss.channels, PA_CHANNEL_MAP_ALSA); + + try_period_size = + pa_usec_to_bytes(ucm->core->default_fragment_size_msec * PA_USEC_PER_MSEC, &try_ss) / + pa_frame_size(&try_ss); + try_buffer_size = ucm->core->default_n_fragments * try_period_size; + + pcm = pa_alsa_open_by_device_string(m->device_strings[0], NULL, &try_ss, + &try_map, mode, &try_period_size, &try_buffer_size, 0, NULL, NULL, exact_channels); + + if (pcm) { + if (!exact_channels) + m->channel_map = try_map; + mapping_init_eld(m, pcm); + } + + return pcm; +} + +static void profile_finalize_probing(pa_alsa_profile *p) { + pa_alsa_mapping *m; + uint32_t idx; + + PA_IDXSET_FOREACH(m, p->output_mappings, idx) { + if (p->supported) + m->supported++; + + if (!m->output_pcm) + continue; + + snd_pcm_close(m->output_pcm); + m->output_pcm = NULL; + } + + PA_IDXSET_FOREACH(m, p->input_mappings, idx) { + if (p->supported) + m->supported++; + + if (!m->input_pcm) + continue; + + snd_pcm_close(m->input_pcm); + m->input_pcm = NULL; + } +} + +static void ucm_mapping_jack_probe(pa_alsa_mapping *m, pa_hashmap *mixers) { + snd_mixer_t *mixer_handle; + pa_alsa_ucm_mapping_context *context = &m->ucm_context; + pa_alsa_ucm_device *dev; + uint32_t idx; + + PA_IDXSET_FOREACH(dev, context->ucm_devices, idx) { + bool has_control; + + if (!dev->jack || !dev->jack->mixer_device_name) + continue; + + mixer_handle = pa_alsa_open_mixer_by_name(mixers, dev->jack->mixer_device_name, true); + if (!mixer_handle) { + pa_log_error("Unable to determine open mixer device '%s' for jack %s", dev->jack->mixer_device_name, dev->jack->name); + continue; + } + + has_control = pa_alsa_mixer_find_card(mixer_handle, &dev->jack->alsa_id, 0) != NULL; + pa_alsa_jack_set_has_control(dev->jack, has_control); + pa_log_info("UCM jack %s has_control=%d", dev->jack->name, dev->jack->has_control); + } +} + +static void ucm_probe_profile_set(pa_alsa_ucm_config *ucm, pa_alsa_profile_set *ps) { + void *state; + pa_alsa_profile *p; + pa_alsa_mapping *m; + uint32_t idx; + + PA_HASHMAP_FOREACH(p, ps->profiles, state) { + /* change verb */ + pa_log_info("Set ucm verb to %s", p->name); + + if ((snd_use_case_set(ucm->ucm_mgr, "_verb", p->name)) < 0) { + pa_log("Failed to set verb %s", p->name); + p->supported = false; + continue; + } + + PA_IDXSET_FOREACH(m, p->output_mappings, idx) { + if (PA_UCM_IS_MODIFIER_MAPPING(m)) { + /* Skip jack probing on modifier PCMs since we expect this to + * only be controlled on the main device/verb PCM. */ + continue; + } + + m->output_pcm = mapping_open_pcm(ucm, m, SND_PCM_STREAM_PLAYBACK); + if (!m->output_pcm) { + p->supported = false; + break; + } + } + + if (p->supported) { + PA_IDXSET_FOREACH(m, p->input_mappings, idx) { + if (PA_UCM_IS_MODIFIER_MAPPING(m)) { + /* Skip jack probing on modifier PCMs since we expect this to + * only be controlled on the main device/verb PCM. */ + continue; + } + + m->input_pcm = mapping_open_pcm(ucm, m, SND_PCM_STREAM_CAPTURE); + if (!m->input_pcm) { + p->supported = false; + break; + } + } + } + + if (!p->supported) { + profile_finalize_probing(p); + continue; + } + + pa_log_debug("Profile %s supported.", p->name); + + PA_IDXSET_FOREACH(m, p->output_mappings, idx) + if (!PA_UCM_IS_MODIFIER_MAPPING(m)) + ucm_mapping_jack_probe(m, ucm->mixers); + + PA_IDXSET_FOREACH(m, p->input_mappings, idx) + if (!PA_UCM_IS_MODIFIER_MAPPING(m)) + ucm_mapping_jack_probe(m, ucm->mixers); + + profile_finalize_probing(p); + } + + /* restore ucm state */ + snd_use_case_set(ucm->ucm_mgr, "_verb", SND_USE_CASE_VERB_INACTIVE); + + pa_alsa_profile_set_drop_unsupported(ps); +} + +pa_alsa_profile_set* pa_alsa_ucm_add_profile_set(pa_alsa_ucm_config *ucm, pa_channel_map *default_channel_map) { + pa_alsa_ucm_verb *verb; + pa_alsa_profile_set *ps; + + ps = pa_xnew0(pa_alsa_profile_set, 1); + ps->mappings = pa_hashmap_new(pa_idxset_string_hash_func, pa_idxset_string_compare_func); + ps->profiles = pa_hashmap_new(pa_idxset_string_hash_func, pa_idxset_string_compare_func); + ps->decibel_fixes = pa_hashmap_new(pa_idxset_string_hash_func, pa_idxset_string_compare_func); + + /* create a profile for each verb */ + PA_LLIST_FOREACH(verb, ucm->verbs) { + const char *verb_name; + const char *verb_desc; + + verb_name = pa_proplist_gets(verb->proplist, PA_ALSA_PROP_UCM_NAME); + verb_desc = pa_proplist_gets(verb->proplist, PA_ALSA_PROP_UCM_DESCRIPTION); + if (verb_name == NULL) { + pa_log("Verb with no name"); + continue; + } + + ucm_create_profile(ucm, ps, verb, verb_name, verb_desc); + } + + ucm_probe_profile_set(ucm, ps); + ps->probed = true; + + return ps; +} + +static void free_verb(pa_alsa_ucm_verb *verb) { + pa_alsa_ucm_device *di, *dn; + pa_alsa_ucm_modifier *mi, *mn; + + PA_LLIST_FOREACH_SAFE(di, dn, verb->devices) { + PA_LLIST_REMOVE(pa_alsa_ucm_device, verb->devices, di); + + if (di->hw_mute_jacks) + pa_dynarray_free(di->hw_mute_jacks); + + if (di->ucm_ports) + pa_dynarray_free(di->ucm_ports); + + if (di->playback_volumes) + pa_hashmap_free(di->playback_volumes); + if (di->capture_volumes) + pa_hashmap_free(di->capture_volumes); + + pa_proplist_free(di->proplist); + + if (di->conflicting_devices) + pa_idxset_free(di->conflicting_devices, NULL); + if (di->supported_devices) + pa_idxset_free(di->supported_devices, NULL); + + pa_xfree(di->eld_mixer_device_name); + + pa_xfree(di); + } + + PA_LLIST_FOREACH_SAFE(mi, mn, verb->modifiers) { + PA_LLIST_REMOVE(pa_alsa_ucm_modifier, verb->modifiers, mi); + pa_proplist_free(mi->proplist); + if (mi->n_suppdev > 0) + snd_use_case_free_list(mi->supported_devices, mi->n_suppdev); + if (mi->n_confdev > 0) + snd_use_case_free_list(mi->conflicting_devices, mi->n_confdev); + pa_xfree(mi->media_role); + pa_xfree(mi); + } + pa_proplist_free(verb->proplist); + pa_xfree(verb); +} + +static pa_alsa_ucm_device *verb_find_device(pa_alsa_ucm_verb *verb, const char *device_name) { + pa_alsa_ucm_device *device; + + pa_assert(verb); + pa_assert(device_name); + + PA_LLIST_FOREACH(device, verb->devices) { + const char *name; + + name = pa_proplist_gets(device->proplist, PA_ALSA_PROP_UCM_NAME); + if (pa_streq(name, device_name)) + return device; + } + + return NULL; +} + +void pa_alsa_ucm_free(pa_alsa_ucm_config *ucm) { + pa_alsa_ucm_verb *vi, *vn; + pa_alsa_jack *ji, *jn; + + PA_LLIST_FOREACH_SAFE(vi, vn, ucm->verbs) { + PA_LLIST_REMOVE(pa_alsa_ucm_verb, ucm->verbs, vi); + free_verb(vi); + } + PA_LLIST_FOREACH_SAFE(ji, jn, ucm->jacks) { + PA_LLIST_REMOVE(pa_alsa_jack, ucm->jacks, ji); + pa_alsa_jack_free(ji); + } + if (ucm->ucm_mgr) { + snd_use_case_mgr_close(ucm->ucm_mgr); + ucm->ucm_mgr = NULL; + } +} + +void pa_alsa_ucm_mapping_context_free(pa_alsa_ucm_mapping_context *context) { + pa_alsa_ucm_device *dev; + pa_alsa_ucm_modifier *mod; + uint32_t idx; + + if (context->ucm_devices) { + /* clear ucm device pointer to mapping */ + PA_IDXSET_FOREACH(dev, context->ucm_devices, idx) { + if (context->direction == PA_DIRECTION_OUTPUT) + dev->playback_mapping = NULL; + else + dev->capture_mapping = NULL; + } + + pa_idxset_free(context->ucm_devices, NULL); + } + + if (context->ucm_modifiers) { + PA_IDXSET_FOREACH(mod, context->ucm_modifiers, idx) { + if (context->direction == PA_DIRECTION_OUTPUT) + mod->playback_mapping = NULL; + else + mod->capture_mapping = NULL; + } + + pa_idxset_free(context->ucm_modifiers, NULL); + } +} + +/* Enable the modifier when the first stream with matched role starts */ +void pa_alsa_ucm_roled_stream_begin(pa_alsa_ucm_config *ucm, const char *role, pa_direction_t dir) { + pa_alsa_ucm_modifier *mod; + + if (!ucm->active_verb) + return; + + PA_LLIST_FOREACH(mod, ucm->active_verb->modifiers) { + if ((mod->action_direction == dir) && (pa_streq(mod->media_role, role))) { + if (mod->enabled_counter == 0) { + const char *mod_name = pa_proplist_gets(mod->proplist, PA_ALSA_PROP_UCM_NAME); + + pa_log_info("Enable ucm modifier %s", mod_name); + if (snd_use_case_set(ucm->ucm_mgr, "_enamod", mod_name) < 0) { + pa_log("Failed to enable ucm modifier %s", mod_name); + } + } + + mod->enabled_counter++; + break; + } + } +} + +/* Disable the modifier when the last stream with matched role ends */ +void pa_alsa_ucm_roled_stream_end(pa_alsa_ucm_config *ucm, const char *role, pa_direction_t dir) { + pa_alsa_ucm_modifier *mod; + + if (!ucm->active_verb) + return; + + PA_LLIST_FOREACH(mod, ucm->active_verb->modifiers) { + if ((mod->action_direction == dir) && (pa_streq(mod->media_role, role))) { + + mod->enabled_counter--; + if (mod->enabled_counter == 0) { + const char *mod_name = pa_proplist_gets(mod->proplist, PA_ALSA_PROP_UCM_NAME); + + pa_log_info("Disable ucm modifier %s", mod_name); + if (snd_use_case_set(ucm->ucm_mgr, "_dismod", mod_name) < 0) { + pa_log("Failed to disable ucm modifier %s", mod_name); + } + } + + break; + } + } +} + +static void device_add_ucm_port(pa_alsa_ucm_device *device, pa_alsa_ucm_port_data *port) { + pa_assert(device); + pa_assert(port); + + pa_dynarray_append(device->ucm_ports, port); +} + +static void device_set_jack(pa_alsa_ucm_device *device, pa_alsa_jack *jack) { + pa_assert(device); + pa_assert(jack); + + device->jack = jack; + pa_alsa_jack_add_ucm_device(jack, device); + + pa_alsa_ucm_device_update_available(device); +} + +static void device_add_hw_mute_jack(pa_alsa_ucm_device *device, pa_alsa_jack *jack) { + pa_assert(device); + pa_assert(jack); + + pa_dynarray_append(device->hw_mute_jacks, jack); + pa_alsa_jack_add_ucm_hw_mute_device(jack, device); + + pa_alsa_ucm_device_update_available(device); +} + +static void device_set_available(pa_alsa_ucm_device *device, pa_available_t available) { + pa_alsa_ucm_port_data *port; + unsigned idx; + + pa_assert(device); + + if (available == device->available) + return; + + device->available = available; + + PA_DYNARRAY_FOREACH(port, device->ucm_ports, idx) + ucm_port_update_available(port); +} + +void pa_alsa_ucm_device_update_available(pa_alsa_ucm_device *device) { + pa_available_t available = PA_AVAILABLE_UNKNOWN; + pa_alsa_jack *jack; + unsigned idx; + + pa_assert(device); + + if (device->jack && device->jack->has_control) + available = device->jack->plugged_in ? PA_AVAILABLE_YES : PA_AVAILABLE_NO; + + PA_DYNARRAY_FOREACH(jack, device->hw_mute_jacks, idx) { + if (jack->plugged_in) { + available = PA_AVAILABLE_NO; + break; + } + } + + device_set_available(device, available); +} + +static void ucm_port_data_init(pa_alsa_ucm_port_data *port, pa_alsa_ucm_config *ucm, pa_device_port *core_port, + pa_alsa_ucm_device **devices, unsigned n_devices) { + unsigned i; + + pa_assert(ucm); + pa_assert(core_port); + pa_assert(devices); + + port->ucm = ucm; + port->core_port = core_port; + port->devices = pa_dynarray_new(NULL); + port->eld_device = -1; + + for (i = 0; i < n_devices; i++) { + pa_dynarray_append(port->devices, devices[i]); + device_add_ucm_port(devices[i], port); + } + + port->paths = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, pa_xfree, + (pa_free_cb_t) pa_alsa_path_free); + + ucm_port_update_available(port); +} + +static void ucm_port_data_free(pa_device_port *port) { + pa_alsa_ucm_port_data *ucm_port; + + pa_assert(port); + + ucm_port = PA_DEVICE_PORT_DATA(port); + + if (ucm_port->devices) + pa_dynarray_free(ucm_port->devices); + + if (ucm_port->paths) + pa_hashmap_free(ucm_port->paths); + + pa_xfree(ucm_port->eld_mixer_device_name); +} + +static void ucm_port_update_available(pa_alsa_ucm_port_data *port) { + pa_alsa_ucm_device *device; + unsigned idx; + pa_available_t available = PA_AVAILABLE_YES; + + pa_assert(port); + + PA_DYNARRAY_FOREACH(device, port->devices, idx) { + if (device->available == PA_AVAILABLE_UNKNOWN) + available = PA_AVAILABLE_UNKNOWN; + else if (device->available == PA_AVAILABLE_NO) { + available = PA_AVAILABLE_NO; + break; + } + } + + pa_device_port_set_available(port->core_port, available); +} + +#else /* HAVE_ALSA_UCM */ + +/* Dummy functions for systems without UCM support */ + +int pa_alsa_ucm_query_profiles(pa_alsa_ucm_config *ucm, int card_index) { + pa_log_info("UCM not available."); + return -1; +} + +pa_alsa_profile_set* pa_alsa_ucm_add_profile_set(pa_alsa_ucm_config *ucm, pa_channel_map *default_channel_map) { + return NULL; +} + +int pa_alsa_ucm_set_profile(pa_alsa_ucm_config *ucm, pa_card *card, const char *new_profile, const char *old_profile) { + return -1; +} + +int pa_alsa_ucm_get_verb(snd_use_case_mgr_t *uc_mgr, const char *verb_name, const char *verb_desc, pa_alsa_ucm_verb **p_verb) { + return -1; +} + +void pa_alsa_ucm_add_ports( + pa_hashmap **hash, + pa_proplist *proplist, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_card *card, + snd_pcm_t *pcm_handle, + bool ignore_dB) { +} + +void pa_alsa_ucm_add_ports_combination( + pa_hashmap *hash, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_hashmap *ports, + pa_card_profile *cp, + pa_core *core) { +} + +int pa_alsa_ucm_set_port(pa_alsa_ucm_mapping_context *context, pa_device_port *port, bool is_sink) { + return -1; +} + +void pa_alsa_ucm_free(pa_alsa_ucm_config *ucm) { +} + +void pa_alsa_ucm_mapping_context_free(pa_alsa_ucm_mapping_context *context) { +} + +void pa_alsa_ucm_roled_stream_begin(pa_alsa_ucm_config *ucm, const char *role, pa_direction_t dir) { +} + +void pa_alsa_ucm_roled_stream_end(pa_alsa_ucm_config *ucm, const char *role, pa_direction_t dir) { +} + +#endif diff --git a/src/modules/alsa/alsa-ucm.h b/src/modules/alsa/alsa-ucm.h new file mode 100644 index 0000000..0bb2eb5 --- /dev/null +++ b/src/modules/alsa/alsa-ucm.h @@ -0,0 +1,294 @@ +#ifndef fooalsaucmhfoo +#define fooalsaucmhfoo + +/*** + This file is part of PulseAudio. + + Copyright 2011 Wolfson Microelectronics PLC + Author Margarita Olaya <magi@slimlogic.co.uk> + Copyright 2012 Feng Wei <wei.feng@freescale.com>, Freescale Ltd. + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_ALSA_UCM +#include <alsa/use-case.h> +#else +typedef void snd_use_case_mgr_t; +#endif + +#include "alsa-mixer.h" + +/** For devices: List of verbs, devices or modifiers available */ +#define PA_ALSA_PROP_UCM_NAME "alsa.ucm.name" + +/** For devices: List of supported devices per verb*/ +#define PA_ALSA_PROP_UCM_DESCRIPTION "alsa.ucm.description" + +/** For devices: Playback device name e.g PlaybackPCM */ +#define PA_ALSA_PROP_UCM_SINK "alsa.ucm.sink" + +/** For devices: Capture device name e.g CapturePCM*/ +#define PA_ALSA_PROP_UCM_SOURCE "alsa.ucm.source" + +/** For devices: Playback roles */ +#define PA_ALSA_PROP_UCM_PLAYBACK_ROLES "alsa.ucm.playback.roles" + +/** For devices: Playback control device name */ +#define PA_ALSA_PROP_UCM_PLAYBACK_CTL_DEVICE "alsa.ucm.playback.ctldev" + +/** For devices: Playback control volume ID string. e.g PlaybackVolume */ +#define PA_ALSA_PROP_UCM_PLAYBACK_VOLUME "alsa.ucm.playback.volume" + +/** For devices: Playback switch e.g PlaybackSwitch */ +#define PA_ALSA_PROP_UCM_PLAYBACK_SWITCH "alsa.ucm.playback.switch" + +/** For devices: Playback mixer device name */ +#define PA_ALSA_PROP_UCM_PLAYBACK_MIXER_DEVICE "alsa.ucm.playback.mixer.device" + +/** For devices: Playback mixer identifier */ +#define PA_ALSA_PROP_UCM_PLAYBACK_MIXER_ELEM "alsa.ucm.playback.mixer.element" + +/** For devices: Playback mixer master identifier */ +#define PA_ALSA_PROP_UCM_PLAYBACK_MASTER_ELEM "alsa.ucm.playback.master.element" + +/** For devices: Playback mixer master type */ +#define PA_ALSA_PROP_UCM_PLAYBACK_MASTER_TYPE "alsa.ucm.playback.master.type" + +/** For devices: Playback mixer master identifier */ +#define PA_ALSA_PROP_UCM_PLAYBACK_MASTER_ID "alsa.ucm.playback.master.id" + +/** For devices: Playback mixer master type */ +#define PA_ALSA_PROP_UCM_PLAYBACK_MASTER_TYPE "alsa.ucm.playback.master.type" + +/** For devices: Playback priority */ +#define PA_ALSA_PROP_UCM_PLAYBACK_PRIORITY "alsa.ucm.playback.priority" + +/** For devices: Playback rate */ +#define PA_ALSA_PROP_UCM_PLAYBACK_RATE "alsa.ucm.playback.rate" + +/** For devices: Playback channels */ +#define PA_ALSA_PROP_UCM_PLAYBACK_CHANNELS "alsa.ucm.playback.channels" + +/** For devices: Capture roles */ +#define PA_ALSA_PROP_UCM_CAPTURE_ROLES "alsa.ucm.capture.roles" + +/** For devices: Capture control device name */ +#define PA_ALSA_PROP_UCM_CAPTURE_CTL_DEVICE "alsa.ucm.capture.ctldev" + +/** For devices: Capture controls volume ID string. e.g CaptureVolume */ +#define PA_ALSA_PROP_UCM_CAPTURE_VOLUME "alsa.ucm.capture.volume" + +/** For devices: Capture switch e.g CaptureSwitch */ +#define PA_ALSA_PROP_UCM_CAPTURE_SWITCH "alsa.ucm.capture.switch" + +/** For devices: Capture mixer device name */ +#define PA_ALSA_PROP_UCM_CAPTURE_MIXER_DEVICE "alsa.ucm.capture.mixer.device" + +/** For devices: Capture mixer identifier */ +#define PA_ALSA_PROP_UCM_CAPTURE_MIXER_ELEM "alsa.ucm.capture.mixer.element" + +/** For devices: Capture mixer identifier */ +#define PA_ALSA_PROP_UCM_CAPTURE_MASTER_ELEM "alsa.ucm.capture.master.element" + +/** For devices: Capture mixer identifier */ +#define PA_ALSA_PROP_UCM_CAPTURE_MASTER_TYPE "alsa.ucm.capture.master.type" + +/** For devices: Capture mixer identifier */ +#define PA_ALSA_PROP_UCM_CAPTURE_MASTER_ID "alsa.ucm.capture.master.id" + +/** For devices: Capture mixer identifier */ +#define PA_ALSA_PROP_UCM_CAPTURE_MASTER_TYPE "alsa.ucm.capture.master.type" + +/** For devices: Capture priority */ +#define PA_ALSA_PROP_UCM_CAPTURE_PRIORITY "alsa.ucm.capture.priority" + +/** For devices: Capture rate */ +#define PA_ALSA_PROP_UCM_CAPTURE_RATE "alsa.ucm.capture.rate" + +/** For devices: Capture channels */ +#define PA_ALSA_PROP_UCM_CAPTURE_CHANNELS "alsa.ucm.capture.channels" + +/** For devices: Quality of Service */ +#define PA_ALSA_PROP_UCM_QOS "alsa.ucm.qos" + +/** For devices: The modifier (if any) that this device corresponds to */ +#define PA_ALSA_PROP_UCM_MODIFIER "alsa.ucm.modifier" + +/* Corresponds to the "JackCTL" UCM value. */ +#define PA_ALSA_PROP_UCM_JACK_DEVICE "alsa.ucm.jack_device" + +/* Corresponds to the "JackControl" UCM value. */ +#define PA_ALSA_PROP_UCM_JACK_CONTROL "alsa.ucm.jack_control" + +/* Corresponds to the "JackHWMute" UCM value. */ +#define PA_ALSA_PROP_UCM_JACK_HW_MUTE "alsa.ucm.jack_hw_mute" + +typedef struct pa_alsa_ucm_verb pa_alsa_ucm_verb; +typedef struct pa_alsa_ucm_modifier pa_alsa_ucm_modifier; +typedef struct pa_alsa_ucm_device pa_alsa_ucm_device; +typedef struct pa_alsa_ucm_config pa_alsa_ucm_config; +typedef struct pa_alsa_ucm_mapping_context pa_alsa_ucm_mapping_context; +typedef struct pa_alsa_ucm_port_data pa_alsa_ucm_port_data; +typedef struct pa_alsa_ucm_volume pa_alsa_ucm_volume; + +int pa_alsa_ucm_query_profiles(pa_alsa_ucm_config *ucm, int card_index); +pa_alsa_profile_set* pa_alsa_ucm_add_profile_set(pa_alsa_ucm_config *ucm, pa_channel_map *default_channel_map); +int pa_alsa_ucm_set_profile(pa_alsa_ucm_config *ucm, pa_card *card, const char *new_profile, const char *old_profile); + +int pa_alsa_ucm_get_verb(snd_use_case_mgr_t *uc_mgr, const char *verb_name, const char *verb_desc, pa_alsa_ucm_verb **p_verb); + +void pa_alsa_ucm_add_ports( + pa_hashmap **hash, + pa_proplist *proplist, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_card *card, + snd_pcm_t *pcm_handle, + bool ignore_dB); +void pa_alsa_ucm_add_ports_combination( + pa_hashmap *hash, + pa_alsa_ucm_mapping_context *context, + bool is_sink, + pa_hashmap *ports, + pa_card_profile *cp, + pa_core *core); +int pa_alsa_ucm_set_port(pa_alsa_ucm_mapping_context *context, pa_device_port *port, bool is_sink); + +void pa_alsa_ucm_free(pa_alsa_ucm_config *ucm); +void pa_alsa_ucm_mapping_context_free(pa_alsa_ucm_mapping_context *context); + +void pa_alsa_ucm_roled_stream_begin(pa_alsa_ucm_config *ucm, const char *role, pa_direction_t dir); +void pa_alsa_ucm_roled_stream_end(pa_alsa_ucm_config *ucm, const char *role, pa_direction_t dir); + +/* UCM - Use Case Manager is available on some audio cards */ + +struct pa_alsa_ucm_device { + PA_LLIST_FIELDS(pa_alsa_ucm_device); + + pa_proplist *proplist; + + pa_device_port_type_t type; + + unsigned playback_priority; + unsigned capture_priority; + + unsigned playback_rate; + unsigned capture_rate; + + unsigned playback_channels; + unsigned capture_channels; + + /* These may be different per verb, so we store this as a hashmap of verb -> volume_control. We might eventually want to + * make this a hashmap of verb -> per-verb-device-properties-struct. */ + pa_hashmap *playback_volumes; + pa_hashmap *capture_volumes; + + pa_alsa_mapping *playback_mapping; + pa_alsa_mapping *capture_mapping; + + pa_idxset *conflicting_devices; + pa_idxset *supported_devices; + + /* One device may be part of multiple ports, since each device has + * a dedicated port, and in addition to that we sometimes generate ports + * that represent combinations of devices. */ + pa_dynarray *ucm_ports; /* struct ucm_port */ + + pa_alsa_jack *jack; + pa_dynarray *hw_mute_jacks; /* pa_alsa_jack */ + pa_available_t available; + + char *eld_mixer_device_name; + int eld_device; +}; + +void pa_alsa_ucm_device_update_available(pa_alsa_ucm_device *device); + +struct pa_alsa_ucm_modifier { + PA_LLIST_FIELDS(pa_alsa_ucm_modifier); + + pa_proplist *proplist; + + int n_confdev; + int n_suppdev; + + const char **conflicting_devices; + const char **supported_devices; + + pa_direction_t action_direction; + + char *media_role; + + /* Non-NULL if the modifier has its own PlaybackPCM/CapturePCM */ + pa_alsa_mapping *playback_mapping; + pa_alsa_mapping *capture_mapping; + + /* Count how many role matched streams are running */ + int enabled_counter; +}; + +struct pa_alsa_ucm_verb { + PA_LLIST_FIELDS(pa_alsa_ucm_verb); + + pa_proplist *proplist; + unsigned priority; + + PA_LLIST_HEAD(pa_alsa_ucm_device, devices); + PA_LLIST_HEAD(pa_alsa_ucm_modifier, modifiers); +}; + +struct pa_alsa_ucm_config { + pa_core *core; + snd_use_case_mgr_t *ucm_mgr; + pa_alsa_ucm_verb *active_verb; + + pa_hashmap *mixers; + PA_LLIST_HEAD(pa_alsa_ucm_verb, verbs); + PA_LLIST_HEAD(pa_alsa_jack, jacks); +}; + +struct pa_alsa_ucm_mapping_context { + pa_alsa_ucm_config *ucm; + pa_direction_t direction; + + pa_idxset *ucm_devices; + pa_idxset *ucm_modifiers; +}; + +struct pa_alsa_ucm_port_data { + pa_alsa_ucm_config *ucm; + pa_device_port *core_port; + + /* A single port will be associated with multiple devices if it represents + * a combination of devices. */ + pa_dynarray *devices; /* pa_alsa_ucm_device */ + + /* profile name -> pa_alsa_path for volume control */ + pa_hashmap *paths; + /* Current path, set when activating profile */ + pa_alsa_path *path; + + /* ELD info */ + char *eld_mixer_device_name; + int eld_device; /* PCM device number */ +}; + +struct pa_alsa_ucm_volume { + char *mixer_elem; /* mixer element identifier */ + char *master_elem; /* master mixer element identifier */ + char *master_type; +}; + +#endif diff --git a/src/modules/alsa/alsa-util.c b/src/modules/alsa/alsa-util.c new file mode 100644 index 0000000..172a7bb --- /dev/null +++ b/src/modules/alsa/alsa-util.c @@ -0,0 +1,1891 @@ +/*** + This file is part of PulseAudio. + + Copyright 2004-2009 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <sys/types.h> +#include <alsa/asoundlib.h> + +#include <pulse/sample.h> +#include <pulse/xmalloc.h> +#include <pulse/timeval.h> +#include <pulse/util.h> +#include <pulse/utf8.h> + +#include <pulsecore/i18n.h> +#include <pulsecore/log.h> +#include <pulsecore/macro.h> +#include <pulsecore/core-util.h> +#include <pulsecore/atomic.h> +#include <pulsecore/core-error.h> +#include <pulsecore/thread.h> +#include <pulsecore/conf-parser.h> +#include <pulsecore/core-rtclock.h> + +#include "alsa-util.h" +#include "alsa-mixer.h" + +#ifdef HAVE_UDEV +#include <modules/udev-util.h> +#endif + +static int set_format(snd_pcm_t *pcm_handle, snd_pcm_hw_params_t *hwparams, pa_sample_format_t *f) { + + static const snd_pcm_format_t format_trans[] = { + [PA_SAMPLE_U8] = SND_PCM_FORMAT_U8, + [PA_SAMPLE_ALAW] = SND_PCM_FORMAT_A_LAW, + [PA_SAMPLE_ULAW] = SND_PCM_FORMAT_MU_LAW, + [PA_SAMPLE_S16LE] = SND_PCM_FORMAT_S16_LE, + [PA_SAMPLE_S16BE] = SND_PCM_FORMAT_S16_BE, + [PA_SAMPLE_FLOAT32LE] = SND_PCM_FORMAT_FLOAT_LE, + [PA_SAMPLE_FLOAT32BE] = SND_PCM_FORMAT_FLOAT_BE, + [PA_SAMPLE_S32LE] = SND_PCM_FORMAT_S32_LE, + [PA_SAMPLE_S32BE] = SND_PCM_FORMAT_S32_BE, + [PA_SAMPLE_S24LE] = SND_PCM_FORMAT_S24_3LE, + [PA_SAMPLE_S24BE] = SND_PCM_FORMAT_S24_3BE, + [PA_SAMPLE_S24_32LE] = SND_PCM_FORMAT_S24_LE, + [PA_SAMPLE_S24_32BE] = SND_PCM_FORMAT_S24_BE, + }; + + static const pa_sample_format_t try_order[] = { + PA_SAMPLE_FLOAT32NE, + PA_SAMPLE_FLOAT32RE, + PA_SAMPLE_S32NE, + PA_SAMPLE_S32RE, + PA_SAMPLE_S24_32NE, + PA_SAMPLE_S24_32RE, + PA_SAMPLE_S24NE, + PA_SAMPLE_S24RE, + PA_SAMPLE_S16NE, + PA_SAMPLE_S16RE, + PA_SAMPLE_ALAW, + PA_SAMPLE_ULAW, + PA_SAMPLE_U8 + }; + + unsigned i; + int ret; + + pa_assert(pcm_handle); + pa_assert(hwparams); + pa_assert(f); + + if ((ret = snd_pcm_hw_params_set_format(pcm_handle, hwparams, format_trans[*f])) >= 0) + return ret; + + pa_log_debug("snd_pcm_hw_params_set_format(%s) failed: %s", + snd_pcm_format_description(format_trans[*f]), + pa_alsa_strerror(ret)); + + if (*f == PA_SAMPLE_FLOAT32BE) + *f = PA_SAMPLE_FLOAT32LE; + else if (*f == PA_SAMPLE_FLOAT32LE) + *f = PA_SAMPLE_FLOAT32BE; + else if (*f == PA_SAMPLE_S24BE) + *f = PA_SAMPLE_S24LE; + else if (*f == PA_SAMPLE_S24LE) + *f = PA_SAMPLE_S24BE; + else if (*f == PA_SAMPLE_S24_32BE) + *f = PA_SAMPLE_S24_32LE; + else if (*f == PA_SAMPLE_S24_32LE) + *f = PA_SAMPLE_S24_32BE; + else if (*f == PA_SAMPLE_S16BE) + *f = PA_SAMPLE_S16LE; + else if (*f == PA_SAMPLE_S16LE) + *f = PA_SAMPLE_S16BE; + else if (*f == PA_SAMPLE_S32BE) + *f = PA_SAMPLE_S32LE; + else if (*f == PA_SAMPLE_S32LE) + *f = PA_SAMPLE_S32BE; + else + goto try_auto; + + if ((ret = snd_pcm_hw_params_set_format(pcm_handle, hwparams, format_trans[*f])) >= 0) + return ret; + + pa_log_debug("snd_pcm_hw_params_set_format(%s) failed: %s", + snd_pcm_format_description(format_trans[*f]), + pa_alsa_strerror(ret)); + +try_auto: + + for (i = 0; i < PA_ELEMENTSOF(try_order); i++) { + *f = try_order[i]; + + if ((ret = snd_pcm_hw_params_set_format(pcm_handle, hwparams, format_trans[*f])) >= 0) + return ret; + + pa_log_debug("snd_pcm_hw_params_set_format(%s) failed: %s", + snd_pcm_format_description(format_trans[*f]), + pa_alsa_strerror(ret)); + } + + return -1; +} + +static int set_period_size(snd_pcm_t *pcm_handle, snd_pcm_hw_params_t *hwparams, snd_pcm_uframes_t size) { + snd_pcm_uframes_t s; + int d, ret; + + pa_assert(pcm_handle); + pa_assert(hwparams); + + s = size; + d = 0; + if (snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &s, &d) < 0) { + s = size; + d = -1; + if (snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &s, &d) < 0) { + s = size; + d = 1; + if ((ret = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &s, &d)) < 0) { + pa_log_info("snd_pcm_hw_params_set_period_size_near() failed: %s", pa_alsa_strerror(ret)); + return ret; + } + } + } + + return 0; +} + +static int set_buffer_size(snd_pcm_t *pcm_handle, snd_pcm_hw_params_t *hwparams, snd_pcm_uframes_t size) { + int ret; + + pa_assert(pcm_handle); + pa_assert(hwparams); + + if ((ret = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &size)) < 0) { + pa_log_info("snd_pcm_hw_params_set_buffer_size_near() failed: %s", pa_alsa_strerror(ret)); + return ret; + } + + return 0; +} + +static void check_access(snd_pcm_t *pcm_handle, snd_pcm_hw_params_t *hwparams, bool use_mmap) { + if ((use_mmap && !snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_MMAP_INTERLEAVED)) || + !snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) + pa_log_error("Weird, PCM claims to support interleaved access, but snd_pcm_hw_params_set_access() failed."); + + if ((use_mmap && !snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_MMAP_NONINTERLEAVED)) || + !snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_NONINTERLEAVED)) + pa_log_debug("PCM seems to support non-interleaved access, but PA doesn't."); + else if (use_mmap && !snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_MMAP_COMPLEX)) { + pa_log_debug("PCM seems to support mmapped complex access, but PA doesn't."); + } +} + +/* Set the hardware parameters of the given ALSA device. Returns the + * selected fragment settings in *buffer_size and *period_size. Determine + * whether mmap and tsched mode can be enabled. */ +int pa_alsa_set_hw_params( + snd_pcm_t *pcm_handle, + pa_sample_spec *ss, + snd_pcm_uframes_t *period_size, + snd_pcm_uframes_t *buffer_size, + snd_pcm_uframes_t tsched_size, + bool *use_mmap, + bool *use_tsched, + bool require_exact_channel_number) { + + int ret = -1; + snd_pcm_hw_params_t *hwparams, *hwparams_copy; + int dir; + snd_pcm_uframes_t _period_size = period_size ? *period_size : 0; + snd_pcm_uframes_t _buffer_size = buffer_size ? *buffer_size : 0; + bool _use_mmap = use_mmap && *use_mmap; + bool _use_tsched = use_tsched && *use_tsched; + pa_sample_spec _ss = *ss; + + pa_assert(pcm_handle); + pa_assert(ss); + + snd_pcm_hw_params_alloca(&hwparams); + snd_pcm_hw_params_alloca(&hwparams_copy); + + if ((ret = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { + pa_log_debug("snd_pcm_hw_params_any() failed: %s", pa_alsa_strerror(ret)); + goto finish; + } + + if ((ret = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 0)) < 0) { + pa_log_debug("snd_pcm_hw_params_set_rate_resample() failed: %s", pa_alsa_strerror(ret)); + goto finish; + } + + if (_use_mmap) { + + if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) { + + /* mmap() didn't work, fall back to interleaved */ + + if ((ret = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { + pa_log_debug("snd_pcm_hw_params_set_access() failed: %s", pa_alsa_strerror(ret)); + check_access(pcm_handle, hwparams, true); + goto finish; + } + + _use_mmap = false; + } + + } else if ((ret = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { + pa_log_debug("snd_pcm_hw_params_set_access() failed: %s", pa_alsa_strerror(ret)); + check_access(pcm_handle, hwparams, false); + goto finish; + } + + if (!_use_mmap) + _use_tsched = false; + + if (!pa_alsa_pcm_is_hw(pcm_handle)) + _use_tsched = false; + + /* The PCM pointer is only updated with period granularity */ + if (snd_pcm_hw_params_is_batch(hwparams)) { + bool is_usb = false; + const char *id; + snd_pcm_info_t* pcm_info; + snd_pcm_info_alloca(&pcm_info); + + if (snd_pcm_info(pcm_handle, pcm_info) == 0 && + (id = snd_pcm_info_get_id(pcm_info))) { + /* This horrible hack makes sure we don't disable tsched on USB + * devices, which have a low enough transfer size for timer-based + * scheduling to work. This can go away when the ALSA API supprots + * querying the block transfer size. */ + if (pa_streq(id, "USB Audio")) + is_usb = true; + } + + if (!is_usb) { + pa_log_info("Disabling tsched mode since BATCH flag is set"); + _use_tsched = false; + } + } + +#if (SND_LIB_VERSION >= ((1<<16)|(0<<8)|24)) /* API additions in 1.0.24 */ + if (_use_tsched) { + + /* try to disable period wakeups if hardware can do so */ + if (snd_pcm_hw_params_can_disable_period_wakeup(hwparams)) { + + if ((ret = snd_pcm_hw_params_set_period_wakeup(pcm_handle, hwparams, false)) < 0) + /* don't bail, keep going with default mode with period wakeups */ + pa_log_debug("snd_pcm_hw_params_set_period_wakeup() failed: %s", pa_alsa_strerror(ret)); + else + pa_log_info("Trying to disable ALSA period wakeups, using timers only"); + } else + pa_log_info("Cannot disable ALSA period wakeups"); + } +#endif + + if ((ret = set_format(pcm_handle, hwparams, &_ss.format)) < 0) + goto finish; + + if ((ret = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &_ss.rate, NULL)) < 0) { + pa_log_debug("snd_pcm_hw_params_set_rate_near() failed: %s", pa_alsa_strerror(ret)); + goto finish; + } + + /* We ignore very small sampling rate deviations */ + if (_ss.rate >= ss->rate*.95 && _ss.rate <= ss->rate*1.05) + _ss.rate = ss->rate; + + if (require_exact_channel_number) { + if ((ret = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, _ss.channels)) < 0) { + pa_log_debug("snd_pcm_hw_params_set_channels(%u) failed: %s", _ss.channels, pa_alsa_strerror(ret)); + goto finish; + } + } else { + unsigned int c = _ss.channels; + + if ((ret = snd_pcm_hw_params_set_channels_near(pcm_handle, hwparams, &c)) < 0) { + pa_log_debug("snd_pcm_hw_params_set_channels_near(%u) failed: %s", _ss.channels, pa_alsa_strerror(ret)); + goto finish; + } + + _ss.channels = c; + } + + if (_use_tsched && tsched_size > 0) { + _buffer_size = (snd_pcm_uframes_t) (((uint64_t) tsched_size * _ss.rate) / ss->rate); + _period_size = _buffer_size; + } else { + _period_size = (snd_pcm_uframes_t) (((uint64_t) _period_size * _ss.rate) / ss->rate); + _buffer_size = (snd_pcm_uframes_t) (((uint64_t) _buffer_size * _ss.rate) / ss->rate); + } + + if (_buffer_size > 0 || _period_size > 0) { + snd_pcm_uframes_t max_frames = 0; + + if ((ret = snd_pcm_hw_params_get_buffer_size_max(hwparams, &max_frames)) < 0) + pa_log_warn("snd_pcm_hw_params_get_buffer_size_max() failed: %s", pa_alsa_strerror(ret)); + else + pa_log_debug("Maximum hw buffer size is %lu ms", (long unsigned) (max_frames * PA_MSEC_PER_SEC / _ss.rate)); + + /* Some ALSA drivers really don't like if we set the buffer + * size first and the number of periods second (which would + * make a lot more sense to me). So, try a few combinations + * before we give up. */ + + if (_buffer_size > 0 && _period_size > 0) { + snd_pcm_hw_params_copy(hwparams_copy, hwparams); + + /* First try: set buffer size first, followed by period size */ + if (set_buffer_size(pcm_handle, hwparams_copy, _buffer_size) >= 0 && + set_period_size(pcm_handle, hwparams_copy, _period_size) >= 0 && + snd_pcm_hw_params(pcm_handle, hwparams_copy) >= 0) { + pa_log_debug("Set buffer size first (to %lu samples), period size second (to %lu samples).", (unsigned long) _buffer_size, (unsigned long) _period_size); + goto success; + } + + snd_pcm_hw_params_copy(hwparams_copy, hwparams); + /* Second try: set period size first, followed by buffer size */ + if (set_period_size(pcm_handle, hwparams_copy, _period_size) >= 0 && + set_buffer_size(pcm_handle, hwparams_copy, _buffer_size) >= 0 && + snd_pcm_hw_params(pcm_handle, hwparams_copy) >= 0) { + pa_log_debug("Set period size first (to %lu samples), buffer size second (to %lu samples).", (unsigned long) _period_size, (unsigned long) _buffer_size); + goto success; + } + } + + if (_buffer_size > 0) { + snd_pcm_hw_params_copy(hwparams_copy, hwparams); + + /* Third try: set only buffer size */ + if (set_buffer_size(pcm_handle, hwparams_copy, _buffer_size) >= 0 && + snd_pcm_hw_params(pcm_handle, hwparams_copy) >= 0) { + pa_log_debug("Set only buffer size (to %lu samples).", (unsigned long) _buffer_size); + goto success; + } + } + + if (_period_size > 0) { + snd_pcm_hw_params_copy(hwparams_copy, hwparams); + + /* Fourth try: set only period size */ + if (set_period_size(pcm_handle, hwparams_copy, _period_size) >= 0 && + snd_pcm_hw_params(pcm_handle, hwparams_copy) >= 0) { + pa_log_debug("Set only period size (to %lu samples).", (unsigned long) _period_size); + goto success; + } + } + } + + pa_log_debug("Set neither period nor buffer size."); + + /* Last chance, set nothing */ + if ((ret = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { + pa_log_info("snd_pcm_hw_params failed: %s", pa_alsa_strerror(ret)); + goto finish; + } + +success: + + if (ss->rate != _ss.rate) + pa_log_info("Device %s doesn't support %u Hz, changed to %u Hz.", snd_pcm_name(pcm_handle), ss->rate, _ss.rate); + + if (ss->channels != _ss.channels) + pa_log_info("Device %s doesn't support %u channels, changed to %u.", snd_pcm_name(pcm_handle), ss->channels, _ss.channels); + + if (ss->format != _ss.format) + pa_log_info("Device %s doesn't support sample format %s, changed to %s.", snd_pcm_name(pcm_handle), pa_sample_format_to_string(ss->format), pa_sample_format_to_string(_ss.format)); + + if ((ret = snd_pcm_hw_params_current(pcm_handle, hwparams)) < 0) { + pa_log_info("snd_pcm_hw_params_current() failed: %s", pa_alsa_strerror(ret)); + goto finish; + } + + if ((ret = snd_pcm_hw_params_get_period_size(hwparams, &_period_size, &dir)) < 0 || + (ret = snd_pcm_hw_params_get_buffer_size(hwparams, &_buffer_size)) < 0) { + pa_log_info("snd_pcm_hw_params_get_{period|buffer}_size() failed: %s", pa_alsa_strerror(ret)); + goto finish; + } + +#if (SND_LIB_VERSION >= ((1<<16)|(0<<8)|24)) /* API additions in 1.0.24 */ + if (_use_tsched) { + unsigned int no_wakeup; + /* see if period wakeups were disabled */ + snd_pcm_hw_params_get_period_wakeup(pcm_handle, hwparams, &no_wakeup); + if (no_wakeup == 0) + pa_log_info("ALSA period wakeups disabled"); + else + pa_log_info("ALSA period wakeups were not disabled"); + } +#endif + + ss->rate = _ss.rate; + ss->channels = _ss.channels; + ss->format = _ss.format; + + pa_assert(_period_size > 0); + pa_assert(_buffer_size > 0); + + if (buffer_size) + *buffer_size = _buffer_size; + + if (period_size) + *period_size = _period_size; + + if (use_mmap) + *use_mmap = _use_mmap; + + if (use_tsched) + *use_tsched = _use_tsched; + + ret = 0; + +finish: + + return ret; +} + +int pa_alsa_set_sw_params(snd_pcm_t *pcm, snd_pcm_uframes_t avail_min, bool period_event) { + snd_pcm_sw_params_t *swparams; + snd_pcm_uframes_t boundary; + int err; + + pa_assert(pcm); + + snd_pcm_sw_params_alloca(&swparams); + + if ((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) { + pa_log_warn("Unable to determine current swparams: %s", pa_alsa_strerror(err)); + return err; + } + + if ((err = snd_pcm_sw_params_set_period_event(pcm, swparams, period_event)) < 0) { + pa_log_warn("Unable to disable period event: %s", pa_alsa_strerror(err)); + return err; + } + + if ((err = snd_pcm_sw_params_set_tstamp_mode(pcm, swparams, SND_PCM_TSTAMP_ENABLE)) < 0) { + pa_log_warn("Unable to enable time stamping: %s", pa_alsa_strerror(err)); + return err; + } + + if ((err = snd_pcm_sw_params_get_boundary(swparams, &boundary)) < 0) { + pa_log_warn("Unable to get boundary: %s", pa_alsa_strerror(err)); + return err; + } + + if ((err = snd_pcm_sw_params_set_stop_threshold(pcm, swparams, boundary)) < 0) { + pa_log_warn("Unable to set stop threshold: %s", pa_alsa_strerror(err)); + return err; + } + + if ((err = snd_pcm_sw_params_set_start_threshold(pcm, swparams, (snd_pcm_uframes_t) -1)) < 0) { + pa_log_warn("Unable to set start threshold: %s", pa_alsa_strerror(err)); + return err; + } + + if ((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0) { + pa_log_error("snd_pcm_sw_params_set_avail_min() failed: %s", pa_alsa_strerror(err)); + return err; + } + + if ((err = snd_pcm_sw_params(pcm, swparams)) < 0) { + pa_log_warn("Unable to set sw params: %s", pa_alsa_strerror(err)); + return err; + } + + return 0; +} + +snd_pcm_t *pa_alsa_open_by_device_id_auto( + const char *dev_id, + char **dev, + pa_sample_spec *ss, + pa_channel_map* map, + int mode, + snd_pcm_uframes_t *period_size, + snd_pcm_uframes_t *buffer_size, + snd_pcm_uframes_t tsched_size, + bool *use_mmap, + bool *use_tsched, + pa_alsa_profile_set *ps, + pa_alsa_mapping **mapping) { + + char *d; + snd_pcm_t *pcm_handle; + void *state; + pa_alsa_mapping *m; + + pa_assert(dev_id); + pa_assert(dev); + pa_assert(ss); + pa_assert(map); + pa_assert(ps); + + /* First we try to find a device string with a superset of the + * requested channel map. We iterate through our device table from + * top to bottom and take the first that matches. If we didn't + * find a working device that way, we iterate backwards, and check + * all devices that do not provide a superset of the requested + * channel map.*/ + + PA_HASHMAP_FOREACH(m, ps->mappings, state) { + if (!pa_channel_map_superset(&m->channel_map, map)) + continue; + + pa_log_debug("Checking for superset %s (%s)", m->name, m->device_strings[0]); + + pcm_handle = pa_alsa_open_by_device_id_mapping( + dev_id, + dev, + ss, + map, + mode, + period_size, + buffer_size, + tsched_size, + use_mmap, + use_tsched, + m); + + if (pcm_handle) { + if (mapping) + *mapping = m; + + return pcm_handle; + } + } + + PA_HASHMAP_FOREACH_BACKWARDS(m, ps->mappings, state) { + if (pa_channel_map_superset(&m->channel_map, map)) + continue; + + pa_log_debug("Checking for subset %s (%s)", m->name, m->device_strings[0]); + + pcm_handle = pa_alsa_open_by_device_id_mapping( + dev_id, + dev, + ss, + map, + mode, + period_size, + buffer_size, + tsched_size, + use_mmap, + use_tsched, + m); + + if (pcm_handle) { + if (mapping) + *mapping = m; + + return pcm_handle; + } + } + + /* OK, we didn't find any good device, so let's try the raw hw: stuff */ + d = pa_sprintf_malloc("hw:%s", dev_id); + pa_log_debug("Trying %s as last resort...", d); + pcm_handle = pa_alsa_open_by_device_string( + d, + dev, + ss, + map, + mode, + period_size, + buffer_size, + tsched_size, + use_mmap, + use_tsched, + false); + pa_xfree(d); + + if (pcm_handle && mapping) + *mapping = NULL; + + return pcm_handle; +} + +snd_pcm_t *pa_alsa_open_by_device_id_mapping( + const char *dev_id, + char **dev, + pa_sample_spec *ss, + pa_channel_map* map, + int mode, + snd_pcm_uframes_t *period_size, + snd_pcm_uframes_t *buffer_size, + snd_pcm_uframes_t tsched_size, + bool *use_mmap, + bool *use_tsched, + pa_alsa_mapping *m) { + + snd_pcm_t *pcm_handle; + pa_sample_spec try_ss; + pa_channel_map try_map; + + pa_assert(dev_id); + pa_assert(dev); + pa_assert(ss); + pa_assert(map); + pa_assert(m); + + try_ss.channels = m->channel_map.channels; + try_ss.rate = ss->rate; + try_ss.format = ss->format; + try_map = m->channel_map; + + pcm_handle = pa_alsa_open_by_template( + m->device_strings, + dev_id, + dev, + &try_ss, + &try_map, + mode, + period_size, + buffer_size, + tsched_size, + use_mmap, + use_tsched, + pa_channel_map_valid(&m->channel_map) /* Query the channel count if we don't know what we want */); + + if (!pcm_handle) + return NULL; + + *ss = try_ss; + *map = try_map; + pa_assert(map->channels == ss->channels); + + return pcm_handle; +} + +snd_pcm_t *pa_alsa_open_by_device_string( + const char *device, + char **dev, + pa_sample_spec *ss, + pa_channel_map* map, + int mode, + snd_pcm_uframes_t *period_size, + snd_pcm_uframes_t *buffer_size, + snd_pcm_uframes_t tsched_size, + bool *use_mmap, + bool *use_tsched, + bool require_exact_channel_number) { + + int err; + char *d; + snd_pcm_t *pcm_handle; + bool reformat = false; + + pa_assert(device); + pa_assert(ss); + pa_assert(map); + + d = pa_xstrdup(device); + + for (;;) { + pa_log_debug("Trying %s %s SND_PCM_NO_AUTO_FORMAT ...", d, reformat ? "without" : "with"); + + if ((err = snd_pcm_open(&pcm_handle, d, mode, + SND_PCM_NONBLOCK| + SND_PCM_NO_AUTO_RESAMPLE| + SND_PCM_NO_AUTO_CHANNELS| + (reformat ? 0 : SND_PCM_NO_AUTO_FORMAT))) < 0) { + pa_log_info("Error opening PCM device %s: %s", d, pa_alsa_strerror(err)); + goto fail; + } + + pa_log_debug("Managed to open %s", d); + + if ((err = pa_alsa_set_hw_params( + pcm_handle, + ss, + period_size, + buffer_size, + tsched_size, + use_mmap, + use_tsched, + require_exact_channel_number)) < 0) { + + if (!reformat) { + reformat = true; + + snd_pcm_close(pcm_handle); + continue; + } + + /* Hmm, some hw is very exotic, so we retry with plug, if without it didn't work */ + if (!pa_startswith(d, "plug:") && !pa_startswith(d, "plughw:")) { + char *t; + + t = pa_sprintf_malloc("plug:%s", d); + pa_xfree(d); + d = t; + + reformat = false; + + snd_pcm_close(pcm_handle); + continue; + } + + pa_log_info("Failed to set hardware parameters on %s: %s", d, pa_alsa_strerror(err)); + snd_pcm_close(pcm_handle); + + goto fail; + } + + if (ss->channels > PA_CHANNELS_MAX) { + pa_log("Device %s has %u channels, but PulseAudio supports only %u channels. Unable to use the device.", + d, ss->channels, PA_CHANNELS_MAX); + snd_pcm_close(pcm_handle); + goto fail; + } + + if (dev) + *dev = d; + else + pa_xfree(d); + + if (ss->channels != map->channels) + pa_channel_map_init_extend(map, ss->channels, PA_CHANNEL_MAP_ALSA); + + return pcm_handle; + } + +fail: + pa_xfree(d); + + return NULL; +} + +snd_pcm_t *pa_alsa_open_by_template( + char **template, + const char *dev_id, + char **dev, + pa_sample_spec *ss, + pa_channel_map* map, + int mode, + snd_pcm_uframes_t *period_size, + snd_pcm_uframes_t *buffer_size, + snd_pcm_uframes_t tsched_size, + bool *use_mmap, + bool *use_tsched, + bool require_exact_channel_number) { + + snd_pcm_t *pcm_handle; + char **i; + + for (i = template; *i; i++) { + char *d; + + d = pa_replace(*i, "%f", dev_id); + + pcm_handle = pa_alsa_open_by_device_string( + d, + dev, + ss, + map, + mode, + period_size, + buffer_size, + tsched_size, + use_mmap, + use_tsched, + require_exact_channel_number); + + pa_xfree(d); + + if (pcm_handle) + return pcm_handle; + } + + return NULL; +} + +void pa_alsa_dump(pa_log_level_t level, snd_pcm_t *pcm) { + int err; + snd_output_t *out; + + pa_assert(pcm); + + pa_assert_se(snd_output_buffer_open(&out) == 0); + + if ((err = snd_pcm_dump(pcm, out)) < 0) + pa_logl(level, "snd_pcm_dump(): %s", pa_alsa_strerror(err)); + else { + char *s = NULL; + snd_output_buffer_string(out, &s); + pa_logl(level, "snd_pcm_dump():\n%s", pa_strnull(s)); + } + + pa_assert_se(snd_output_close(out) == 0); +} + +void pa_alsa_dump_status(snd_pcm_t *pcm) { + int err; + snd_output_t *out; + snd_pcm_status_t *status; + char *s = NULL; + + pa_assert(pcm); + + snd_pcm_status_alloca(&status); + + if ((err = snd_output_buffer_open(&out)) < 0) { + pa_log_debug("snd_output_buffer_open() failed: %s", pa_cstrerror(err)); + return; + } + + if ((err = snd_pcm_status(pcm, status)) < 0) { + pa_log_debug("snd_pcm_status() failed: %s", pa_cstrerror(err)); + goto finish; + } + + if ((err = snd_pcm_status_dump(status, out)) < 0) { + pa_log_debug("snd_pcm_status_dump(): %s", pa_alsa_strerror(err)); + goto finish; + } + + snd_output_buffer_string(out, &s); + pa_log_debug("snd_pcm_status_dump():\n%s", pa_strnull(s)); + +finish: + + snd_output_close(out); +} + +static void alsa_error_handler(const char *file, int line, const char *function, int err, const char *fmt,...) { + va_list ap; + char *alsa_file; + + alsa_file = pa_sprintf_malloc("(alsa-lib)%s", file); + + va_start(ap, fmt); + + pa_log_levelv_meta(PA_LOG_INFO, alsa_file, line, function, fmt, ap); + + va_end(ap); + + pa_xfree(alsa_file); +} + +static pa_atomic_t n_error_handler_installed = PA_ATOMIC_INIT(0); + +void pa_alsa_refcnt_inc(void) { + /* This is not really thread safe, but we do our best */ + + if (pa_atomic_inc(&n_error_handler_installed) == 0) + snd_lib_error_set_handler(alsa_error_handler); +} + +void pa_alsa_refcnt_dec(void) { + int r; + + pa_assert_se((r = pa_atomic_dec(&n_error_handler_installed)) >= 1); + + if (r == 1) { + snd_lib_error_set_handler(NULL); + snd_config_update_free_global(); + } +} + +bool pa_alsa_init_description(pa_proplist *p, pa_card *card) { + const char *d, *k; + pa_assert(p); + + if (pa_device_init_description(p, card)) + return true; + + if (!(d = pa_proplist_gets(p, "alsa.card_name"))) + d = pa_proplist_gets(p, "alsa.name"); + + if (!d) + return false; + + k = pa_proplist_gets(p, PA_PROP_DEVICE_PROFILE_DESCRIPTION); + + if (d && k) + pa_proplist_setf(p, PA_PROP_DEVICE_DESCRIPTION, "%s %s", d, k); + else if (d) + pa_proplist_sets(p, PA_PROP_DEVICE_DESCRIPTION, d); + + return false; +} + +void pa_alsa_init_proplist_card(pa_core *c, pa_proplist *p, int card) { + char *cn, *lcn, *dn; + + pa_assert(p); + pa_assert(card >= 0); + + pa_proplist_setf(p, "alsa.card", "%i", card); + + if (snd_card_get_name(card, &cn) >= 0) { + pa_proplist_sets(p, "alsa.card_name", pa_strip(cn)); + free(cn); + } + + if (snd_card_get_longname(card, &lcn) >= 0) { + pa_proplist_sets(p, "alsa.long_card_name", pa_strip(lcn)); + free(lcn); + } + + if ((dn = pa_alsa_get_driver_name(card))) { + pa_proplist_sets(p, "alsa.driver_name", dn); + pa_xfree(dn); + } + +#ifdef HAVE_UDEV + pa_udev_get_info(card, p); +#endif +} + +void pa_alsa_init_proplist_pcm_info(pa_core *c, pa_proplist *p, snd_pcm_info_t *pcm_info) { + + static const char * const alsa_class_table[SND_PCM_CLASS_LAST+1] = { + [SND_PCM_CLASS_GENERIC] = "generic", + [SND_PCM_CLASS_MULTI] = "multi", + [SND_PCM_CLASS_MODEM] = "modem", + [SND_PCM_CLASS_DIGITIZER] = "digitizer" + }; + static const char * const class_table[SND_PCM_CLASS_LAST+1] = { + [SND_PCM_CLASS_GENERIC] = "sound", + [SND_PCM_CLASS_MULTI] = NULL, + [SND_PCM_CLASS_MODEM] = "modem", + [SND_PCM_CLASS_DIGITIZER] = NULL + }; + static const char * const alsa_subclass_table[SND_PCM_SUBCLASS_LAST+1] = { + [SND_PCM_SUBCLASS_GENERIC_MIX] = "generic-mix", + [SND_PCM_SUBCLASS_MULTI_MIX] = "multi-mix" + }; + + snd_pcm_class_t class; + snd_pcm_subclass_t subclass; + const char *n, *id, *sdn; + int card; + + pa_assert(p); + pa_assert(pcm_info); + + pa_proplist_sets(p, PA_PROP_DEVICE_API, "alsa"); + + if ((class = snd_pcm_info_get_class(pcm_info)) <= SND_PCM_CLASS_LAST) { + if (class_table[class]) + pa_proplist_sets(p, PA_PROP_DEVICE_CLASS, class_table[class]); + if (alsa_class_table[class]) + pa_proplist_sets(p, "alsa.class", alsa_class_table[class]); + } + + if ((subclass = snd_pcm_info_get_subclass(pcm_info)) <= SND_PCM_SUBCLASS_LAST) + if (alsa_subclass_table[subclass]) + pa_proplist_sets(p, "alsa.subclass", alsa_subclass_table[subclass]); + + if ((n = snd_pcm_info_get_name(pcm_info))) { + char *t = pa_xstrdup(n); + pa_proplist_sets(p, "alsa.name", pa_strip(t)); + pa_xfree(t); + } + + if ((id = snd_pcm_info_get_id(pcm_info))) + pa_proplist_sets(p, "alsa.id", id); + + pa_proplist_setf(p, "alsa.subdevice", "%u", snd_pcm_info_get_subdevice(pcm_info)); + if ((sdn = snd_pcm_info_get_subdevice_name(pcm_info))) + pa_proplist_sets(p, "alsa.subdevice_name", sdn); + + pa_proplist_setf(p, "alsa.device", "%u", snd_pcm_info_get_device(pcm_info)); + + if ((card = snd_pcm_info_get_card(pcm_info)) >= 0) + pa_alsa_init_proplist_card(c, p, card); +} + +void pa_alsa_init_proplist_pcm(pa_core *c, pa_proplist *p, snd_pcm_t *pcm) { + snd_pcm_hw_params_t *hwparams; + snd_pcm_info_t *info; + int bits, err; + + snd_pcm_hw_params_alloca(&hwparams); + snd_pcm_info_alloca(&info); + + if ((err = snd_pcm_hw_params_current(pcm, hwparams)) < 0) + pa_log_warn("Error fetching hardware parameter info: %s", pa_alsa_strerror(err)); + else { + + if ((bits = snd_pcm_hw_params_get_sbits(hwparams)) >= 0) + pa_proplist_setf(p, "alsa.resolution_bits", "%i", bits); + } + + if ((err = snd_pcm_info(pcm, info)) < 0) + pa_log_warn("Error fetching PCM info: %s", pa_alsa_strerror(err)); + else + pa_alsa_init_proplist_pcm_info(c, p, info); +} + +void pa_alsa_init_proplist_ctl(pa_proplist *p, const char *name) { + int err; + snd_ctl_t *ctl; + snd_ctl_card_info_t *info; + const char *t; + + pa_assert(p); + + snd_ctl_card_info_alloca(&info); + + if ((err = snd_ctl_open(&ctl, name, 0)) < 0) { + pa_log_warn("Error opening low-level control device '%s': %s", name, snd_strerror(err)); + return; + } + + if ((err = snd_ctl_card_info(ctl, info)) < 0) { + pa_log_warn("Control device %s card info: %s", name, snd_strerror(err)); + snd_ctl_close(ctl); + return; + } + + if ((t = snd_ctl_card_info_get_mixername(info)) && *t) + pa_proplist_sets(p, "alsa.mixer_name", t); + + if ((t = snd_ctl_card_info_get_components(info)) && *t) + pa_proplist_sets(p, "alsa.components", t); + + snd_ctl_close(ctl); +} + +int pa_alsa_recover_from_poll(snd_pcm_t *pcm, int revents) { + snd_pcm_state_t state; + snd_pcm_hw_params_t *hwparams; + int err; + + pa_assert(pcm); + + if (revents & POLLERR) + pa_log_debug("Got POLLERR from ALSA"); + if (revents & POLLNVAL) + pa_log_warn("Got POLLNVAL from ALSA"); + if (revents & POLLHUP) + pa_log_warn("Got POLLHUP from ALSA"); + if (revents & POLLPRI) + pa_log_warn("Got POLLPRI from ALSA"); + if (revents & POLLIN) + pa_log_debug("Got POLLIN from ALSA"); + if (revents & POLLOUT) + pa_log_debug("Got POLLOUT from ALSA"); + + state = snd_pcm_state(pcm); + pa_log_debug("PCM state is %s", snd_pcm_state_name(state)); + + /* Try to recover from this error */ + + switch (state) { + + case SND_PCM_STATE_DISCONNECTED: + /* Do not try to recover */ + pa_log_info("Device disconnected."); + return -1; + + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_recover(pcm, -EPIPE, 1)) != 0) { + pa_log_warn("Could not recover from POLLERR|POLLNVAL|POLLHUP and XRUN: %s", pa_alsa_strerror(err)); + return -1; + } + break; + + case SND_PCM_STATE_SUSPENDED: + snd_pcm_hw_params_alloca(&hwparams); + + if ((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) { + pa_log_debug("snd_pcm_hw_params_any() failed: %s", pa_alsa_strerror(err)); + return -1; + } + + if (snd_pcm_hw_params_can_resume(hwparams)) { + /* Retry resume 3 times before giving up, then fallback to restarting the stream. */ + for (int i = 0; i < 3; i++) { + if ((err = snd_pcm_resume(pcm)) == 0) + return 0; + if (err != -EAGAIN) + break; + pa_msleep(25); + } + pa_log_warn("Could not recover alsa device from SUSPENDED state, trying to restart PCM"); + } + /* Fall through */ + + default: + + snd_pcm_drop(pcm); + return 1; + } + + return 0; +} + +pa_rtpoll_item* pa_alsa_build_pollfd(snd_pcm_t *pcm, pa_rtpoll *rtpoll) { + int n, err; + struct pollfd *pollfd; + pa_rtpoll_item *item; + + pa_assert(pcm); + + if ((n = snd_pcm_poll_descriptors_count(pcm)) < 0) { + pa_log("snd_pcm_poll_descriptors_count() failed: %s", pa_alsa_strerror(n)); + return NULL; + } + + item = pa_rtpoll_item_new(rtpoll, PA_RTPOLL_NEVER, (unsigned) n); + pollfd = pa_rtpoll_item_get_pollfd(item, NULL); + + if ((err = snd_pcm_poll_descriptors(pcm, pollfd, (unsigned) n)) < 0) { + pa_log("snd_pcm_poll_descriptors() failed: %s", pa_alsa_strerror(err)); + pa_rtpoll_item_free(item); + return NULL; + } + + return item; +} + +snd_pcm_sframes_t pa_alsa_safe_avail(snd_pcm_t *pcm, size_t hwbuf_size, const pa_sample_spec *ss) { + snd_pcm_sframes_t n; + size_t k; + + pa_assert(pcm); + pa_assert(hwbuf_size > 0); + pa_assert(ss); + + /* Some ALSA driver expose weird bugs, let's inform the user about + * what is going on */ + + n = snd_pcm_avail(pcm); + + if (n <= 0) + return n; + + k = (size_t) n * pa_frame_size(ss); + + if (PA_UNLIKELY(k >= hwbuf_size * 5 || + k >= pa_bytes_per_second(ss)*10)) { + + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(pcm); + pa_log_debug(ngettext("snd_pcm_avail() returned a value that is exceptionally large: %lu byte (%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + "snd_pcm_avail() returned a value that is exceptionally large: %lu bytes (%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + (unsigned long) k), + (unsigned long) k, + (unsigned long) (pa_bytes_to_usec(k, ss) / PA_USEC_PER_MSEC), + pa_strnull(dn)); + pa_xfree(dn); + pa_alsa_dump(PA_LOG_DEBUG, pcm); + } PA_ONCE_END; + + /* Mhmm, let's try not to fail completely */ + n = (snd_pcm_sframes_t) (hwbuf_size / pa_frame_size(ss)); + } + + return n; +} + +int pa_alsa_safe_delay(snd_pcm_t *pcm, snd_pcm_status_t *status, snd_pcm_sframes_t *delay, size_t hwbuf_size, const pa_sample_spec *ss, + bool capture) { + ssize_t k; + size_t abs_k; + int err; + snd_pcm_sframes_t avail = 0; +#if (SND_LIB_VERSION >= ((1<<16)|(1<<8)|0)) /* API additions in 1.1.0 */ + snd_pcm_audio_tstamp_config_t tstamp_config; +#endif + + pa_assert(pcm); + pa_assert(delay); + pa_assert(hwbuf_size > 0); + pa_assert(ss); + + /* Some ALSA driver expose weird bugs, let's inform the user about + * what is going on. We're going to get both the avail and delay values so + * that we can compare and check them for capture. + * This is done with snd_pcm_status() which provides + * avail, delay and timestamp values in a single kernel call to improve + * timer-based scheduling */ + +#if (SND_LIB_VERSION >= ((1<<16)|(1<<8)|0)) /* API additions in 1.1.0 */ + + /* The time stamp configuration needs to be set so that the + * ALSA code will use the internal delay reported by the driver. + * The time stamp configuration was introduced in alsa version 1.1.0. */ + tstamp_config.type_requested = 1; /* ALSA default time stamp type */ + tstamp_config.report_delay = 1; + snd_pcm_status_set_audio_htstamp_config(status, &tstamp_config); +#endif + + if ((err = snd_pcm_status(pcm, status)) < 0) + return err; + + avail = snd_pcm_status_get_avail(status); + *delay = snd_pcm_status_get_delay(status); + + k = (ssize_t) *delay * (ssize_t) pa_frame_size(ss); + + abs_k = k >= 0 ? (size_t) k : (size_t) -k; + + if (PA_UNLIKELY(abs_k >= hwbuf_size * 5 || + abs_k >= pa_bytes_per_second(ss)*10)) { + + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(pcm); + pa_log_debug(ngettext("snd_pcm_delay() returned a value that is exceptionally large: %li byte (%s%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + "snd_pcm_delay() returned a value that is exceptionally large: %li bytes (%s%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + (signed long) k), + (signed long) k, + k < 0 ? "-" : "", + (unsigned long) (pa_bytes_to_usec(abs_k, ss) / PA_USEC_PER_MSEC), + pa_strnull(dn)); + pa_xfree(dn); + pa_alsa_dump(PA_LOG_DEBUG, pcm); + } PA_ONCE_END; + + /* Mhmm, let's try not to fail completely */ + if (k < 0) + *delay = -(snd_pcm_sframes_t) (hwbuf_size / pa_frame_size(ss)); + else + *delay = (snd_pcm_sframes_t) (hwbuf_size / pa_frame_size(ss)); + } + + if (capture) { + abs_k = (size_t) avail * pa_frame_size(ss); + + if (PA_UNLIKELY(abs_k >= hwbuf_size * 5 || + abs_k >= pa_bytes_per_second(ss)*10)) { + + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(pcm); + pa_log_debug(ngettext("snd_pcm_avail() returned a value that is exceptionally large: %lu byte (%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + "snd_pcm_avail() returned a value that is exceptionally large: %lu bytes (%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + (unsigned long) k), + (unsigned long) k, + (unsigned long) (pa_bytes_to_usec(k, ss) / PA_USEC_PER_MSEC), + pa_strnull(dn)); + pa_xfree(dn); + pa_alsa_dump(PA_LOG_DEBUG, pcm); + } PA_ONCE_END; + + /* Mhmm, let's try not to fail completely */ + avail = (snd_pcm_sframes_t) (hwbuf_size / pa_frame_size(ss)); + } + + if (PA_UNLIKELY(*delay < avail)) { + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(pcm); + pa_log(_("snd_pcm_avail_delay() returned strange values: delay %lu is less than avail %lu.\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers."), + (unsigned long) *delay, + (unsigned long) avail, + pa_strnull(dn)); + pa_xfree(dn); + pa_alsa_dump(PA_LOG_ERROR, pcm); + } PA_ONCE_END; + + /* try to fixup */ + *delay = avail; + } + } + + return 0; +} + +int pa_alsa_safe_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames, size_t hwbuf_size, const pa_sample_spec *ss) { + int r; + snd_pcm_uframes_t before; + size_t k; + + pa_assert(pcm); + pa_assert(areas); + pa_assert(offset); + pa_assert(frames); + pa_assert(hwbuf_size > 0); + pa_assert(ss); + + before = *frames; + + r = snd_pcm_mmap_begin(pcm, areas, offset, frames); + + if (r < 0) + return r; + + k = (size_t) *frames * pa_frame_size(ss); + + if (PA_UNLIKELY(*frames > before || + k >= hwbuf_size * 3 || + k >= pa_bytes_per_second(ss)*10)) + PA_ONCE_BEGIN { + char *dn = pa_alsa_get_driver_name_by_pcm(pcm); + pa_log_debug(ngettext("snd_pcm_mmap_begin() returned a value that is exceptionally large: %lu byte (%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + "snd_pcm_mmap_begin() returned a value that is exceptionally large: %lu bytes (%lu ms).\n" + "Most likely this is a bug in the ALSA driver '%s'. Please report this issue to the ALSA developers.", + (unsigned long) k), + (unsigned long) k, + (unsigned long) (pa_bytes_to_usec(k, ss) / PA_USEC_PER_MSEC), + pa_strnull(dn)); + pa_xfree(dn); + pa_alsa_dump(PA_LOG_DEBUG, pcm); + } PA_ONCE_END; + + return r; +} + +char *pa_alsa_get_driver_name(int card) { + char *t, *m, *n; + + pa_assert(card >= 0); + + t = pa_sprintf_malloc("/sys/class/sound/card%i/device/driver/module", card); + m = pa_readlink(t); + pa_xfree(t); + + if (!m) + return NULL; + + n = pa_xstrdup(pa_path_get_filename(m)); + pa_xfree(m); + + return n; +} + +char *pa_alsa_get_driver_name_by_pcm(snd_pcm_t *pcm) { + int card; + snd_pcm_info_t* info; + snd_pcm_info_alloca(&info); + + pa_assert(pcm); + + if (snd_pcm_info(pcm, info) < 0) + return NULL; + + if ((card = snd_pcm_info_get_card(info)) < 0) + return NULL; + + return pa_alsa_get_driver_name(card); +} + +char *pa_alsa_get_reserve_name(const char *device) { + const char *t; + int i; + + pa_assert(device); + + if ((t = strchr(device, ':'))) + device = t+1; + + if ((i = snd_card_get_index(device)) < 0) { + int32_t k; + + if (pa_atoi(device, &k) < 0) + return NULL; + + i = (int) k; + } + + return pa_sprintf_malloc("Audio%i", i); +} + +unsigned int *pa_alsa_get_supported_rates(snd_pcm_t *pcm, unsigned int fallback_rate) { + static unsigned int all_rates[] = { 8000, 11025, 12000, + 16000, 22050, 24000, + 32000, 44100, 48000, + 64000, 88200, 96000, + 128000, 176400, 192000, + 384000 }; + bool supported[PA_ELEMENTSOF(all_rates)] = { false, }; + snd_pcm_hw_params_t *hwparams; + unsigned int i, j, n, *rates = NULL; + int ret; + + snd_pcm_hw_params_alloca(&hwparams); + + if ((ret = snd_pcm_hw_params_any(pcm, hwparams)) < 0) { + pa_log_debug("snd_pcm_hw_params_any() failed: %s", pa_alsa_strerror(ret)); + return NULL; + } + + for (i = 0, n = 0; i < PA_ELEMENTSOF(all_rates); i++) { + if (snd_pcm_hw_params_test_rate(pcm, hwparams, all_rates[i], 0) == 0) { + supported[i] = true; + n++; + } + } + + if (n > 0) { + rates = pa_xnew(unsigned int, n + 1); + + for (i = 0, j = 0; i < PA_ELEMENTSOF(all_rates); i++) { + if (supported[i]) + rates[j++] = all_rates[i]; + } + + rates[j] = 0; + } else { + rates = pa_xnew(unsigned int, 2); + + rates[0] = fallback_rate; + if ((ret = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rates[0], NULL)) < 0) { + pa_log_debug("snd_pcm_hw_params_set_rate_near() failed: %s", pa_alsa_strerror(ret)); + pa_xfree(rates); + return NULL; + } + + rates[1] = 0; + } + + return rates; +} + +pa_sample_format_t *pa_alsa_get_supported_formats(snd_pcm_t *pcm, pa_sample_format_t fallback_format) { + static const snd_pcm_format_t format_trans_to_pa[] = { + [SND_PCM_FORMAT_U8] = PA_SAMPLE_U8, + [SND_PCM_FORMAT_A_LAW] = PA_SAMPLE_ALAW, + [SND_PCM_FORMAT_MU_LAW] = PA_SAMPLE_ULAW, + [SND_PCM_FORMAT_S16_LE] = PA_SAMPLE_S16LE, + [SND_PCM_FORMAT_S16_BE] = PA_SAMPLE_S16BE, + [SND_PCM_FORMAT_FLOAT_LE] = PA_SAMPLE_FLOAT32LE, + [SND_PCM_FORMAT_FLOAT_BE] = PA_SAMPLE_FLOAT32BE, + [SND_PCM_FORMAT_S32_LE] = PA_SAMPLE_S32LE, + [SND_PCM_FORMAT_S32_BE] = PA_SAMPLE_S32BE, + [SND_PCM_FORMAT_S24_3LE] = PA_SAMPLE_S24LE, + [SND_PCM_FORMAT_S24_3BE] = PA_SAMPLE_S24BE, + [SND_PCM_FORMAT_S24_LE] = PA_SAMPLE_S24_32LE, + [SND_PCM_FORMAT_S24_BE] = PA_SAMPLE_S24_32BE, + }; + static const snd_pcm_format_t all_formats[] = { + SND_PCM_FORMAT_U8, + SND_PCM_FORMAT_A_LAW, + SND_PCM_FORMAT_MU_LAW, + SND_PCM_FORMAT_S16_LE, + SND_PCM_FORMAT_S16_BE, + SND_PCM_FORMAT_FLOAT_LE, + SND_PCM_FORMAT_FLOAT_BE, + SND_PCM_FORMAT_S32_LE, + SND_PCM_FORMAT_S32_BE, + SND_PCM_FORMAT_S24_3LE, + SND_PCM_FORMAT_S24_3BE, + SND_PCM_FORMAT_S24_LE, + SND_PCM_FORMAT_S24_BE, + }; + bool supported[PA_ELEMENTSOF(all_formats)] = { + false, + }; + snd_pcm_hw_params_t *hwparams; + unsigned int i, j, n; + pa_sample_format_t *formats = NULL; + int ret; + + snd_pcm_hw_params_alloca(&hwparams); + + if ((ret = snd_pcm_hw_params_any(pcm, hwparams)) < 0) { + pa_log_debug("snd_pcm_hw_params_any() failed: %s", pa_alsa_strerror(ret)); + return NULL; + } + + for (i = 0, n = 0; i < PA_ELEMENTSOF(all_formats); i++) { + if (snd_pcm_hw_params_test_format(pcm, hwparams, all_formats[i]) == 0) { + supported[i] = true; + n++; + } + } + + if (n > 0) { + formats = pa_xnew(pa_sample_format_t, n + 1); + + for (i = 0, j = 0; i < PA_ELEMENTSOF(all_formats); i++) { + if (supported[i]) + formats[j++] = format_trans_to_pa[all_formats[i]]; + } + + formats[j] = PA_SAMPLE_MAX; + } else { + formats = pa_xnew(pa_sample_format_t, 2); + + formats[0] = fallback_format; + if ((ret = snd_pcm_hw_params_set_format(pcm, hwparams, format_trans_to_pa[formats[0]])) < 0) { + pa_log_debug("snd_pcm_hw_params_set_format() failed: %s", pa_alsa_strerror(ret)); + pa_xfree(formats); + return NULL; + } + + formats[1] = PA_SAMPLE_MAX; + } + + return formats; +} + +bool pa_alsa_pcm_is_hw(snd_pcm_t *pcm) { + snd_pcm_info_t* info; + snd_pcm_info_alloca(&info); + + pa_assert(pcm); + + if (snd_pcm_info(pcm, info) < 0) + return false; + + return snd_pcm_info_get_card(info) >= 0; +} + +bool pa_alsa_pcm_is_modem(snd_pcm_t *pcm) { + snd_pcm_info_t* info; + snd_pcm_info_alloca(&info); + + pa_assert(pcm); + + if (snd_pcm_info(pcm, info) < 0) + return false; + + return snd_pcm_info_get_class(info) == SND_PCM_CLASS_MODEM; +} + +PA_STATIC_TLS_DECLARE(cstrerror, pa_xfree); + +const char* pa_alsa_strerror(int errnum) { + const char *original = NULL; + char *translated, *t; + char errbuf[128]; + + if ((t = PA_STATIC_TLS_GET(cstrerror))) + pa_xfree(t); + + original = snd_strerror(errnum); + + if (!original) { + pa_snprintf(errbuf, sizeof(errbuf), "Unknown error %i", errnum); + original = errbuf; + } + + if (!(translated = pa_locale_to_utf8(original))) { + pa_log_warn("Unable to convert error string to locale, filtering."); + translated = pa_utf8_filter(original); + } + + PA_STATIC_TLS_SET(cstrerror, translated); + + return translated; +} + +bool pa_alsa_may_tsched(bool want) { + + if (!want) + return false; + + if (!pa_rtclock_hrtimer()) { + /* We cannot depend on being woken up in time when the timers + are inaccurate, so let's fallback to classic IO based playback + then. */ + pa_log_notice("Disabling timer-based scheduling because high-resolution timers are not available from the kernel."); + return false; } + + if (pa_running_in_vm()) { + /* We cannot depend on being woken up when we ask for in a VM, + * so let's fallback to classic IO based playback then. */ + pa_log_notice("Disabling timer-based scheduling because running inside a VM."); + return false; + } + + return true; +} + +#define SND_MIXER_ELEM_PULSEAUDIO (SND_MIXER_ELEM_LAST + 10) + +static snd_mixer_elem_t *pa_alsa_mixer_find(snd_mixer_t *mixer, + snd_ctl_elem_iface_t iface, + const char *name, + unsigned int index, + unsigned int device) { + snd_mixer_elem_t *elem; + + for (elem = snd_mixer_first_elem(mixer); elem; elem = snd_mixer_elem_next(elem)) { + snd_hctl_elem_t *helem; + if (snd_mixer_elem_get_type(elem) != SND_MIXER_ELEM_PULSEAUDIO) + continue; + helem = snd_mixer_elem_get_private(elem); + if (snd_hctl_elem_get_interface(helem) != iface) + continue; + if (!pa_streq(snd_hctl_elem_get_name(helem), name)) + continue; + if (snd_hctl_elem_get_index(helem) != index) + continue; + if (snd_hctl_elem_get_device(helem) != device) + continue; + return elem; + } + return NULL; +} + +snd_mixer_elem_t *pa_alsa_mixer_find_card(snd_mixer_t *mixer, struct pa_alsa_mixer_id *alsa_id, unsigned int device) { + return pa_alsa_mixer_find(mixer, SND_CTL_ELEM_IFACE_CARD, alsa_id->name, alsa_id->index, device); +} + +snd_mixer_elem_t *pa_alsa_mixer_find_pcm(snd_mixer_t *mixer, const char *name, unsigned int device) { + return pa_alsa_mixer_find(mixer, SND_CTL_ELEM_IFACE_PCM, name, 0, device); +} + +static int mixer_class_compare(const snd_mixer_elem_t *c1, const snd_mixer_elem_t *c2) +{ + /* Dummy compare function */ + return c1 == c2 ? 0 : (c1 > c2 ? 1 : -1); +} + +static int mixer_class_event(snd_mixer_class_t *class, unsigned int mask, + snd_hctl_elem_t *helem, snd_mixer_elem_t *melem) +{ + int err; + const char *name = snd_hctl_elem_get_name(helem); + if (mask & SND_CTL_EVENT_MASK_ADD) { + snd_ctl_elem_iface_t iface = snd_hctl_elem_get_interface(helem); + if (iface == SND_CTL_ELEM_IFACE_CARD || iface == SND_CTL_ELEM_IFACE_PCM) { + snd_mixer_elem_t *new_melem; + + /* Put the hctl pointer as our private data - it will be useful for callbacks */ + if ((err = snd_mixer_elem_new(&new_melem, SND_MIXER_ELEM_PULSEAUDIO, 0, helem, NULL)) < 0) { + pa_log_warn("snd_mixer_elem_new failed: %s", pa_alsa_strerror(err)); + return 0; + } + + if ((err = snd_mixer_elem_attach(new_melem, helem)) < 0) { + pa_log_warn("snd_mixer_elem_attach failed: %s", pa_alsa_strerror(err)); + snd_mixer_elem_free(melem); + return 0; + } + + if ((err = snd_mixer_elem_add(new_melem, class)) < 0) { + pa_log_warn("snd_mixer_elem_add failed: %s", pa_alsa_strerror(err)); + return 0; + } + } + } + else if (mask & SND_CTL_EVENT_MASK_VALUE) { + snd_mixer_elem_value(melem); /* Calls the element callback */ + return 0; + } + else + pa_log_info("Got an unknown mixer class event for %s: mask 0x%x", name, mask); + + return 0; +} + +static int prepare_mixer(snd_mixer_t *mixer, const char *dev) { + int err; + snd_mixer_class_t *class; + + pa_assert(mixer); + pa_assert(dev); + + if ((err = snd_mixer_attach(mixer, dev)) < 0) { + pa_log_info("Unable to attach to mixer %s: %s", dev, pa_alsa_strerror(err)); + return -1; + } + + if (snd_mixer_class_malloc(&class)) { + pa_log_info("Failed to allocate mixer class for %s", dev); + return -1; + } + snd_mixer_class_set_event(class, mixer_class_event); + snd_mixer_class_set_compare(class, mixer_class_compare); + if ((err = snd_mixer_class_register(class, mixer)) < 0) { + pa_log_info("Unable register mixer class for %s: %s", dev, pa_alsa_strerror(err)); + snd_mixer_class_free(class); + return -1; + } + /* From here on, the mixer class is deallocated by alsa on snd_mixer_close/free. */ + + if ((err = snd_mixer_selem_register(mixer, NULL, NULL)) < 0) { + pa_log_warn("Unable to register mixer: %s", pa_alsa_strerror(err)); + return -1; + } + + if ((err = snd_mixer_load(mixer)) < 0) { + pa_log_warn("Unable to load mixer: %s", pa_alsa_strerror(err)); + return -1; + } + + pa_log_info("Successfully attached to mixer '%s'", dev); + return 0; +} + +snd_mixer_t *pa_alsa_open_mixer(pa_hashmap *mixers, int alsa_card_index, bool probe) { + char *md = pa_sprintf_malloc("hw:%i", alsa_card_index); + snd_mixer_t *m = pa_alsa_open_mixer_by_name(mixers, md, probe); + pa_xfree(md); + return m; +} + +snd_mixer_t *pa_alsa_open_mixer_by_name(pa_hashmap *mixers, const char *dev, bool probe) { + int err; + snd_mixer_t *m; + pa_alsa_mixer *pm; + char *dev2; + void *state; + + pa_assert(mixers); + pa_assert(dev); + + pm = pa_hashmap_get(mixers, dev); + + /* The quick card number/index lookup (hw:#) + * We already know the card number/index, thus use the mixer + * from the cache at first. + */ + if (!pm && pa_strneq(dev, "hw:", 3)) { + const char *s = dev + 3; + int card_index; + while (*s && *s >= '0' && *s <= '9') s++; + if (*s == '\0' && pa_atoi(dev + 3, &card_index) >= 0) { + PA_HASHMAP_FOREACH_KV(dev2, pm, mixers, state) { + if (pm->card_index == card_index) { + dev = dev2; + pm = pa_hashmap_get(mixers, dev); + break; + } + } + } + } + + if (pm) { + if (!probe) + pm->used_for_probe_only = false; + return pm->mixer_handle; + } + + if ((err = snd_mixer_open(&m, 0)) < 0) { + pa_log("Error opening mixer: %s", pa_alsa_strerror(err)); + return NULL; + } + + if (prepare_mixer(m, dev) >= 0) { + pm = pa_xnew0(pa_alsa_mixer, 1); + if (pm) { + snd_hctl_t *hctl; + pm->card_index = -1; + /* determine the ALSA card number (index) and store it to card_index */ + err = snd_mixer_get_hctl(m, dev, &hctl); + if (err >= 0) { + snd_ctl_card_info_t *info; + snd_ctl_card_info_alloca(&info); + err = snd_ctl_card_info(snd_hctl_ctl(hctl), info); + if (err >= 0) + pm->card_index = snd_ctl_card_info_get_card(info); + } + pm->used_for_probe_only = probe; + pm->mixer_handle = m; + pa_hashmap_put(mixers, pa_xstrdup(dev), pm); + return m; + } + } + + snd_mixer_close(m); + return NULL; +} + +snd_mixer_t *pa_alsa_open_mixer_for_pcm(pa_hashmap *mixers, snd_pcm_t *pcm, bool probe) { + snd_pcm_info_t* info; + snd_pcm_info_alloca(&info); + + pa_assert(pcm); + + if (snd_pcm_info(pcm, info) >= 0) { + int card_idx; + + if ((card_idx = snd_pcm_info_get_card(info)) >= 0) + return pa_alsa_open_mixer(mixers, card_idx, probe); + } + + return NULL; +} + +void pa_alsa_mixer_set_fdlist(pa_hashmap *mixers, snd_mixer_t *mixer_handle, pa_mainloop_api *ml) +{ + pa_alsa_mixer *pm; + void *state; + + PA_HASHMAP_FOREACH(pm, mixers, state) + if (pm->mixer_handle == mixer_handle) { + pm->used_for_probe_only = false; + if (!pm->fdl) { + pm->fdl = pa_alsa_fdlist_new(); + if (pm->fdl) + pa_alsa_fdlist_set_handle(pm->fdl, pm->mixer_handle, NULL, ml); + } + } +} + +void pa_alsa_mixer_free(pa_alsa_mixer *mixer) +{ + if (mixer->fdl) + pa_alsa_fdlist_free(mixer->fdl); + if (mixer->mixer_handle) + snd_mixer_close(mixer->mixer_handle); + pa_xfree(mixer); +} + +int pa_alsa_get_hdmi_eld(snd_hctl_elem_t *elem, pa_hdmi_eld *eld) { + + /* The ELD format is specific to HDA Intel sound cards and defined in the + HDA specification: http://www.intel.com/content/www/us/en/standards/high-definition-audio-specification.html */ + int err; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *value; + uint8_t *elddata; + unsigned int eldsize, mnl; + unsigned int device; + + pa_assert(eld != NULL); + pa_assert(elem != NULL); + + /* Does it have any contents? */ + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&value); + if ((err = snd_hctl_elem_info(elem, info)) < 0 || + (err = snd_hctl_elem_read(elem, value)) < 0) { + pa_log_warn("Accessing ELD control failed with error %s", snd_strerror(err)); + return -1; + } + + device = snd_hctl_elem_get_device(elem); + eldsize = snd_ctl_elem_info_get_count(info); + elddata = (unsigned char *) snd_ctl_elem_value_get_bytes(value); + if (elddata == NULL || eldsize == 0) { + pa_log_debug("ELD info empty (for device=%d)", device); + return -1; + } + if (eldsize < 20 || eldsize > 256) { + pa_log_debug("ELD info has wrong size (for device=%d)", device); + return -1; + } + + /* Try to fetch monitor name */ + mnl = elddata[4] & 0x1f; + if (mnl == 0 || mnl > 16 || 20 + mnl > eldsize) { + pa_log_debug("No monitor name in ELD info (for device=%d)", device); + mnl = 0; + } + memcpy(eld->monitor_name, &elddata[20], mnl); + eld->monitor_name[mnl] = '\0'; + if (mnl) + pa_log_debug("Monitor name in ELD info is '%s' (for device=%d)", eld->monitor_name, device); + + return 0; +} diff --git a/src/modules/alsa/alsa-util.h b/src/modules/alsa/alsa-util.h new file mode 100644 index 0000000..2eed3ea --- /dev/null +++ b/src/modules/alsa/alsa-util.h @@ -0,0 +1,167 @@ +#ifndef fooalsautilhfoo +#define fooalsautilhfoo + +/*** + This file is part of PulseAudio. + + Copyright 2004-2006 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#include <alsa/asoundlib.h> + +#include <pulse/sample.h> +#include <pulse/channelmap.h> +#include <pulse/proplist.h> + +#include <pulsecore/rtpoll.h> +#include <pulsecore/core.h> +#include <pulsecore/log.h> + +#include "alsa-mixer.h" + +enum { + PA_ALSA_ERR_UNSPECIFIED = 1, + PA_ALSA_ERR_UCM_OPEN = 1000, + PA_ALSA_ERR_UCM_NO_VERB = 1001, + PA_ALSA_ERR_UCM_LINKED = 1002 +}; + +int pa_alsa_set_hw_params( + snd_pcm_t *pcm_handle, + pa_sample_spec *ss, /* modified at return */ + snd_pcm_uframes_t *period_size, /* modified at return */ + snd_pcm_uframes_t *buffer_size, /* modified at return */ + snd_pcm_uframes_t tsched_size, + bool *use_mmap, /* modified at return */ + bool *use_tsched, /* modified at return */ + bool require_exact_channel_number); + +int pa_alsa_set_sw_params( + snd_pcm_t *pcm, + snd_pcm_uframes_t avail_min, + bool period_event); + +/* Picks a working mapping from the profile set based on the specified ss/map */ +snd_pcm_t *pa_alsa_open_by_device_id_auto( + const char *dev_id, + char **dev, /* modified at return */ + pa_sample_spec *ss, /* modified at return */ + pa_channel_map* map, /* modified at return */ + int mode, + snd_pcm_uframes_t *period_size, /* modified at return */ + snd_pcm_uframes_t *buffer_size, /* modified at return */ + snd_pcm_uframes_t tsched_size, + bool *use_mmap, /* modified at return */ + bool *use_tsched, /* modified at return */ + pa_alsa_profile_set *ps, + pa_alsa_mapping **mapping); /* modified at return */ + +/* Uses the specified mapping */ +snd_pcm_t *pa_alsa_open_by_device_id_mapping( + const char *dev_id, + char **dev, /* modified at return */ + pa_sample_spec *ss, /* modified at return */ + pa_channel_map* map, /* modified at return */ + int mode, + snd_pcm_uframes_t *period_size, /* modified at return */ + snd_pcm_uframes_t *buffer_size, /* modified at return */ + snd_pcm_uframes_t tsched_size, + bool *use_mmap, /* modified at return */ + bool *use_tsched, /* modified at return */ + pa_alsa_mapping *mapping); + +/* Opens the explicit ALSA device */ +snd_pcm_t *pa_alsa_open_by_device_string( + const char *dir, + char **dev, /* modified at return */ + pa_sample_spec *ss, /* modified at return */ + pa_channel_map* map, /* modified at return */ + int mode, + snd_pcm_uframes_t *period_size, /* modified at return */ + snd_pcm_uframes_t *buffer_size, /* modified at return */ + snd_pcm_uframes_t tsched_size, + bool *use_mmap, /* modified at return */ + bool *use_tsched, /* modified at return */ + bool require_exact_channel_number); + +/* Opens the explicit ALSA device with a fallback list */ +snd_pcm_t *pa_alsa_open_by_template( + char **template, + const char *dev_id, + char **dev, /* modified at return */ + pa_sample_spec *ss, /* modified at return */ + pa_channel_map* map, /* modified at return */ + int mode, + snd_pcm_uframes_t *period_size, /* modified at return */ + snd_pcm_uframes_t *buffer_size, /* modified at return */ + snd_pcm_uframes_t tsched_size, + bool *use_mmap, /* modified at return */ + bool *use_tsched, /* modified at return */ + bool require_exact_channel_number); + +void pa_alsa_dump(pa_log_level_t level, snd_pcm_t *pcm); +void pa_alsa_dump_status(snd_pcm_t *pcm); + +void pa_alsa_refcnt_inc(void); +void pa_alsa_refcnt_dec(void); + +void pa_alsa_init_proplist_pcm_info(pa_core *c, pa_proplist *p, snd_pcm_info_t *pcm_info); +void pa_alsa_init_proplist_card(pa_core *c, pa_proplist *p, int card); +void pa_alsa_init_proplist_pcm(pa_core *c, pa_proplist *p, snd_pcm_t *pcm); +void pa_alsa_init_proplist_ctl(pa_proplist *p, const char *name); +bool pa_alsa_init_description(pa_proplist *p, pa_card *card); + +int pa_alsa_recover_from_poll(snd_pcm_t *pcm, int revents); + +pa_rtpoll_item* pa_alsa_build_pollfd(snd_pcm_t *pcm, pa_rtpoll *rtpoll); + +snd_pcm_sframes_t pa_alsa_safe_avail(snd_pcm_t *pcm, size_t hwbuf_size, const pa_sample_spec *ss); +int pa_alsa_safe_delay(snd_pcm_t *pcm, snd_pcm_status_t *status, snd_pcm_sframes_t *delay, size_t hwbuf_size, const pa_sample_spec *ss, bool capture); +int pa_alsa_safe_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames, size_t hwbuf_size, const pa_sample_spec *ss); + +char *pa_alsa_get_driver_name(int card); +char *pa_alsa_get_driver_name_by_pcm(snd_pcm_t *pcm); + +char *pa_alsa_get_reserve_name(const char *device); + +unsigned int *pa_alsa_get_supported_rates(snd_pcm_t *pcm, unsigned int fallback_rate); +pa_sample_format_t *pa_alsa_get_supported_formats(snd_pcm_t *pcm, pa_sample_format_t fallback_format); + +bool pa_alsa_pcm_is_hw(snd_pcm_t *pcm); +bool pa_alsa_pcm_is_modem(snd_pcm_t *pcm); + +const char* pa_alsa_strerror(int errnum); + +bool pa_alsa_may_tsched(bool want); + +snd_mixer_elem_t *pa_alsa_mixer_find_card(snd_mixer_t *mixer, struct pa_alsa_mixer_id *alsa_id, unsigned int device); +snd_mixer_elem_t *pa_alsa_mixer_find_pcm(snd_mixer_t *mixer, const char *name, unsigned int device); + +snd_mixer_t *pa_alsa_open_mixer(pa_hashmap *mixers, int alsa_card_index, bool probe); +snd_mixer_t *pa_alsa_open_mixer_by_name(pa_hashmap *mixers, const char *dev, bool probe); +snd_mixer_t *pa_alsa_open_mixer_for_pcm(pa_hashmap *mixers, snd_pcm_t *pcm, bool probe); +void pa_alsa_mixer_set_fdlist(pa_hashmap *mixers, snd_mixer_t *mixer, pa_mainloop_api *ml); +void pa_alsa_mixer_free(pa_alsa_mixer *mixer); + +typedef struct pa_hdmi_eld pa_hdmi_eld; +struct pa_hdmi_eld { + char monitor_name[17]; +}; + +int pa_alsa_get_hdmi_eld(snd_hctl_elem_t *elem, pa_hdmi_eld *eld); + +#endif diff --git a/src/modules/alsa/meson.build b/src/modules/alsa/meson.build new file mode 100644 index 0000000..f31eeb5 --- /dev/null +++ b/src/modules/alsa/meson.build @@ -0,0 +1,51 @@ +libalsa_util_sources = [ + 'alsa-util.c', + 'alsa-ucm.c', + 'alsa-mixer.c', + 'alsa-sink.c', + 'alsa-source.c', + '../reserve-wrap.c', +] + +libalsa_util_headers = [ + 'alsa-util.h', + 'alsa-ucm.h', + 'alsa-mixer.h', + 'alsa-sink.h', + 'alsa-source.h', + '../reserve-wrap.h', +] + +if dbus_dep.found() + libalsa_util_sources += [ '../reserve.c', '../reserve-monitor.c' ] + libalsa_util_headers += [ '../reserve.h', '../reserve-monitor.h' ] +endif + +if udev_dep.found() + libalsa_util_sources += [ '../udev-util.c' ] + libalsa_util_headers += [ '../udev-util.h' ] +endif + +libalsa_util = shared_library('alsa-util', + libalsa_util_sources, + libalsa_util_headers, + c_args : [pa_c_args, server_c_args], + link_args : [nodelete_link_args], + include_directories : [configinc, topinc], + dependencies : [libpulse_dep, libpulsecommon_dep, libpulsecore_dep, alsa_dep, dbus_dep, libatomic_ops_dep, libm_dep, udev_dep, libintl_dep], + install : true, + install_rpath : privlibdir, + install_dir : modlibexecdir, +) + +alsa_udevrules = [ + '90-pulseaudio.rules', +] + +if udev_dep.found() + install_data(alsa_udevrules, + install_dir : udevrulesdir, + ) +endif + +subdir('mixer') diff --git a/src/modules/alsa/mixer/meson.build b/src/modules/alsa/mixer/meson.build new file mode 100644 index 0000000..d4327b8 --- /dev/null +++ b/src/modules/alsa/mixer/meson.build @@ -0,0 +1,7 @@ +install_subdir('paths', + install_dir : alsadatadir +) + +install_subdir('profile-sets', + install_dir : alsadatadir +) diff --git a/src/modules/alsa/mixer/paths/analog-input-aux.conf b/src/modules/alsa/mixer/paths/analog-input-aux.conf new file mode 100644 index 0000000..47e22c5 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-aux.conf @@ -0,0 +1,65 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where an 'Aux' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 80 +description-key = analog-input + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Line] +switch = off +volume = off + +[Element Aux] +required = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Video] +switch = off +volume = off + +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +switch = off +volume = off + +.include analog-input.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-dock-mic.conf b/src/modules/alsa/mixer/paths/analog-input-dock-mic.conf new file mode 100644 index 0000000..96861e7 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-dock-mic.conf @@ -0,0 +1,104 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Dock Mic' or 'Dock Mic Boost' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 78 +description-key = analog-input-microphone-dock + +[Jack Dock Mic] +required-any = any + +[Jack Dock Mic Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Dock Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Dock Mic Boost:on] +name = input-boost-on + +[Option Dock Mic Boost:off] +name = input-boost-off + +[Element Dock Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Dock Mic] +name = analog-input-microphone-dock +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Dock Mic] +name = analog-input-microphone-dock +required-any = any + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Front Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Mic Boost] +switch = off +volume = off + +[Element Internal Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-fm.conf b/src/modules/alsa/mixer/paths/analog-input-fm.conf new file mode 100644 index 0000000..d3501a8 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-fm.conf @@ -0,0 +1,65 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where an 'FM' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 70 +description-key = analog-input-radio + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Line] +switch = off +volume = off + +[Element Aux] +switch = off +volume = off + +[Element Video] +switch = off +volume = off + +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +required = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +.include analog-input.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-front-mic.conf b/src/modules/alsa/mixer/paths/analog-input-front-mic.conf new file mode 100644 index 0000000..6e7775c --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-front-mic.conf @@ -0,0 +1,104 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Front Mic' or 'Front Mic Boost' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 85 +description-key = analog-input-microphone-front + +[Jack Front Mic] +required-any = any + +[Jack Front Mic Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Front Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Front Mic Boost:on] +name = input-boost-on + +[Option Front Mic Boost:off] +name = input-boost-off + +[Element Front Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Front Mic] +name = analog-input-microphone-front +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Front Mic] +name = analog-input-microphone-front +required-any = any + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Mic Boost] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Internal Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-headphone-mic.conf b/src/modules/alsa/mixer/paths/analog-input-headphone-mic.conf new file mode 100644 index 0000000..eb5740a --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-headphone-mic.conf @@ -0,0 +1,102 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For some ASUS netbooks that have one jack that can be either a Headphone +; *or* a mic. This path will be active only when it is used as a mic. +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 87 +description-key = analog-input-microphone + +[Jack Headphone Mic] +required-any = any +state.plugged = unknown + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Headphone Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Headphone Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Headphone Mic] +name = analog-input-microphone +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Headphone Mic] +name = analog-input-microphone +required-any = any + +; Make sure the internal speakers are not auto-muted when you plug a mic in +[Element Auto-Mute Mode] +enumeration = select + +[Option Auto-Mute Mode:Disabled] +name = analog-input-microphone + +[Element Front Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Internal Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-headset-mic.conf b/src/modules/alsa/mixer/paths/analog-input-headset-mic.conf new file mode 100644 index 0000000..579db6b --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-headset-mic.conf @@ -0,0 +1,114 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Headset Mic' or 'Headset Mic Boost' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 88 +description-key = analog-input-microphone-headset + +[Jack Headset Mic] +required-any = any + +[Jack Headset Mic Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Headphone] +state.plugged = unknown + +[Jack Front Headphone] +state.plugged = unknown + +[Jack Headphone Mic] +state.plugged = unknown + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Headset Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Headset Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Headset] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Headset Mic] +name = Headset Microphone +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Headset Mic] +name = Headset Microphone +required-any = any + +[Element Front Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Internal Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-internal-mic-always.conf b/src/modules/alsa/mixer/paths/analog-input-internal-mic-always.conf new file mode 100644 index 0000000..9e22008 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-internal-mic-always.conf @@ -0,0 +1,133 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Internal Mic' or 'Internal Mic Boost' element exists +; 'Int Mic' and 'Int Mic Boost' are for compatibility with kernels < 2.6.38 +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 89 +description-key = analog-input-microphone-internal + +[Jack Mic] +state.plugged = no +state.unplugged = unknown + +[Jack Dock Mic] +state.plugged = no +state.unplugged = unknown + +[Jack Front Mic] +state.plugged = no +state.unplugged = unknown + +[Jack Rear Mic] +state.plugged = no +state.unplugged = unknown + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Internal Mic Boost] +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Internal Mic Boost:on] +name = input-boost-on + +[Option Internal Mic Boost:off] +name = input-boost-off + +[Element Int Mic Boost] +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Int Mic Boost:on] +name = input-boost-on + +[Option Int Mic Boost:off] +name = input-boost-off + +[Element Internal Mic] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Int Mic] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Internal Mic] +name = analog-input-microphone-internal + +[Option Input Source:Int Mic] +name = analog-input-microphone-internal + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Internal Mic] +name = analog-input-microphone-internal + +[Option Capture Source:Int Mic] +name = analog-input-microphone-internal + +[Element Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Front Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Mic Boost] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-internal-mic.conf b/src/modules/alsa/mixer/paths/analog-input-internal-mic.conf new file mode 100644 index 0000000..898410a --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-internal-mic.conf @@ -0,0 +1,154 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Internal Mic' or 'Internal Mic Boost' element exists +; 'Int Mic' and 'Int Mic Boost' are for compatibility with kernels < 2.6.38 +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 89 +description-key = analog-input-microphone-internal + +[Jack Mic] +state.plugged = no +state.unplugged = unknown + +[Jack Dock Mic] +state.plugged = no +state.unplugged = unknown + +[Jack Front Mic] +state.plugged = no +state.unplugged = unknown + +[Jack Rear Mic] +state.plugged = no +state.unplugged = unknown + +[Jack Internal Mic Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Internal Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Internal Mic Boost:on] +name = input-boost-on + +[Option Internal Mic Boost:off] +name = input-boost-off + +[Element Int Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Int Mic Boost:on] +name = input-boost-on + +[Option Int Mic Boost:off] +name = input-boost-off + +[Element Internal Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Int Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Internal Mic] +name = analog-input-microphone-internal +required-any = any + +[Option Input Source:Int Mic] +name = analog-input-microphone-internal +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Internal Mic] +name = analog-input-microphone-internal +required-any = any + +[Option Capture Source:Int Mic] +name = analog-input-microphone-internal +required-any = any + +[Element Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Front Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Headphone Mic] +switch = off +volume = off + +[Element Headphone Mic Boost] +switch = off +volume = off + +[Element Mic Boost] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-linein.conf b/src/modules/alsa/mixer/paths/analog-input-linein.conf new file mode 100644 index 0000000..cf20790 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-linein.conf @@ -0,0 +1,144 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Line' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 81 + +[Jack Line] +required-any = any + +[Jack Line Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Jack Line - Input] +required-any = any + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Line Boost] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Line] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Line] +name = analog-input-linein +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Line] +name = analog-input-linein +required-any = any + +[Element PCM Capture Source] +enumeration = select + +[Option PCM Capture Source:Line] +name = analog-input-linein +required-any = any + +[Option PCM Capture Source:Line In] +name = analog-input-linein +required-any = any + +[Element Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Front Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Mic Boost] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Internal Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +[Element Aux] +switch = off +volume = off + +[Element Video] +switch = off +volume = off + +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +switch = off +volume = off + +[Element Mic Jack Mode] +enumeration = select + +[Option Mic Jack Mode:Line In] +priority = 19 +name = input-linein diff --git a/src/modules/alsa/mixer/paths/analog-input-mic-line.conf b/src/modules/alsa/mixer/paths/analog-input-mic-line.conf new file mode 100644 index 0000000..7147d20 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-mic-line.conf @@ -0,0 +1,66 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Mic/Line' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 85 +description-key = analog-input + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Line] +switch = off +volume = off + +[Element Aux] +switch = off +volume = off + +[Element Video] +switch = off +volume = off + +[Element Mic/Line] +required = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +switch = off +volume = off + +.include analog-input.conf.common +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-mic.conf b/src/modules/alsa/mixer/paths/analog-input-mic.conf new file mode 100644 index 0000000..53c03c8 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-mic.conf @@ -0,0 +1,141 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Mic' or 'Mic Boost' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 87 +description-key = analog-input-microphone + +[Jack Mic] +required-any = any + +[Jack Mic Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Jack Mic - Input] +required-any = any + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Mic Boost:on] +name = input-boost-on + +[Option Mic Boost:off] +name = input-boost-off + +[Element Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Mic] +name = analog-input-microphone +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Mic] +name = analog-input-microphone +required-any = any + +[Element PCM Capture Source] +enumeration = select + +[Option PCM Capture Source:Mic] +name = analog-input-microphone +required-any = any + +[Option PCM Capture Source:Mic-In/Mic Array] +name = analog-input-microphone +required-any = any + +;;; Some AC'97s have "Mic Select" and "Mic Boost (+20dB)" + +[Element Mic Select] +enumeration = select + +[Option Mic Select:Mic1] +name = input-microphone +priority = 20 + +[Option Mic Select:Mic2] +name = input-microphone +priority = 19 + +[Element Mic Boost (+20dB)] +switch = select +volume = merge + +[Option Mic Boost (+20dB):on] +name = input-boost-on + +[Option Mic Boost (+20dB):off] +name = input-boost-off + +[Element Front Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Rear Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Internal Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +[Element Rear Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-mic.conf.common b/src/modules/alsa/mixer/paths/analog-input-mic.conf.common new file mode 100644 index 0000000..e5ced21 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-mic.conf.common @@ -0,0 +1,60 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Common element for all microphone inputs +; +; See analog-output.conf.common for an explanation on the directives + +[Properties] +device.icon_name = audio-input-microphone + +[Element Line] +switch = off +volume = off + +[Element Line Boost] +switch = off +volume = off + +[Element Aux] +switch = off +volume = off + +[Element Video] +switch = off +volume = off + +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +switch = off +volume = off + +[Element Inverted Internal Mic] +switch = off +volume = off + +[Element Mic Jack Mode] +enumeration = select + +[Option Mic Jack Mode:Mic In] +priority = 19 +name = input-microphone diff --git a/src/modules/alsa/mixer/paths/analog-input-rear-mic.conf b/src/modules/alsa/mixer/paths/analog-input-rear-mic.conf new file mode 100644 index 0000000..7136193 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-rear-mic.conf @@ -0,0 +1,104 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Rear Mic' or 'Rear Mic Boost' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 82 +description-key = analog-input-microphone-rear + +[Jack Rear Mic] +required-any = any + +[Jack Rear Mic Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Rear Mic Boost] +required-any = any +switch = select +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Option Rear Mic Boost:on] +name = input-boost-on + +[Option Rear Mic Boost:off] +name = input-boost-off + +[Element Rear Mic] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Input Source] +enumeration = select + +[Option Input Source:Rear Mic] +name = analog-input-microphone-rear +required-any = any + +[Element Capture Source] +enumeration = select + +[Option Capture Source:Rear Mic] +name = analog-input-microphone-rear +required-any = any + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Front Mic] +switch = off +volume = off + +[Element Dock Mic] +switch = off +volume = off + +[Element Mic Boost] +switch = off +volume = off + +[Element Dock Mic Boost] +switch = off +volume = off + +[Element Internal Mic Boost] +switch = off +volume = off + +[Element Front Mic Boost] +switch = off +volume = off + +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf b/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf new file mode 100644 index 0000000..99d1d79 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf @@ -0,0 +1,65 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'TV Tuner' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 70 +description-key = analog-input-video + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Line] +switch = off +volume = off + +[Element Aux] +switch = off +volume = off + +[Element Video] +switch = off +volume = off + +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +required = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element FM] +switch = off +volume = off + +.include analog-input.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-video.conf b/src/modules/alsa/mixer/paths/analog-input-video.conf new file mode 100644 index 0000000..50c999e --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input-video.conf @@ -0,0 +1,64 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; For devices where a 'Video' element exists +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 70 + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic] +switch = off +volume = off + +[Element Internal Mic] +switch = off +volume = off + +[Element Line] +switch = off +volume = off + +[Element Aux] +switch = off +volume = off + +[Element Video] +required = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +switch = off +volume = off + +.include analog-input.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input.conf b/src/modules/alsa/mixer/paths/analog-input.conf new file mode 100644 index 0000000..c9db677 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input.conf @@ -0,0 +1,102 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; A fallback for devices that lack separate Mic/Line/Aux/Video/TV +; Tuner/FM elements +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 100 + +[Element Capture] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Mic] +required-absent = any + +[Element Mic Boost] +required-absent = any + +[Element Dock Mic] +required-absent = any + +[Element Dock Mic Boost] +required-absent = any + +[Element Front Mic] +required-absent = any + +[Element Front Mic Boost] +required-absent = any + +[Element Int Mic] +required-absent = any + +[Element Int Mic Boost] +required-absent = any + +[Element Internal Mic] +required-absent = any + +[Element Internal Mic Boost] +required-absent = any + +[Element Rear Mic] +required-absent = any + +[Element Rear Mic Boost] +required-absent = any + +[Element Headset] +required-absent = any + +[Element Headset Mic] +required-absent = any + +[Element Headset Mic Boost] +required-absent = any + +[Element Headphone Mic] +required-absent = any + +[Element Headphone Mic Boost] +required-absent = any + +[Element Line] +required-absent = any + +[Element Line Boost] +required-absent = any + +[Element Aux] +required-absent = any + +[Element Video] +required-absent = any + +[Element Mic/Line] +required-absent = any + +[Element TV Tuner] +required-absent = any + +[Element FM] +required-absent = any + +.include analog-input.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input.conf.common b/src/modules/alsa/mixer/paths/analog-input.conf.common new file mode 100644 index 0000000..201087e --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-input.conf.common @@ -0,0 +1,289 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Mixer path for PulseAudio's ALSA backend, common elements for all +; input paths. If multiple options by the same id are discovered they +; will be suffixed with a number to distinguish them, in the same +; order they appear here. +; +; Source selection should use the following names: +; +; input -- If we don't know the exact kind of input +; input-microphone +; input-microphone-internal +; input-microphone-external +; input-linein +; input-video +; input-radio +; input-docking-microphone +; input-docking-linein +; input-docking +; +; We explicitly don't want to wrap the following sources: +; +; CD +; Synth/MIDI +; Phone +; Mix +; Digital/SPDIF +; Master +; PC Speaker +; +; See analog-output.conf.common for an explanation on the directives + +;;; 'Input Source Select' + +[Element Input Source Select] +enumeration = select + +[Option Input Source Select:Input1] +name = input +priority = 10 + +[Option Input Source Select:Input2] +name = input +priority = 5 + +;;; 'Input Source' + +[Element Input Source] +enumeration = select + +[Option Input Source:Digital Mic] +name = input-microphone +priority = 20 + +[Option Input Source:Microphone] +name = input-microphone +priority = 20 + +[Option Input Source:Front Microphone] +name = input-microphone +priority = 19 + +[Option Input Source:Internal Mic 1] +name = input-microphone +priority = 19 + +[Option Input Source:Line-In] +name = input-linein +priority = 18 + +[Option Input Source:Line In] +name = input-linein +priority = 18 + +[Option Input Source:Docking-Station] +name = input-docking +priority = 17 + +[Option Input Source:AUX IN] +name = input +priority = 10 + +;;; 'Capture Source' + +[Element Capture Source] +enumeration = select + +[Option Capture Source:TV Tuner] +name = input-video + +[Option Capture Source:FM] +name = input-radio + +[Option Capture Source:Mic/Line] +name = input + +[Option Capture Source:Line/Mic] +name = input + +[Option Capture Source:Microphone] +name = input-microphone + +[Option Capture Source:Int DMic] +name = input-microphone-internal + +[Option Capture Source:iMic] +name = input-microphone-internal + +[Option Capture Source:i-Mic] +name = input-microphone-internal + +[Option Capture Source:Internal Microphone] +name = input-microphone-internal + +[Option Capture Source:Front Microphone] +name = input-microphone + +[Option Capture Source:Mic1] +name = input-microphone + +[Option Capture Source:Mic2] +name = input-microphone + +[Option Capture Source:D-Mic] +name = input-microphone + +[Option Capture Source:IntMic] +name = input-microphone-internal + +[Option Capture Source:ExtMic] +name = input-microphone-external + +[Option Capture Source:Ext Mic] +name = input-microphone-external + +[Option Capture Source:E-Mic] +name = input-microphone-external + +[Option Capture Source:e-Mic] +name = input-microphone-external + +[Option Capture Source:LineIn] +name = input-linein + +[Option Capture Source:Analog] +name = input + +[Option Capture Source:Line-In] +name = input-linein + +[Option Capture Source:Line In] +name = input-linein + +[Option Capture Source:Video] +name = input-video + +[Option Capture Source:Aux] +name = input + +[Option Capture Source:Aux0] +name = input + +[Option Capture Source:Aux1] +name = input + +[Option Capture Source:Aux2] +name = input + +[Option Capture Source:Aux3] +name = input + +[Option Capture Source:AUX IN] +name = input + +[Option Capture Source:Aux In] +name = input + +[Option Capture Source:AOUT] +name = input + +[Option Capture Source:AUX] +name = input + +[Option Capture Source:Cam Mic] +name = input-microphone + +[Option Capture Source:Digital Mic] +name = input-microphone + +[Option Capture Source:Digital Mic 1] +name = input-microphone + +[Option Capture Source:Digital Mic 2] +name = input-microphone + +[Option Capture Source:Analog Inputs] +name = input + +[Option Capture Source:Unknown1] +name = input + +[Option Capture Source:Unknown2] +name = input + +[Option Capture Source:Docking-Station] +name = input-docking + +;;; 'Mic Jack Mode' + +[Element Mic Jack Mode] +enumeration = select + +[Option Mic Jack Mode:Mic In] +name = input-microphone + +[Option Mic Jack Mode:Line In] +name = input-linein + +;;; 'Digital Input Source' + +[Element Digital Input Source] +enumeration = select + +[Option Digital Input Source:Digital Mic 1] +name = input-microphone + +[Option Digital Input Source:Analog Inputs] +name = input + +[Option Digital Input Source:Digital Mic 2] +name = input-microphone + +;;; 'Analog Source' + +[Element Analog Source] +enumeration = select + +[Option Analog Source:Mic] +name = input-microphone + +[Option Analog Source:Line in] +name = input-linein + +[Option Analog Source:Aux] +name = input + +;;; 'Shared Mic/Line in' + +[Element Shared Mic/Line in] +enumeration = select + +[Option Shared Mic/Line in:Mic in] +name = input-microphone + +[Option Shared Mic/Line in:Line in] +name = input-linein + +;;; Various Boosts + +[Element Capture Boost] +switch = select + +[Option Capture Boost:on] +name = input-boost-on + +[Option Capture Boost:off] +name = input-boost-off + +[Element Auto Gain Control] +switch = select + +[Option Auto Gain Control:on] +name = input-agc-on + +[Option Auto Gain Control:off] +name = input-agc-off diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf new file mode 100644 index 0000000..1789990 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf @@ -0,0 +1,116 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Path for the second headphone output on dual-headphone machines. +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 98 + +[Properties] +device.icon_name = audio-headphones + +; HP EliteDesk 800 SFF Headphone +[Jack Front Headphone,1] +required-any = any + +; HP EliteDesk 800 DM Headphone +[Jack Front Headphone Surround] +required-any = any + +[Element Hardware Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master Mono] +switch = off +volume = off + +; This profile path is intended to control the second headphones, not +; the first headphones. But it should not hurt if we leave the +; headphone jack enabled nonetheless. +[Element Headphone] +switch = mute +volume = zero + +[Element Headphone,1] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Headphone+LO] +switch = mute +volume = zero + +[Element Speaker+LO] +switch = off +volume = off + +[Element Headphone2] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Speaker] +switch = off +volume = off + +[Element Desktop Speaker] +switch = off +volume = off + +; On some machines Front is actually a part of the Headphone path +[Element Front] +switch = mute +volume = zero + +[Element Rear] +switch = off +volume = off + +[Element Surround] +switch = off +volume = off + +[Element Side] +switch = off +volume = off + +[Element Center] +switch = off +volume = off + +[Element LFE] +switch = off +volume = off + +[Element Bass Speaker] +switch = off +volume = off + +.include analog-output.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones.conf b/src/modules/alsa/mixer/paths/analog-output-headphones.conf new file mode 100644 index 0000000..88907f0 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output-headphones.conf @@ -0,0 +1,174 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Path for mixers that have a 'Headphone' control +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 99 +description-key = analog-output-headphones + +[Properties] +device.icon_name = audio-headphones + +[Jack Dock Headphone] +required-any = any + +[Jack Dock Headphone Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Jack Front Headphone] +required-any = any + +; HP EliteDesk 800 DM Headset +[Jack Front Headphone Front] +required-any = any + +[Jack Front Headphone Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Jack Headphone] +required-any = any + +[Jack Headphone Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +# This jack can be either a headphone *or* a mic. Used on some ASUS netbooks. +[Jack Headphone Mic] +required-any = any + +[Jack Headphone - Output] +required-any = any + +[Element Hardware Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master Mono] +switch = off +volume = off + +[Element Speaker+LO] +switch = off +volume = off + +[Element Headphone+LO] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Headphone] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +; This path is intended to control the first headphones, not +; the second headphones. But it should not hurt if we leave the second +; headphone jack enabled nonetheless. +[Element Headphone,1] +switch = mute +volume = zero + +[Element Headset] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Line HP Swap] +switch = on +required-any = any + +; This profile path is intended to control the first headphones, not +; the second headphones. But it should not hurt if we leave the second +; headphone jack enabled nonetheless. +[Element Headphone2] +switch = mute +volume = zero + +[Element Speaker] +switch = off +volume = off + +[Element Desktop Speaker] +switch = off +volume = off + +; On some machines Front is actually a part of the Headphone path +[Element Front] +switch = mute +volume = zero + +[Element Rear] +switch = off +volume = off + +[Element Surround] +switch = off +volume = off + +[Element Side] +switch = off +volume = off + +[Element Center] +switch = off +volume = off + +[Element LFE] +switch = off +volume = off + +[Element Bass Speaker] +switch = off +volume = off + +[Element Speaker Front] +switch = off +volume = off + +[Element Speaker Surround] +switch = off +volume = off + +[Element Speaker Side] +switch = off +volume = off + +[Element Speaker CLFE] +switch = off +volume = off + +.include analog-output.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-output-lineout.conf b/src/modules/alsa/mixer/paths/analog-output-lineout.conf new file mode 100644 index 0000000..2dde159 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output-lineout.conf @@ -0,0 +1,208 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +[General] +priority = 90 +description-key = analog-output-lineout + +[Jack Line Out] +required-any = any + +[Jack Line Out Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Front Line Out] +required-any = any + +[Jack Front Line Out Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Rear Line Out] +required-any = any + +[Jack Rear Line Out Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Line Out Front] +required-any = any + +[Jack Line Out Front Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Line Out CLFE] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Line Out CLFE Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Line Out Surround] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Line Out Surround Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Line Out Side] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Line Out Side Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Jack Dock Line Out] +required-any = any + +[Jack Dock Line Out Phantom] +state.plugged = unknown +state.unplugged = unknown +required-any = any + +[Element Hardware Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Speaker+LO] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right +required-any = any + +[Element Headphone+LO] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right +required-any = any + +[Element Master Mono] +switch = off +volume = off + +[Element Line HP Swap] +switch = off +required-any = any + +; This profile path is intended to control line out, let's mute headphones +; else there will be a spike when plugging in headphones +[Element Headphone] +switch = off +volume = off + +[Element Headphone,1] +switch = off +volume = off + +[Element Headphone2] +switch = off +volume = off + +[Element Speaker] +switch = off +volume = off + +[Element Desktop Speaker] +switch = off +volume = off + +[Element Front] +switch = mute +volume = merge +override-map.1 = all-front +override-map.2 = front-left,front-right + +[Element Rear] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Surround] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Side] +switch = mute +volume = merge +override-map.1 = all-side +override-map.2 = side-left,side-right + +[Element Center] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,all-center + +[Element LFE] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe + +[Element CLFE] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,lfe + +[Element Bass Speaker] +switch = off +volume = off + +[Element Speaker Front] +switch = off +volume = off + +[Element Speaker Surround] +switch = off +volume = off + +[Element Speaker Side] +switch = off +volume = off + +[Element Speaker CLFE] +switch = off +volume = off + +.include analog-output.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-output-mono.conf b/src/modules/alsa/mixer/paths/analog-output-mono.conf new file mode 100644 index 0000000..5e49405 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output-mono.conf @@ -0,0 +1,99 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Intended for usage on boards that have a separate Mono output plug. +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 50 + +[Element Hardware Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master] +switch = off +volume = off + +[Element Master Mono] +required = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +; This profile path is intended to control the speaker, not the +; headphones. But it should not hurt if we leave the headphone jack +; enabled nonetheless. +[Element Headphone] +switch = mute +volume = zero + +[Element Headphone,1] +switch = mute +volume = zero + +[Element Headphone+LO] +switch = mute +volume = zero + +[Element Headphone2] +switch = mute +volume = zero + +[Element Speaker] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Speaker+LO] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Desktop Speaker] +switch = off +volume = off + +[Element Front] +switch = off +volume = off + +[Element Rear] +switch = off +volume = off + +[Element Surround] +switch = off +volume = off + +[Element Side] +switch = off +volume = off + +[Element Center] +switch = off +volume = off + +[Element LFE] +switch = off +volume = off + +.include analog-output.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf b/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf new file mode 100644 index 0000000..4ee72f5 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf @@ -0,0 +1,181 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Path for mixers that don't have a 'Speaker' control, but where we +; force enable the speaker paths nonetheless. +; Needed for some older Dell laptops. +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 100 +description-key = analog-output-speaker + +[Properties] +device.icon_name = audio-speakers + +[Jack Headphone] +state.plugged = no +state.unplugged = unknown + +[Jack Front Headphone] +state.plugged = no +state.unplugged = unknown + +[Jack Line Out] +state.plugged = no +state.unplugged = unknown + +[Jack Line Out Front] +state.plugged = no +state.unplugged = unknown + +[Jack Front Line Out] +state.plugged = no +state.unplugged = unknown + +[Jack Rear Line Out] +state.plugged = no +state.unplugged = unknown + +[Jack Dock Line Out] +state.plugged = no +state.unplugged = unknown + +[Element Hardware Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master Mono] +switch = off +volume = off + +; This profile path is intended to control the speaker, not the +; headphones. But it should not hurt if we leave the headphone jack +; enabled nonetheless. +[Element Headphone] +switch = mute +volume = zero + +[Element Headphone,1] +switch = mute +volume = zero + +[Element Headphone2] +switch = mute +volume = zero + +[Element Headphone+LO] +switch = off +volume = off + +[Element Speaker+LO] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Speaker] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Desktop Speaker] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Front] +switch = mute +volume = merge +override-map.1 = all-front +override-map.2 = front-left,front-right + +[Element Front Speaker] +switch = mute +volume = merge +override-map.1 = all-front +override-map.2 = front-left,front-right + +[Element Rear] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Surround] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Surround Speaker] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Side] +switch = mute +volume = merge +override-map.1 = all-side +override-map.2 = side-left,side-right + +[Element Center] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,all-center + +[Element Center Speaker] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,all-center + +[Element LFE] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe + +[Element LFE Speaker] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe + +[Element Bass Speaker] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe + +[Element CLFE] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,lfe + +.include analog-output.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-speaker.conf new file mode 100644 index 0000000..fcf2f5c --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output-speaker.conf @@ -0,0 +1,233 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Path for mixers that have a 'Speaker' control +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 100 +description-key = analog-output-speaker + +[Properties] +device.icon_name = audio-speakers + +[Jack Headphone] +state.plugged = no +state.unplugged = unknown + +[Jack Dock Headphone] +state.plugged = no +state.unplugged = unknown + +[Jack Front Headphone] +state.plugged = no +state.unplugged = unknown + +[Jack Line Out] +state.plugged = no +state.unplugged = unknown + +[Jack Line Out Front] +state.plugged = no +state.unplugged = unknown + +[Jack Front Line Out] +state.plugged = no +state.unplugged = unknown + +[Jack Rear Line Out] +state.plugged = no +state.unplugged = unknown + +[Jack Dock Line Out] +state.plugged = no +state.unplugged = unknown + +[Jack Speaker] +required-any = any + +[Jack Speaker Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Jack Speaker Front Phantom] +required-any = any +state.plugged = unknown +state.unplugged = unknown + +[Jack Speaker - Output] +required-any = any + +[Element Hardware Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master Mono] +switch = off +volume = off + +; This profile path is intended to control the speaker, let's mute headphones +; else there will be a spike when plugging in headphones +[Element Headphone] +switch = off +volume = off + +[Element Headphone,1] +switch = off +volume = off + +[Element Headphone2] +switch = off +volume = off + +[Element Headphone+LO] +switch = off +volume = off + +[Element Speaker+LO] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Speaker] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Desktop Speaker] +required-any = any +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Front] +switch = mute +volume = merge +override-map.1 = all-front +override-map.2 = front-left,front-right + +[Element Front Speaker] +switch = mute +volume = merge +override-map.1 = all-front +override-map.2 = front-left,front-right +required-any = any + +[Element Speaker Front] +switch = mute +volume = merge +override-map.1 = all-front +override-map.2 = front-left,front-right +required-any = any + +[Element Rear] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Surround] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Surround Speaker] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right +required-any = any + +[Element Speaker Surround] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right +required-any = any + +[Element Side] +switch = mute +volume = merge +override-map.1 = all-side +override-map.2 = side-left,side-right + +[Element Speaker Side] +switch = mute +volume = merge +override-map.1 = all-side +override-map.2 = side-left,side-right + +[Element Center] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,all-center + +[Element Center Speaker] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,all-center +required-any = any + +[Element LFE] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe + +[Element LFE Speaker] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe +required-any = any + +[Element Bass Speaker] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe +required-any = any + +[Element CLFE] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,lfe + +[Element Speaker CLFE] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,lfe + +.include analog-output.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-output.conf b/src/modules/alsa/mixer/paths/analog-output.conf new file mode 100644 index 0000000..e6ba983 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output.conf @@ -0,0 +1,82 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Intended for the 'default' output. Note that a-o-speaker.conf has a +; higher priority than this +; +; See analog-output.conf.common for an explanation on the directives + +[General] +priority = 99 + +[Element Hardware Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element Master Mono] +switch = off +volume = off + +[Element Front] +switch = mute +volume = merge +override-map.1 = all-front +override-map.2 = front-left,front-right + +[Element Rear] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Surround] +switch = mute +volume = merge +override-map.1 = all-rear +override-map.2 = rear-left,rear-right + +[Element Side] +switch = mute +volume = merge +override-map.1 = all-side +override-map.2 = side-left,side-right + +[Element Center] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,all-center + +[Element LFE] +switch = mute +volume = merge +override-map.1 = lfe +override-map.2 = lfe,lfe + +[Element CLFE] +switch = mute +volume = merge +override-map.1 = all-center +override-map.2 = all-center,lfe + +.include analog-output.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-output.conf.common b/src/modules/alsa/mixer/paths/analog-output.conf.common new file mode 100644 index 0000000..31d4b44 --- /dev/null +++ b/src/modules/alsa/mixer/paths/analog-output.conf.common @@ -0,0 +1,186 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Common part of all paths + +; So here's generally how mixer paths are used by PA: PA goes through +; a mixer path file from top to bottom and checks if a mixer element +; described therein exists. If so it is added to the list of mixer +; elements PA will control, keeping the order it read them in. If a +; mixer element described here has set the required= or +; required-absent= directives a path might not be accepted as valid +; and is ignored in its entirety (see below). However usually if a +; element listed here is missing this one element is ignored but not +; the entire path. +; +; When a device shall be muted/unmuted *all* elements listed in a path +; file with "switch = mute" will be toggled. +; +; When a device shall change its volume, PA will got through the list +; of all elements with "volume = merge" and set the volume on the +; first element. If that element does not support dB volumes, this is +; where the story ends. If it does support dB volumes, PA divides the +; requested volume by the volume that was set on this element, and +; then go on to the next element with "volume = merge" and then set +; that there, and so on. That way the first volume element in the +; path will be the one that does the 'biggest' part of the overall +; volume adjustment, with the remaining elements usually being set to +; some value next to 0dB. This logic makes sure we get the full range +; over all volume sliders and a very high granularity of volumes +; already in hardware. +; +; All switches and enumerations set to "select" are exposed via the +; "port" functionality of sinks/sources. Basically every possible +; switch setting and every possible enumeration setting will be +; combined and made into a "port". So make sure you don't list too +; many switches/enums for exposing, because the number of ports might +; rise exponentially. +; +; Only one path can be selected at a time. All paths that are valid +; for an audio device will be exposed as "port" for the sink/source. + + +; [General] +; type = ... # The device type. It's highly recommended to set a type for every path. +; # See parse_type() in alsa-mixer.c for supported values. +; priority = ... # Priority for this path +; description-key = ... # The path description is looked up from a table in path_verify() in +; # src/modules/alsa/alsa-mixer.c. By default the path name (i.e. the file name +; # minus the ".conf" suffix) is used as the lookup key, but if this option is +; # set, then the given string is used as the key instead. In any case the +; # "description" option can be used to override the path description. +; description = ... # Description for this path. Overrides the normal description lookup logic, as +; # described in the "description-key" documentation above. +; mute-during-activation = yes | no # If this path supports hardware mute, should the hw mute be used while activating this +; # path? In some cases this can reduce extra noises during port switching, while in other +; # cases this can increase such noises. Default: no. +; eld-device = ... # If this is an HDMI port, set to "auto" so that PulseAudio will try to read +; # the monitor ELD information from the ALSA mixer. By default the ELD information +; # is not read, because it's only applicable with HDMI. Earlier the "auto" option +; # didn't exist, and the hw device index had to be manually configured. For +; # backwards compatibility, it's still possible to manually configure the device +; # index using this option. +; +; [Properties] # Property list for this path. The list is merged into the port property list. +; <key> = <value> # Each property is defined on its own line. +; ... +; +; [Option ...:...] # For each option of an enumeration or switch element +; # that shall be exposed as a sink/source port. Needs to +; # be named after the Element, followed by a colon, followed +; # by the option name, resp. on/off if the element is a switch. +; name = ... # Logical name to use in the path identifier +; priority = ... # Priority if this is made into a device port +; required = ignore | enumeration | any # In this element, this option must exist or the path will be invalid. ("any" is an alias for "enumeration".) +; required-any = ignore | enumeration | any # In this element, either this or another option must exist (or an element) +; required-absent = ignore | enumeration | any # In this element, this option must not exist or the path will be invalid +; +; [Element ...] # For each element that we shall control. The "..." here is the element name, +; # or name and index separated by a comma. +; required = ignore | switch | volume | enumeration | any # If set, require this element to be of this kind and available, +; # otherwise don't consider this path valid for the card +; required-any = ignore | switch | volume | enumeration | any # If set, at least one of the elements or jacks with required-any in this +; # path must be present, otherwise this path is invalid for the card +; required-absent = ignore | switch | volume # If set, require this element to not be of this kind and not +; # available, otherwise don't consider this path valid for the card +; +; switch = ignore | mute | off | on | select # What to do with this switch: ignore it, make it follow mute status, +; # always set it to off, always to on, or make it selectable as port. +; # If set to 'select' you need to define an Option section for on +; # and off +; volume = ignore | merge | off | zero | <volume step> # What to do with this volume: ignore it, merge it into the device +; # volume slider, always set it to the lowest value possible, or always +; # set it to 0 dB (for whatever that means), or always set it to +; # <volume step> (this only makes sense in path configurations where +; # the exact hardware and driver are known beforehand). +; volume-limit = <volume step> # Limit the maximum volume by disabling the volume steps above <volume step>. +; enumeration = ignore | select # What to do with this enumeration, ignore it or make it selectable +; # via device ports. If set to 'select' you need to define an Option section +; # for each of the items you want to expose +; direction = playback | capture # Is this relevant only for playback or capture? If not set this will implicitly be +; # set the direction of the PCM device is opened as. Generally this doesn't need to be set +; # unless you have a broken driver that has playback controls marked for capture or vice +; # versa +; direction-try-other = no | yes # If the element does not supported what is requested, try the other direction, too? +; +; override-map.1 = ... # Override the channel mask of the mixer control if the control only exposes a single channel +; override-map.2 = ... # Override the channel masks of the mixer control if the control only exposes two channels +; # Override maps should list for each element channel which high-level channels it controls via a +; # channel mask. A channel mask may either be the name of a single channel, or the words "all-left", +; # "all-right", "all-center", "all-front", "all-rear", and "all" to encode a specific subset of +; # channels in a mask +; [Jack ...] # For each jack that we will use for jack detection +; # The name 'Jack Foo' must match ALSA's 'Foo Jack' control. +; required = ignore | any # If not set to ignore, make the path invalid if this jack control is not present. +; required-absent = ignore | any # If not set to ignore, make the path invalid if this jack control is present. +; required-any = ignore | any # If not set to ignore, make the path invalid if no jack controls and no elements with +; # the required-any are present. +; state.plugged = yes | no | unknown # Normally a plugged jack would mean the port becomes available, and an unplugged means it's +; state.unplugged = yes | no | unknown # unavailable, but the port status can be overridden by specifying state.plugged and/or state.unplugged. +; append-pcm-to-name = no | yes # Add ",pcm=N" to the jack name? N is the hw PCM device index. HDMI jacks have +; # the PCM device index in their name, but different drivers use different +; # numbering schemes, so we can't hardcode the full jack name in our configuration +; # files. + +[Element PCM] +switch = mute +volume = merge +override-map.1 = all +override-map.2 = all-left,all-right + +[Element External Amplifier] +switch = select + +[Option External Amplifier:on] +name = output-amplifier-on +priority = 10 + +[Option External Amplifier:off] +name = output-amplifier-off +priority = 0 + +[Element Bass Boost] +switch = select + +[Option Bass Boost:on] +name = output-bass-boost-on +priority = 0 + +[Option Bass Boost:off] +name = output-bass-boost-off +priority = 10 + +[Element IEC958] +switch = off + +[Element IEC958 Optical Raw] +switch = off + +;;; 'Analog Output' + +[Element Analog Output] +enumeration = select + +[Option Analog Output:Speakers] +name = output-speaker +priority = 10 + +[Option Analog Output:Headphones] +name = output-headphones +priority = 9 + +[Option Analog Output:FP Headphones] +name = output-headphones +priority = 8 diff --git a/src/modules/alsa/mixer/paths/hdmi-output-0.conf b/src/modules/alsa/mixer/paths/hdmi-output-0.conf new file mode 100644 index 0000000..bb3cec1 --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-0.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort +type = hdmi +priority = 59 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/hdmi-output-1.conf b/src/modules/alsa/mixer/paths/hdmi-output-1.conf new file mode 100644 index 0000000..3389a72 --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-1.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort 2 +type = hdmi +priority = 58 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/hdmi-output-2.conf b/src/modules/alsa/mixer/paths/hdmi-output-2.conf new file mode 100644 index 0000000..316d810 --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-2.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort 3 +type = hdmi +priority = 57 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/hdmi-output-3.conf b/src/modules/alsa/mixer/paths/hdmi-output-3.conf new file mode 100644 index 0000000..0601ef7 --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-3.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort 4 +type = hdmi +priority = 56 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/hdmi-output-4.conf b/src/modules/alsa/mixer/paths/hdmi-output-4.conf new file mode 100644 index 0000000..ded155b --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-4.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort 5 +type = hdmi +priority = 55 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/hdmi-output-5.conf b/src/modules/alsa/mixer/paths/hdmi-output-5.conf new file mode 100644 index 0000000..de31791 --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-5.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort 6 +type = hdmi +priority = 54 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/hdmi-output-6.conf b/src/modules/alsa/mixer/paths/hdmi-output-6.conf new file mode 100644 index 0000000..6d72176 --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-6.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort 7 +type = hdmi +priority = 53 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/hdmi-output-7.conf b/src/modules/alsa/mixer/paths/hdmi-output-7.conf new file mode 100644 index 0000000..d5d0771 --- /dev/null +++ b/src/modules/alsa/mixer/paths/hdmi-output-7.conf @@ -0,0 +1,12 @@ +[General] +description = HDMI / DisplayPort 8 +type = hdmi +priority = 52 +eld-device = auto + +[Properties] +device.icon_name = video-display + +[Jack HDMI/DP] +append-pcm-to-name = yes +required = ignore diff --git a/src/modules/alsa/mixer/paths/iec958-stereo-input.conf b/src/modules/alsa/mixer/paths/iec958-stereo-input.conf new file mode 100644 index 0000000..babc839 --- /dev/null +++ b/src/modules/alsa/mixer/paths/iec958-stereo-input.conf @@ -0,0 +1,20 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +[Element PCM Capture Source] +enumeration = select + +[Option PCM Capture Source:IEC958 In] +name = iec958-input diff --git a/src/modules/alsa/mixer/paths/iec958-stereo-output.conf b/src/modules/alsa/mixer/paths/iec958-stereo-output.conf new file mode 100644 index 0000000..d47e5eb --- /dev/null +++ b/src/modules/alsa/mixer/paths/iec958-stereo-output.conf @@ -0,0 +1,18 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + + +[Element IEC958] +switch = mute diff --git a/src/modules/alsa/mixer/paths/steelseries-arctis-output-chat-common.conf b/src/modules/alsa/mixer/paths/steelseries-arctis-output-chat-common.conf new file mode 100644 index 0000000..5842bfe --- /dev/null +++ b/src/modules/alsa/mixer/paths/steelseries-arctis-output-chat-common.conf @@ -0,0 +1,27 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Steelseries Arctis 5 USB headset stereo chat path. The headset has two +; output devices. The first one is meant for voice audio, and the second +; one meant for everything else. The purpose of this unusual design is to +; provide separate volume controls for voice and other audio, which can be +; useful in gaming. + +[General] +priority = 50 + +[Element Com Speaker] +switch = mute +volume = merge diff --git a/src/modules/alsa/mixer/paths/steelseries-arctis-output-game-common.conf b/src/modules/alsa/mixer/paths/steelseries-arctis-output-game-common.conf new file mode 100644 index 0000000..b758a6f --- /dev/null +++ b/src/modules/alsa/mixer/paths/steelseries-arctis-output-game-common.conf @@ -0,0 +1,27 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Steelseries Arctis 5 USB headset stereo game path. The headset has two +; output devices. The first one is meant for voice audio, and the second +; one meant for everything else. The purpose of this unusual design is to +; provide separate volume controls for voice and other audio, which can be +; useful in gaming. + +[General] +priority = 99 + +[Element PCM] +switch = mute +volume = merge diff --git a/src/modules/alsa/mixer/paths/usb-gaming-headset-input.conf b/src/modules/alsa/mixer/paths/usb-gaming-headset-input.conf new file mode 100644 index 0000000..9fa7fe9 --- /dev/null +++ b/src/modules/alsa/mixer/paths/usb-gaming-headset-input.conf @@ -0,0 +1,34 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; USB gaming headset microphone input path. These headsets usually have two +; output devices. The first one is mono, meant for voice audio, and the second +; one is stereo, meant for everything else. The purpose of this unusual design +; is to provide separate volume controls for voice and other audio, which can +; be useful in gaming. +; +; Works with: +; Steelseries Arctis 7 +; Steelseries Arctis Pro Wireless. +; Lucidsound LS31 + +[General] +description-key = analog-input-microphone-headset + +[Element Headset] +volume = merge +switch = mute +override-map.1 = all +override-map.2 = all-left,all-right diff --git a/src/modules/alsa/mixer/paths/usb-gaming-headset-output-mono.conf b/src/modules/alsa/mixer/paths/usb-gaming-headset-output-mono.conf new file mode 100644 index 0000000..6df662f --- /dev/null +++ b/src/modules/alsa/mixer/paths/usb-gaming-headset-output-mono.conf @@ -0,0 +1,34 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; USB gaming headset mono output path. These headsets usually have two +; output devices. The first one is mono, meant for voice audio, and the second +; one is stereo, meant for everything else. The purpose of this unusual design +; is to provide separate volume controls for voice and other audio, which can +; be useful in gaming. +; +; Works with: +; Steelseries Arctis 7 +; Steelseries Arctis Pro Wireless. +; Lucidsound LS31 + +[General] +description-key = analog-output-headphones-mono + +[Element PCM] +volume = merge +switch = mute +override-map.1 = all +override-map.2 = all-left,all-right diff --git a/src/modules/alsa/mixer/paths/usb-gaming-headset-output-stereo.conf b/src/modules/alsa/mixer/paths/usb-gaming-headset-output-stereo.conf new file mode 100644 index 0000000..e3f91cd --- /dev/null +++ b/src/modules/alsa/mixer/paths/usb-gaming-headset-output-stereo.conf @@ -0,0 +1,32 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; USB gaming headset mono output path. These headsets usually have two +; output devices. The first one is mono, meant for voice audio, and the second +; one is stereo, meant for everything else. The purpose of this unusual design +; is to provide separate volume controls for voice and other audio, which can +; be useful in gaming. +; +; Works with: +; Steelseries Arctis 7 +; Steelseries Arctis Pro Wireless. +; Lucidsound LS31 +; +; This path doesn't provide hardware volume control, because the stereo +; output is controlled by the PCM element with index 1, and currently +; PulseAudio only supports elements with index 0. + +[General] +description-key = analog-output-headphones diff --git a/src/modules/alsa/mixer/profile-sets/audigy.conf b/src/modules/alsa/mixer/profile-sets/audigy.conf new file mode 100644 index 0000000..043596e --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/audigy.conf @@ -0,0 +1,94 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Creative Sound Blaster Audigy product line +; +; These are just copies of the mappings we find in default.conf, with the +; small change of making analog-stereo and analog-mono non-fallback mappings. +; This is needed because these cards only support duplex profiles with mono +; inputs, and in the default configuration, with stereo being a fallback +; mapping, the mono mapping is never tried. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = yes + +# Based on stereo-fallback +[Mapping analog-stereo] +device-strings = hw:%f +channel-map = front-left,front-right +paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headphone-mic analog-input-headset-mic +priority = 1 + +# Based on mono-fallback +[Mapping analog-mono] +device-strings = hw:%f +channel-map = mono +paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headset-mic +priority = 1 + +# The rest of these are identical to what's in default.conf +[Mapping analog-surround-21] +device-strings = surround21:%f +channel-map = front-left,front-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 13 +direction = output + +[Mapping analog-surround-40] +device-strings = surround40:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 12 +direction = output + +[Mapping analog-surround-41] +device-strings = surround41:%f +channel-map = front-left,front-right,rear-left,rear-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 13 +direction = output + +[Mapping analog-surround-50] +device-strings = surround50:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 12 +direction = output + +[Mapping analog-surround-51] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 13 +direction = output + +[Mapping analog-surround-71] +device-strings = surround71:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +description = Analog Surround 7.1 +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 12 +direction = output + +[Mapping iec958-stereo] +device-strings = iec958:%f +channel-map = left,right +paths-input = iec958-stereo-input +paths-output = iec958-stereo-output +priority = 5 diff --git a/src/modules/alsa/mixer/profile-sets/cmedia-high-speed-true-hdaudio.conf b/src/modules/alsa/mixer/profile-sets/cmedia-high-speed-true-hdaudio.conf new file mode 100644 index 0000000..1b6f61c --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/cmedia-high-speed-true-hdaudio.conf @@ -0,0 +1,66 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +# Config for CMEDIA USB2.0 High-Speed True HD Audio 147a:e055 +# Added by Jean-Philippe Guillemin <h1p8r10n@gmail.com> + + +[General] +auto-profiles = yes + +[Mapping analog-stereo] +device-strings = front:%f +channel-map = left,right +paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headphone-mic analog-input-headset-mic +priority = 10 + +# If everything else fails, try to use hw:0 as a stereo device. +[Mapping stereo-fallback] +device-strings = hw:%f +fallback = yes +channel-map = front-left,front-right +paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headphone-mic analog-input-headset-mic +priority = 1 + +[Mapping analog-surround-21] +device-strings = surround21:%f +channel-map = front-left,front-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 8 +direction = output + +[Mapping analog-surround-51] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 8 +direction = output + +[Mapping analog-surround-71] +device-strings = surround71:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +description = Analog Surround 7.1 +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 7 +direction = output + +[Mapping iec958-stereo] +device-strings = hw:%f,2 hw:%f,0 +channel-map = left,right +paths-output = iec958-stereo-output +paths-input = iec958-stereo-input +priority = 5 diff --git a/src/modules/alsa/mixer/profile-sets/default.conf b/src/modules/alsa/mixer/profile-sets/default.conf new file mode 100644 index 0000000..9b691fe --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/default.conf @@ -0,0 +1,484 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Default profile definitions for the ALSA backend of PulseAudio. This +; is used as fallback for all cards that have no special mapping +; assigned (and should be good enough for the vast majority of +; cards). If you want to assign a different profile set than this one +; to a device, either set the udev property PULSE_PROFILE_SET for the +; card, or use the "profile_set" module argument when loading +; module-alsa-card. +; +; So what is this about? Simply, what we do here is map ALSA devices +; to how they are exposed in PA. We say which ALSA device string to +; use to open a device, which channel mapping to use then, and which +; mixer path to use. This is encoded in a 'mapping'. Multiple of these +; mappings can be bound together in a 'profile' which is then directly +; exposed in the UI as a card profile. Each mapping assigned to a +; profile will result in one sink/source to be created if the profile +; is selected for the card. +; +; Additionally, the path set configuration files can describe the +; decibel values assigned to the steps of the volume elements. This +; can be used to work around situations when the alsa driver doesn't +; provide any decibel information, or when the information is +; incorrect. + + +; [General] +; auto-profiles = no | yes # Instead of defining all profiles manually, autogenerate +; # them by combining every input mapping with every output mapping. +; +; [Mapping id] +; device-strings = ... # ALSA device string. %f will be replaced by the card identifier. +; channel-map = ... # Channel mapping to use for this device +; description = ... # Description for the mapping. Note that it's better to set the description +; # in the well_known_descriptions table in alsa-mixer.c than with this +; # option, because the descriptions in alsa-mixer.c are translatable. +; description-key = ... # A custom key for the well_known_descriptions table (by default the mapping +; # name is used). +; paths-input = ... # A list of mixer paths to use. Every path in this list will be probed. +; # If multiple are found to be working they will be available as device ports +; paths-output = ... +; element-input = ... # Instead of configuring a full mixer path simply configure a single +; # mixer element for volume/mute handling. The value can be an element +; # name, or name and index separated by a comma. +; element-output = ... +; priority = ... +; direction = any | input | output # Only useful for? +; +; exact-channels = yes | no # If no, and the exact number of channels is not supported, +; # allow device to be opened with another channel count +; fallback = no | yes # This mapping will only be considered if all non-fallback mappings fail +; intended-roles = ... # Set the device.intended_roles property for the sink/source. +; +; [Profile id] +; input-mappings = ... # Lists mappings for sources on this profile, those mapping must be +; # defined in this file too +; output-mappings = ... # Lists mappings for sinks on this profile, those mappings must be +; # defined in this file too +; description = ... +; priority = ... # Numeric value to deduce priority for this profile +; skip-probe = no | yes # Skip probing for availability? If this is yes then this profile +; # will be assumed as working without probing. Makes initialization +; # a bit faster but only works if the card is really known well. +; +; fallback = no | yes # This profile will only be considered if all non-fallback profiles fail +; [DecibelFix element] # Decibel fixes can be used to work around missing or incorrect dB +; # information from alsa. A decibel fix is a table that maps volume steps +; # to decibel values for one volume element. The "element" part in the +; # section title is the name of the volume element (or name and index +; # separated by a comma). +; # +; # NOTE: This feature is meant just as a help for figuring out the correct +; # decibel values. PulseAudio is not the correct place to maintain the +; # decibel mappings! +; # +; # If you need this feature, then you should make sure that when you have +; # the correct values figured out, the alsa driver developers get informed +; # too, so that they can fix the driver. +; +; db-values = ... # The option value consists of pairs of step numbers and decibel values. +; # The pairs are separated with whitespace, and steps are separated from +; # the corresponding decibel values with a colon. The values must be in an +; # increasing order. Here's an example of a valid string: +; # +; # "0:-40.50 1:-38.70 3:-33.00 11:0" +; # +; # The lowest step imposes a lower limit for hardware volume and the +; # highest step correspondingly imposes a higher limit. That means that +; # that the mixer will never be set outside those values - the rest of the +; # volume scale is done using software volume. +; # +; # As can be seen in the example, you don't need to specify a dB value for +; # each step. The dB values for skipped steps will be linearly interpolated +; # using the nearest steps that are given. + +[General] +auto-profiles = yes + +[Mapping analog-stereo] +device-strings = front:%f +channel-map = left,right +paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headphone-mic analog-input-headset-mic +priority = 15 + +# If everything else fails, try to use hw:0 as a stereo device... +[Mapping stereo-fallback] +device-strings = hw:%f +fallback = yes +channel-map = front-left,front-right +paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headphone-mic analog-input-headset-mic +priority = 1 + +# ...and if even that fails, try to use hw:0 as a mono device. +[Mapping mono-fallback] +device-strings = hw:%f +fallback = yes +channel-map = mono +paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headset-mic +priority = 1 + +[Mapping analog-surround-21] +device-strings = surround21:%f +channel-map = front-left,front-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 13 +direction = output + +[Mapping analog-surround-40] +device-strings = surround40:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 12 +direction = output + +[Mapping analog-surround-41] +device-strings = surround41:%f +channel-map = front-left,front-right,rear-left,rear-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 13 +direction = output + +[Mapping analog-surround-50] +device-strings = surround50:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 12 +direction = output + +[Mapping analog-surround-51] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 13 +direction = output + +[Mapping analog-surround-71] +device-strings = surround71:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +description = Analog Surround 7.1 +paths-output = analog-output analog-output-lineout analog-output-speaker +priority = 12 +direction = output + +[Mapping iec958-stereo] +device-strings = iec958:%f +channel-map = left,right +paths-input = iec958-stereo-input +paths-output = iec958-stereo-output +priority = 5 + +[Mapping iec958-ac3-surround-40] +device-strings = a52:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = iec958-stereo-output +priority = 2 +direction = output + +[Mapping iec958-ac3-surround-51] +device-strings = a52:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = iec958-stereo-output +priority = 3 +direction = output + +[Mapping iec958-dts-surround-51] +device-strings = dca:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = iec958-stereo-output +priority = 3 +direction = output + +[Mapping hdmi-stereo] +description = Digital Stereo (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = left,right +priority = 9 +direction = output + +[Mapping hdmi-surround] +description = Digital Surround 5.1 (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 8 +direction = output + +[Mapping hdmi-surround71] +description = Digital Surround 7.1 (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 8 +direction = output + +[Mapping hdmi-dts-surround] +description = Digital Surround 5.1 (HDMI/DTS) +device-strings = dcahdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-stereo-extra1] +description = Digital Stereo (HDMI 2) +device-strings = hdmi:%f,1 +paths-output = hdmi-output-1 +channel-map = left,right +priority = 7 +direction = output + +[Mapping hdmi-surround-extra1] +description = Digital Surround 5.1 (HDMI 2) +device-strings = hdmi:%f,1 +paths-output = hdmi-output-1 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-surround71-extra1] +description = Digital Surround 7.1 (HDMI 2) +device-strings = hdmi:%f,1 +paths-output = hdmi-output-1 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 6 +direction = output + +[Mapping hdmi-dts-surround-extra1] +description = Digital Surround 5.1 (HDMI 2/DTS) +device-strings = dcahdmi:%f,1 +paths-output = hdmi-output-1 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-stereo-extra2] +description = Digital Stereo (HDMI 3) +device-strings = hdmi:%f,2 +paths-output = hdmi-output-2 +channel-map = left,right +priority = 7 +direction = output + +[Mapping hdmi-surround-extra2] +description = Digital Surround 5.1 (HDMI 3) +device-strings = hdmi:%f,2 +paths-output = hdmi-output-2 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-surround71-extra2] +description = Digital Surround 7.1 (HDMI 3) +device-strings = hdmi:%f,2 +paths-output = hdmi-output-2 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 6 +direction = output + +[Mapping hdmi-dts-surround-extra2] +description = Digital Surround 5.1 (HDMI 3/DTS) +device-strings = dcahdmi:%f,2 +paths-output = hdmi-output-2 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-stereo-extra3] +description = Digital Stereo (HDMI 4) +device-strings = hdmi:%f,3 +paths-output = hdmi-output-3 +channel-map = left,right +priority = 7 +direction = output + +[Mapping hdmi-surround-extra3] +description = Digital Surround 5.1 (HDMI 4) +device-strings = hdmi:%f,3 +paths-output = hdmi-output-3 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-surround71-extra3] +description = Digital Surround 7.1 (HDMI 4) +device-strings = hdmi:%f,3 +paths-output = hdmi-output-3 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 6 +direction = output + +[Mapping hdmi-dts-surround-extra3] +description = Digital Surround 5.1 (HDMI 4/DTS) +device-strings = dcahdmi:%f,3 +paths-output = hdmi-output-3 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-stereo-extra4] +description = Digital Stereo (HDMI 5) +device-strings = hdmi:%f,4 +paths-output = hdmi-output-4 +channel-map = left,right +priority = 7 +direction = output + +[Mapping hdmi-surround-extra4] +description = Digital Surround 5.1 (HDMI 5) +device-strings = hdmi:%f,4 +paths-output = hdmi-output-4 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-surround71-extra4] +description = Digital Surround 7.1 (HDMI 5) +device-strings = hdmi:%f,4 +paths-output = hdmi-output-4 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 6 +direction = output + +[Mapping hdmi-dts-surround-extra4] +description = Digital Surround 5.1 (HDMI 5/DTS) +device-strings = dcahdmi:%f,4 +paths-output = hdmi-output-4 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-stereo-extra5] +description = Digital Stereo (HDMI 6) +device-strings = hdmi:%f,5 +paths-output = hdmi-output-5 +channel-map = left,right +priority = 7 +direction = output + +[Mapping hdmi-surround-extra5] +description = Digital Surround 5.1 (HDMI 6) +device-strings = hdmi:%f,5 +paths-output = hdmi-output-5 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-surround71-extra5] +description = Digital Surround 7.1 (HDMI 6) +device-strings = hdmi:%f,5 +paths-output = hdmi-output-5 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 6 +direction = output + +[Mapping hdmi-dts-surround-extra5] +description = Digital Surround 5.1 (HDMI 6/DTS) +device-strings = dcahdmi:%f,5 +paths-output = hdmi-output-5 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-stereo-extra6] +description = Digital Stereo (HDMI 7) +device-strings = hdmi:%f,6 +paths-output = hdmi-output-6 +channel-map = left,right +priority = 7 +direction = output + +[Mapping hdmi-surround-extra6] +description = Digital Surround 5.1 (HDMI 7) +device-strings = hdmi:%f,6 +paths-output = hdmi-output-6 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-surround71-extra6] +description = Digital Surround 7.1 (HDMI 7) +device-strings = hdmi:%f,6 +paths-output = hdmi-output-6 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 6 +direction = output + +[Mapping hdmi-dts-surround-extra6] +description = Digital Surround 5.1 (HDMI 7/DTS) +device-strings = dcahdmi:%f,6 +paths-output = hdmi-output-6 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-stereo-extra7] +description = Digital Stereo (HDMI 8) +device-strings = hdmi:%f,7 +paths-output = hdmi-output-7 +channel-map = left,right +priority = 7 +direction = output + +[Mapping hdmi-surround-extra7] +description = Digital Surround 5.1 (HDMI 8) +device-strings = hdmi:%f,7 +paths-output = hdmi-output-7 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping hdmi-surround71-extra7] +description = Digital Surround 7.1 (HDMI 8) +device-strings = hdmi:%f,7 +paths-output = hdmi-output-7 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 6 +direction = output + +[Mapping hdmi-dts-surround-extra7] +description = Digital Surround 5.1 (HDMI 8/DTS) +device-strings = dcahdmi:%f,7 +paths-output = hdmi-output-7 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 6 +direction = output + +[Mapping multichannel-output] +device-strings = hw:%f +channel-map = left,right,rear-left,rear-right +exact-channels = false +fallback = yes +priority = 1 +direction = output + +[Mapping multichannel-input] +device-strings = hw:%f +channel-map = left,right,rear-left,rear-right +exact-channels = false +fallback = yes +priority = 1 +direction = input + +; An example for defining multiple-sink profiles +#[Profile output:analog-stereo+output:iec958-stereo+input:analog-stereo] +#description = Foobar +#output-mappings = analog-stereo iec958-stereo +#input-mappings = analog-stereo diff --git a/src/modules/alsa/mixer/profile-sets/dell-dock-tb16-usb-audio.conf b/src/modules/alsa/mixer/profile-sets/dell-dock-tb16-usb-audio.conf new file mode 100644 index 0000000..1186552 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/dell-dock-tb16-usb-audio.conf @@ -0,0 +1,55 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Dell Dock TB16 USB audio +; +; This card has two stereo pairs of output, One Mono input. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-headphone] +description = Headphone +device-strings = hw:%f,0,0 +channel-map = left,right +direction = output + +[Mapping analog-stereo-speaker] +description = Speaker +device-strings = hw:%f,1,0 +channel-map = left,right +direction = output + +[Mapping analog-stereo-mic] +description = Headset-Mic +device-strings = hw:%f,0,0 +channel-map = left,right +direction = input + + +[Profile output:analog-stereo-speaker] +description = Speaker +output-mappings = analog-stereo-speaker +priority = 60 +skip-probe = yes + +[Profile output:analog-stereo-headphone+input:analog-stereo-mic] +description = Headset +output-mappings = analog-stereo-headphone +input-mappings = analog-stereo-mic +priority = 80 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf b/src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf new file mode 100644 index 0000000..41924f4 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf @@ -0,0 +1,153 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; This profile forces speaker and internal mic ports even if we have no way +; of identifying those. +; See default.conf for explanations. + +[General] +auto-profiles = yes + +[Mapping analog-mono] +device-strings = hw:%f +channel-map = mono +paths-output = analog-output analog-output-lineout analog-output-speaker-always analog-output-headphones analog-output-headphones-2 analog-output-mono +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic-always analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line +priority = 1 + +[Mapping analog-stereo] +device-strings = front:%f hw:%f +channel-map = left,right +paths-output = analog-output analog-output-lineout analog-output-speaker-always analog-output-headphones analog-output-headphones-2 analog-output-mono +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic-always analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line +priority = 10 + +[Mapping analog-surround-21] +device-strings = surround21:%f +channel-map = front-left,front-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 8 +direction = output + +[Mapping analog-surround-40] +device-strings = surround40:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 7 +direction = output + +[Mapping analog-surround-41] +device-strings = surround41:%f +channel-map = front-left,front-right,rear-left,rear-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 8 +direction = output + +[Mapping analog-surround-50] +device-strings = surround50:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 7 +direction = output + +[Mapping analog-surround-51] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 8 +direction = output + +[Mapping analog-surround-71] +device-strings = surround71:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +description = Analog Surround 7.1 +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 7 +direction = output + +[Mapping analog-4-channel-input] +# Alsa doesn't currently provide any better device name than "hw" for 4-channel +# input. If this causes trouble at some point, then we will need to get a new +# device name standardized in alsa. +device-strings = hw:%f +channel-map = aux0,aux1,aux2,aux3 +priority = 1 +direction = input + +[Mapping iec958-stereo] +device-strings = iec958:%f +channel-map = left,right +paths-input = iec958-stereo-input +paths-output = iec958-stereo-output +priority = 5 + +[Mapping iec958-ac3-surround-40] +device-strings = a52:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = iec958-stereo-output +priority = 2 +direction = output + +[Mapping iec958-ac3-surround-51] +device-strings = a52:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = iec958-stereo-output +priority = 3 +direction = output + +[Mapping iec958-dts-surround-51] +device-strings = dca:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = iec958-stereo-output +priority = 3 +direction = output + +[Mapping hdmi-stereo] +description = Digital Stereo (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = left,right +priority = 4 +direction = output + +[Mapping hdmi-surround] +description = Digital Surround 5.1 (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 3 +direction = output + +[Mapping hdmi-surround71] +description = Digital Surround 7.1 (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 3 +direction = output + +[Mapping hdmi-dts-surround] +description = Digital Surround 5.1 (HDMI/DTS) +device-strings = dcahdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 1 +direction = output + +; An example for defining multiple-sink profiles +#[Profile output:analog-stereo+output:iec958-stereo+input:analog-stereo] +#description = Foobar +#output-mappings = analog-stereo iec958-stereo +#input-mappings = analog-stereo diff --git a/src/modules/alsa/mixer/profile-sets/force-speaker.conf b/src/modules/alsa/mixer/profile-sets/force-speaker.conf new file mode 100644 index 0000000..dec57d5 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/force-speaker.conf @@ -0,0 +1,152 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; This profile forces a speaker port even if we have no way of identifying it. +; See default.conf for explanations. + +[General] +auto-profiles = yes + +[Mapping analog-mono] +device-strings = hw:%f +channel-map = mono +paths-output = analog-output analog-output-lineout analog-output-speaker-always analog-output-headphones analog-output-headphones-2 analog-output-mono +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line +priority = 1 + +[Mapping analog-stereo] +device-strings = front:%f hw:%f +channel-map = left,right +paths-output = analog-output analog-output-lineout analog-output-speaker-always analog-output-headphones analog-output-headphones-2 analog-output-mono +paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line +priority = 10 + +[Mapping analog-surround-21] +device-strings = surround21:%f +channel-map = front-left,front-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 8 +direction = output + +[Mapping analog-surround-40] +device-strings = surround40:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 7 +direction = output + +[Mapping analog-surround-41] +device-strings = surround41:%f +channel-map = front-left,front-right,rear-left,rear-right,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 8 +direction = output + +[Mapping analog-surround-50] +device-strings = surround50:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 7 +direction = output + +[Mapping analog-surround-51] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 8 +direction = output + +[Mapping analog-surround-71] +device-strings = surround71:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +description = Analog Surround 7.1 +paths-output = analog-output analog-output-lineout analog-output-speaker-always +priority = 7 +direction = output + +[Mapping analog-4-channel-input] +# Alsa doesn't currently provide any better device name than "hw" for 4-channel +# input. If this causes trouble at some point, then we will need to get a new +# device name standardized in alsa. +device-strings = hw:%f +channel-map = aux0,aux1,aux2,aux3 +priority = 1 +direction = input + +[Mapping iec958-stereo] +device-strings = iec958:%f +channel-map = left,right +paths-input = iec958-stereo-input +paths-output = iec958-stereo-output +priority = 5 + +[Mapping iec958-ac3-surround-40] +device-strings = a52:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = iec958-stereo-output +priority = 2 +direction = output + +[Mapping iec958-ac3-surround-51] +device-strings = a52:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = iec958-stereo-output +priority = 3 +direction = output + +[Mapping iec958-dts-surround-51] +device-strings = dca:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = iec958-stereo-output +priority = 3 +direction = output + +[Mapping hdmi-stereo] +description = Digital Stereo (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = left,right +priority = 4 +direction = output + +[Mapping hdmi-surround] +description = Digital Surround 5.1 (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 3 +direction = output + +[Mapping hdmi-surround71] +description = Digital Surround 7.1 (HDMI) +device-strings = hdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right +priority = 3 +direction = output + +[Mapping hdmi-dts-surround] +description = Digital Surround 5.1 (HDMI/DTS) +device-strings = dcahdmi:%f +paths-output = hdmi-output-0 +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +priority = 1 +direction = output + +; An example for defining multiple-sink profiles +#[Profile output:analog-stereo+output:iec958-stereo+input:analog-stereo] +#description = Foobar +#output-mappings = analog-stereo iec958-stereo +#input-mappings = analog-stereo diff --git a/src/modules/alsa/mixer/profile-sets/kinect-audio.conf b/src/modules/alsa/mixer/profile-sets/kinect-audio.conf new file mode 100644 index 0000000..d51fd17 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/kinect-audio.conf @@ -0,0 +1,38 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Audio profile for the Microsoft Kinect Sensor device in UAC mode. +; +; Copyright (C) 2011 Antonio Ospite <ospite@studenti.unina.it> +; +; This device has an array of four microphones, and no playback capability. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping input-4-channels] +device-strings = hw:%f +channel-map = front-left,front-right,rear-left,rear-right +description = 4 Channels Input +direction = input +priority = 5 + +[Profile input:mic-array] +description = Microphone Array +input-mappings = input-4-channels +priority = 2 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/maudio-fasttrack-pro.conf b/src/modules/alsa/mixer/profile-sets/maudio-fasttrack-pro.conf new file mode 100644 index 0000000..5122907 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/maudio-fasttrack-pro.conf @@ -0,0 +1,86 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; M-Audio FastTrack Pro +; +; This card has one duplex stereo channel called A and an additional +; stereo output channel called B. +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-a-output] +description = Analog Stereo Channel A +device-strings = hw:%f,0,0 +channel-map = left,right +direction = output + +; Try both device 0 and device 1 for input, see +; http://mailman.alsa-project.org/pipermail/alsa-devel/2012-March/050701.html +[Mapping analog-stereo-a-input] +description = Analog Stereo Channel A +device-strings = hw:%f,0,0 hw:%f,1,0 +channel-map = left,right +direction = input + +[Mapping analog-stereo-b-output] +description = Analog Stereo Channel B +device-strings = hw:%f,1,0 +channel-map = left,right +direction = output + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Channel A, Analog Stereo output Channel B +output-mappings = analog-stereo-a-output analog-stereo-b-output +input-mappings = analog-stereo-a-input +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-a-output+input:analog-stereo-a-input] +description = Analog Stereo Duplex Channel A +output-mappings = analog-stereo-a-output +input-mappings = analog-stereo-a-input +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-b+input:analog-stereo-b] +description = Analog Stereo Output Channel B +output-mappings = analog-stereo-b-output +input-mappings = +priority = 50 +skip-probe = yes + +[Profile output:analog-stereo-a] +description = Analog Stereo Output Channel A +output-mappings = analog-stereo-a-output +priority = 5 +skip-probe = yes + +[Profile output:analog-stereo-b] +description = Analog Stereo Output Channel B +output-mappings = analog-stereo-b-output +priority = 6 +skip-probe = yes + +[Profile input:analog-stereo-a] +description = Analog Stereo Input Channel A +input-mappings = analog-stereo-a-input +priority = 2 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-audio4dj.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-audio4dj.conf new file mode 100644 index 0000000..f7cbc15 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-audio4dj.conf @@ -0,0 +1,90 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Native Instruments Audio 4 DJ +; +; This card has two stereo pairs of input and two stereo pairs of +; output, named channels A and B. Channel B has an additional +; Headphone connector. +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-b-output] +description = Analog Stereo Channel B (Headphones) +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-b-input] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Channels A, B (Headphones) +output-mappings = analog-stereo-a analog-stereo-b-output +input-mappings = analog-stereo-a analog-stereo-b-input +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-a] +description = Analog Stereo Duplex Channel A +output-mappings = analog-stereo-a +input-mappings = analog-stereo-a +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-b+input:analog-stereo-b] +description = Analog Stereo Duplex Channel B (Headphones) +output-mappings = analog-stereo-b-output +input-mappings = analog-stereo-b-input +priority = 50 +skip-probe = yes + +[Profile output:analog-stereo-a] +description = Analog Stereo Output Channel A +output-mappings = analog-stereo-a +priority = 5 +skip-probe = yes + +[Profile output:analog-stereo-b] +description = Analog Stereo Output Channel B (Headphones) +output-mappings = analog-stereo-b-output +priority = 6 +skip-probe = yes + +[Profile input:analog-stereo-a] +description = Analog Stereo Input Channel A +input-mappings = analog-stereo-a +priority = 2 +skip-probe = yes + +[Profile input:analog-stereo-b] +description = Analog Stereo Input Channel B +input-mappings = analog-stereo-b-input +priority = 1 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-audio8dj.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-audio8dj.conf new file mode 100644 index 0000000..dc1b780 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-audio8dj.conf @@ -0,0 +1,161 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Native Instruments Audio 8 DJ +; +; This card has four stereo pairs of input and four stereo pairs of +; output, named channels A to D. Channel C has an additional Mic/Line +; connector, channel D an additional Headphone connector. +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-b] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right + +# Since we want to set a different description for channel C's/D's input +# and output we define two separate mappings for them +[Mapping analog-stereo-c-output] +description = Analog Stereo Channel C +device-strings = hw:%f,0,2 +channel-map = left,right +direction = output + +[Mapping analog-stereo-c-input] +description = Analog Stereo Channel C (Line/Mic) +device-strings = hw:%f,0,2 +channel-map = left,right +direction = input + +[Mapping analog-stereo-d-output] +description = Analog Stereo Channel D (Headphones) +device-strings = hw:%f,0,3 +channel-map = left,right +direction = output + +[Mapping analog-stereo-d-input] +description = Analog Stereo Channel D +device-strings = hw:%f,0,3 +channel-map = left,right +direction = input + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Channels A, B, C (Line/Mic), D (Headphones) +output-mappings = analog-stereo-a analog-stereo-b analog-stereo-c-output analog-stereo-d-output +input-mappings = analog-stereo-a analog-stereo-b analog-stereo-c-input analog-stereo-d-input +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-d+input:analog-stereo-c] +description = Analog Stereo Channel D (Headphones) Output, Channel C (Line/Mic) Input +output-mappings = analog-stereo-d-output +input-mappings = analog-stereo-c-input +priority = 90 +skip-probe = yes + +[Profile output:analog-stereo-c-d+input:analog-stereo-c-d] +description = Analog Stereo Duplex Channels C (Line/Mic), D (Line/Mic) +output-mappings = analog-stereo-c-output analog-stereo-d-output +input-mappings = analog-stereo-c-input analog-stereo-d-input +priority = 80 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-a] +description = Analog Stereo Duplex Channel A +output-mappings = analog-stereo-a +input-mappings = analog-stereo-a +priority = 50 +skip-probe = yes + +[Profile output:analog-stereo-b+input:analog-stereo-b] +description = Analog Stereo Duplex Channel B +output-mappings = analog-stereo-b +input-mappings = analog-stereo-b +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-c+input:analog-stereo-c] +description = Analog Stereo Duplex Channel C (Line/Mic) +output-mappings = analog-stereo-c-output +input-mappings = analog-stereo-c-input +priority = 60 +skip-probe = yes + +[Profile output:analog-stereo-d+input:analog-stereo-d] +description = Analog Stereo Duplex Channel D (Headphones) +output-mappings = analog-stereo-d-output +input-mappings = analog-stereo-d-input +priority = 70 +skip-probe = yes + +[Profile output:analog-stereo-a] +description = Analog Stereo Output Channel A +output-mappings = analog-stereo-a +priority = 6 +skip-probe = yes + +[Profile output:analog-stereo-b] +description = Analog Stereo Output Channel B +output-mappings = analog-stereo-b +priority = 5 +skip-probe = yes + +[Profile output:analog-stereo-c] +description = Analog Stereo Output Channel C +output-mappings = analog-stereo-c-output +priority = 7 +skip-probe = yes + +[Profile output:analog-stereo-d] +description = Analog Stereo Output Channel D (Headphones) +output-mappings = analog-stereo-d-output +priority = 8 +skip-probe = yes + +[Profile input:analog-stereo-a] +description = Analog Stereo Input Channel A +input-mappings = analog-stereo-a +priority = 2 +skip-probe = yes + +[Profile input:analog-stereo-b] +description = Analog Stereo Input Channel B +input-mappings = analog-stereo-b +priority = 1 +skip-probe = yes + +[Profile input:analog-stereo-c] +description = Analog Stereo Input Channel C (Line/Mic) +input-mappings = analog-stereo-c-input +priority = 4 +skip-probe = yes + +[Profile input:analog-stereo-d] +description = Analog Stereo Input Channel D +input-mappings = analog-stereo-d-input +priority = 3 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-korecontroller.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-korecontroller.conf new file mode 100644 index 0000000..35b3d06 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-korecontroller.conf @@ -0,0 +1,84 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Native Instruments Kore Controller +; +; This card has one stereo pairs of input and two stereo pairs of +; output, named "Master" and "Headphone". The master channel has +; an additional Coax S/PDIF connector which is always on. +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-master-out] +description = Analog Stereo Master Channel +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-headphone-out] +description = Analog Stereo Headphone Channel +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-input] +description = Analog Stereo +device-strings = hw:%f,0,0 +channel-map = left,right +direction = input + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Master Output, Headphones Output +output-mappings = analog-stereo-master-out analog-stereo-headphone-out +input-mappings = analog-stereo-input +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-master+input:analog-stereo-input] +description = Analog Stereo Duplex Master Output +output-mappings = analog-stereo-master-out +input-mappings = analog-stereo-input +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-headphone-out+input:analog-stereo-input] +description = Analog Stereo Headphones Output +output-mappings = analog-stereo-headphone-out +input-mappings = analog-stereo-input +priority = 30 +skip-probe = yes + +[Profile output:analog-stereo-master] +description = Analog Stereo Master Output +output-mappings = analog-stereo-master-out +priority = 3 +skip-probe = yes + +[Profile output:analog-stereo-headphone] +description = Analog Stereo Headphones Output +output-mappings = analog-stereo-headphone-out +priority = 2 +skip-probe = yes + +[Profile input:analog-stereo-input] +description = Analog Stereo Input +input-mappings = analog-stereo-input +priority = 1 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio10.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio10.conf new file mode 100644 index 0000000..c210297 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio10.conf @@ -0,0 +1,130 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Native Instruments Audio 10 DJ +; +; This card has five stereo pairs of input and five stereo pairs of +; output +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-out-main] +description = Analog Stereo Main +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-out-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-out-b] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-out-c] +description = Analog Stereo Channel C +device-strings = hw:%f,0,2 +channel-map = left,right +direction = output + +[Mapping analog-stereo-out-d] +description = Analog Stereo Channel D +device-strings = hw:%f,0,3 +channel-map = left,right +direction = output + +[Mapping analog-stereo-in-main] +description = Analog Stereo Main +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-in-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + +[Mapping analog-stereo-in-b] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + +[Mapping analog-stereo-in-c] +description = Analog Stereo Channel C +device-strings = hw:%f,0,2 +channel-map = left,right +direction = input + +[Mapping analog-stereo-in-d] +description = Analog Stereo Channel D +device-strings = hw:%f,0,3 +channel-map = left,right +direction = input + + + + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Channels Main, A, B, C, D +output-mappings = analog-stereo-out-main analog-stereo-out-a analog-stereo-out-b analog-stereo-out-c analog-stereo-out-d +input-mappings = analog-stereo-in-main analog-stereo-in-a analog-stereo-in-b analog-stereo-in-c analog-stereo-in-d +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-main+input:analog-stereo-main] +description = Analog Stereo Duplex Main +output-mappings = analog-stereo-out-main +input-mappings = analog-stereo-in-main +priority = 50 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-a] +description = Analog Stereo Duplex Channel A +output-mappings = analog-stereo-out-a +input-mappings = analog-stereo-in-a +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-b+input:analog-stereo-b] +description = Analog Stereo Duplex Channel B +output-mappings = analog-stereo-out-b +input-mappings = analog-stereo-in-b +priority = 30 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-c] +description = Analog Stereo Duplex Channel C +output-mappings = analog-stereo-out-c +input-mappings = analog-stereo-in-c +priority = 20 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-d] +description = Analog Stereo Duplex Channel D +output-mappings = analog-stereo-out-d +input-mappings = analog-stereo-in-d +priority = 10 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio2.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio2.conf new file mode 100644 index 0000000..145dace --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio2.conf @@ -0,0 +1,53 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Native Instruments Traktor Audio 2 +; +; This card has two stereo pairs of output. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,0 +channel-map = left,right +direction = output + +[Mapping analog-stereo-b] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Profile output:analog-stereo-a] +description = Analog Stereo Output Channel A +output-mappings = analog-stereo-a +priority = 60 +skip-probe = yes + +[Profile output:analog-stereo-b] +description = Analog Stereo Output Channel B +output-mappings = analog-stereo-b +priority = 50 +skip-probe = yes + +[Profile analog-stereo-all] +description = Analog Stereo Output Channels A & B +output-mappings = analog-stereo-a analog-stereo-b +priority = 100 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio6.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio6.conf new file mode 100644 index 0000000..a08e9fc --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-traktor-audio6.conf @@ -0,0 +1,91 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Native Instruments Audio 6 DJ +; +; This card has three stereo pairs of input and three stereo pairs of +; output +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-out-main] +description = Analog Stereo Main +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-out-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-out-b] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-in-main] +description = Analog Stereo Main +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-in-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + +[Mapping analog-stereo-in-b] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + + + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Channels A, B (Headphones) +output-mappings = analog-stereo-out-main analog-stereo-out-a analog-stereo-out-b +input-mappings = analog-stereo-in-main analog-stereo-in-a analog-stereo-in-b +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-main+input:analog-stereo-main] +description = Analog Stereo Duplex Channel Main +output-mappings = analog-stereo-out-main +input-mappings = analog-stereo-in-main +priority = 50 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-a] +description = Analog Stereo Duplex Channel A +output-mappings = analog-stereo-out-a +input-mappings = analog-stereo-in-a +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-b+input:analog-stereo-b] +description = Analog Stereo Duplex Channel B +output-mappings = analog-stereo-out-b +input-mappings = analog-stereo-in-b +priority = 30 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-traktorkontrol-s4.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-traktorkontrol-s4.conf new file mode 100644 index 0000000..934965f --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-traktorkontrol-s4.conf @@ -0,0 +1,80 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Native Instruments Traktor Kontrol S4 +; +; This controller has two stereo pairs of input (named "Channel C" and +; "Channel D") and two stereo pairs of output, one "Main Out" and +; "Headphone Out". +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-output-main] +description = Analog Stereo Main Out +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-output-headphone] +description = Analog Stereo Headphones Out +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-c-input] +description = Analog Stereo Channel C +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + +[Mapping analog-stereo-d-input] +description = Analog Stereo Channel D +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex +output-mappings = analog-stereo-output-main analog-stereo-output-headphone +input-mappings = analog-stereo-c-input analog-stereo-d-input +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-main] +description = Analog Stereo Main Output +output-mappings = analog-stereo-output-main +priority = 4 +skip-probe = yes + +[Profile output:analog-stereo-headphone] +description = Analog Stereo Output Headphones Out +output-mappings = analog-stereo-output-headphone +priority = 3 +skip-probe = yes + +[Profile input:analog-stereo-c] +description = Analog Stereo Input Channel C +input-mappings = analog-stereo-c-input +priority = 2 +skip-probe = yes + +[Profile input:analog-stereo-d] +description = Analog Stereo Input Channel D +input-mappings = analog-stereo-d-input +priority = 1 +skip-probe = yes + diff --git a/src/modules/alsa/mixer/profile-sets/sb-omni-surround-5.1.conf b/src/modules/alsa/mixer/profile-sets/sb-omni-surround-5.1.conf new file mode 100644 index 0000000..d5d1d65 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/sb-omni-surround-5.1.conf @@ -0,0 +1,112 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; Creative Sound Blaster Omni Surround 5.1 +; +; This config supports Linux 4.3-rc1+. +; By default there are some non-existing (physically) inputs and outputs that +; are not present in this config. +; Also in addition to natively supported modes (such as stereo, 5.1 and stereo +; S/PDIF) following useful output modes are added: 2.1, 4.0, 4.1 and 5.0. +; +; NOTE: in 2.1 and 4.1 physical LFE output will be different than in 5.1 mode. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-input] +device-strings = hw:%f +channel-map = left,right +paths-input = analog-input-mic analog-input-linein +direction = input + +[Mapping analog-stereo-output] +device-strings = front:%f +channel-map = left,right +paths-output = analog-output +direction = output + +[Mapping analog-surround-21] +device-strings = surround51:%f +channel-map = front-left,front-right,aux1,aux2,aux3,lfe +paths-output = analog-output +direction = output + +[Mapping analog-surround-40] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right +paths-output = analog-output +direction = output + +[Mapping analog-surround-41] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,aux1,lfe +paths-output = analog-output +direction = output + +[Mapping analog-surround-50] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center +paths-output = analog-output +direction = output + +[Mapping analog-surround-51] +device-strings = surround51:%f +channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe +paths-output = analog-output +direction = output + +[Mapping iec958-stereo] +device-strings = iec958:%f +channel-map = left,right +paths-output = iec958-stereo-output +direction = output + +[Profile output:analog-stereo-output+input:analog-stereo-input] +output-mappings = analog-stereo-output +input-mappings = analog-stereo-input +priority = 7 + +[Profile output:analog-surround-21+input:analog-stereo-input] +output-mappings = analog-surround-21 +input-mappings = analog-stereo-input +priority = 6 + +[Profile output:analog-surround-40+input:analog-stereo-input] +output-mappings = analog-surround-40 +input-mappings = analog-stereo-input +priority = 5 + +[Profile output:analog-surround-41+input:analog-stereo-input] +output-mappings = analog-surround-41 +input-mappings = analog-stereo-input +priority = 4 + +[Profile output:analog-surround-50+input:analog-stereo-input] +output-mappings = analog-surround-50 +input-mappings = analog-stereo-input +priority = 3 + +[Profile output:analog-surround-51+input:analog-stereo-input] +output-mappings = analog-surround-51 +input-mappings = analog-stereo-input +priority = 2 + +[Profile output:iec958-stereo+input:analog-stereo-input] +output-mappings = iec958-stereo +input-mappings = analog-stereo-input +priority = 1 diff --git a/src/modules/alsa/mixer/profile-sets/steelseries-arctis-common-usb-audio.conf b/src/modules/alsa/mixer/profile-sets/steelseries-arctis-common-usb-audio.conf new file mode 100644 index 0000000..0c58917 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/steelseries-arctis-common-usb-audio.conf @@ -0,0 +1,23 @@ +[General] +auto-profiles = yes + +[Mapping analog-chat] +description-key = gaming-headset-chat +device-strings = hw:%f,0,0 +channel-map = left,right +paths-input = analog-input-mic +paths-output = steelseries-arctis-output-chat-common +intended-roles = phone + +[Mapping analog-game] +description-key = gaming-headset-game +device-strings = hw:%f,1,0 +channel-map = left,right +paths-output = steelseries-arctis-output-game-common +direction = output + +[Profile output:analog-chat+output:analog-game+input:analog-chat] +output-mappings = analog-chat analog-game +input-mappings = analog-chat +priority = 5100 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/usb-gaming-headset.conf b/src/modules/alsa/mixer/profile-sets/usb-gaming-headset.conf new file mode 100644 index 0000000..adda54d --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/usb-gaming-headset.conf @@ -0,0 +1,64 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. + +; USB gaming headset. +; These headsets usually have two output devices. The first one is meant +; for voice audio, and the second one is meant for everything else. +; The purpose of this unusual design is to provide separate volume +; controls for voice and other audio, which can be useful in gaming. +; +; Works with: +; Steelseries Arctis 7 +; Steelseries Arctis Pro Wireless. +; Lucidsound LS31 +; Astro A50 +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = yes + +[Mapping mono-chat] +description-key = gaming-headset-chat +device-strings = hw:%f,0,0 +channel-map = mono +paths-output = usb-gaming-headset-output-mono +paths-input = usb-gaming-headset-input +intended-roles = phone + +[Mapping stereo-chat] +description-key = gaming-headset-chat +device-strings = hw:%f,0,0 +channel-map = left,right +paths-output = usb-gaming-headset-output-stereo +paths-input = usb-gaming-headset-input +intended-roles = phone + +[Mapping stereo-game] +description-key = gaming-headset-game +device-strings = hw:%f,1,0 +channel-map = left,right +paths-output = usb-gaming-headset-output-stereo +direction = output + +[Profile output:mono-chat+output:stereo-game+input:mono-chat] +output-mappings = mono-chat stereo-game +input-mappings = mono-chat +priority = 5100 + +[Profile output:stereo-game+output:stereo-chat+input:mono-chat] +output-mappings = stereo-game stereo-chat +input-mappings = mono-chat +priority = 5100 diff --git a/src/modules/alsa/module-alsa-card.c b/src/modules/alsa/module-alsa-card.c new file mode 100644 index 0000000..08e655e --- /dev/null +++ b/src/modules/alsa/module-alsa-card.c @@ -0,0 +1,1115 @@ +/*** + This file is part of PulseAudio. + + Copyright 2009 Lennart Poettering + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <pulse/xmalloc.h> + +#include <pulsecore/core-util.h> +#include <pulsecore/i18n.h> +#include <pulsecore/modargs.h> +#include <pulsecore/queue.h> + +#include <modules/reserve-wrap.h> + +#ifdef HAVE_UDEV +#include <modules/udev-util.h> +#endif + +#include "alsa-util.h" +#include "alsa-ucm.h" +#include "alsa-sink.h" +#include "alsa-source.h" + +PA_MODULE_AUTHOR("Lennart Poettering"); +PA_MODULE_DESCRIPTION("ALSA Card"); +PA_MODULE_VERSION(PACKAGE_VERSION); +PA_MODULE_LOAD_ONCE(false); +PA_MODULE_USAGE( + "name=<name for the card/sink/source, to be prefixed> " + "card_name=<name for the card> " + "card_properties=<properties for the card> " + "sink_name=<name for the sink> " + "sink_properties=<properties for the sink> " + "source_name=<name for the source> " + "source_properties=<properties for the source> " + "namereg_fail=<when false attempt to synthesise new names if they are already taken> " + "device_id=<ALSA card index> " + "format=<sample format> " + "rate=<sample rate> " + "fragments=<number of fragments> " + "fragment_size=<fragment size> " + "mmap=<enable memory mapping?> " + "tsched=<enable system timer based scheduling mode?> " + "tsched_buffer_size=<buffer size when using timer based scheduling> " + "tsched_buffer_watermark=<lower fill watermark> " + "profile=<profile name> " + "fixed_latency_range=<disable latency range changes on underrun?> " + "ignore_dB=<ignore dB information from the device?> " + "deferred_volume=<Synchronize software and hardware volume changes to avoid momentary jumps?> " + "profile_set=<profile set configuration file> " + "paths_dir=<directory containing the path configuration files> " + "use_ucm=<load use case manager> " + "avoid_resampling=<use stream original sample rate if possible?> " + "control=<name of mixer control> " +); + +static const char* const valid_modargs[] = { + "name", + "card_name", + "card_properties", + "sink_name", + "sink_properties", + "source_name", + "source_properties", + "namereg_fail", + "device_id", + "format", + "rate", + "fragments", + "fragment_size", + "mmap", + "tsched", + "tsched_buffer_size", + "tsched_buffer_watermark", + "fixed_latency_range", + "profile", + "ignore_dB", + "deferred_volume", + "profile_set", + "paths_dir", + "use_ucm", + "avoid_resampling", + "control", + NULL +}; + +#define DEFAULT_DEVICE_ID "0" + +struct userdata { + pa_core *core; + pa_module *module; + + char *device_id; + int alsa_card_index; + + pa_hashmap *mixers; + pa_hashmap *jacks; + + pa_card *card; + + pa_modargs *modargs; + + pa_alsa_profile_set *profile_set; + + /* ucm stuffs */ + bool use_ucm; + pa_alsa_ucm_config ucm; + +}; + +struct profile_data { + pa_alsa_profile *profile; +}; + +static void add_profiles(struct userdata *u, pa_hashmap *h, pa_hashmap *ports) { + pa_alsa_profile *ap; + void *state; + + pa_assert(u); + pa_assert(h); + + PA_HASHMAP_FOREACH(ap, u->profile_set->profiles, state) { + struct profile_data *d; + pa_card_profile *cp; + pa_alsa_mapping *m; + uint32_t idx; + + cp = pa_card_profile_new(ap->name, ap->description, sizeof(struct profile_data)); + cp->priority = ap->priority ? ap->priority : 1; + cp->input_name = pa_xstrdup(ap->input_name); + cp->output_name = pa_xstrdup(ap->output_name); + + if (ap->output_mappings) { + cp->n_sinks = pa_idxset_size(ap->output_mappings); + + PA_IDXSET_FOREACH(m, ap->output_mappings, idx) { + if (u->use_ucm) + pa_alsa_ucm_add_ports_combination(NULL, &m->ucm_context, true, ports, cp, u->core); + else + pa_alsa_path_set_add_ports(m->output_path_set, cp, ports, NULL, u->core); + if (m->channel_map.channels > cp->max_sink_channels) + cp->max_sink_channels = m->channel_map.channels; + } + } + + if (ap->input_mappings) { + cp->n_sources = pa_idxset_size(ap->input_mappings); + + PA_IDXSET_FOREACH(m, ap->input_mappings, idx) { + if (u->use_ucm) + pa_alsa_ucm_add_ports_combination(NULL, &m->ucm_context, false, ports, cp, u->core); + else + pa_alsa_path_set_add_ports(m->input_path_set, cp, ports, NULL, u->core); + if (m->channel_map.channels > cp->max_source_channels) + cp->max_source_channels = m->channel_map.channels; + } + } + + d = PA_CARD_PROFILE_DATA(cp); + d->profile = ap; + + pa_hashmap_put(h, cp->name, cp); + } +} + +static void add_disabled_profile(pa_hashmap *profiles) { + pa_card_profile *p; + struct profile_data *d; + + p = pa_card_profile_new("off", _("Off"), sizeof(struct profile_data)); + + d = PA_CARD_PROFILE_DATA(p); + d->profile = NULL; + + pa_hashmap_put(profiles, p->name, p); +} + +static int card_set_profile(pa_card *c, pa_card_profile *new_profile) { + struct userdata *u; + struct profile_data *nd, *od; + uint32_t idx; + pa_alsa_mapping *am; + pa_queue *sink_inputs = NULL, *source_outputs = NULL; + int ret = 0; + + pa_assert(c); + pa_assert(new_profile); + pa_assert_se(u = c->userdata); + + nd = PA_CARD_PROFILE_DATA(new_profile); + od = PA_CARD_PROFILE_DATA(c->active_profile); + + if (od->profile && od->profile->output_mappings) + PA_IDXSET_FOREACH(am, od->profile->output_mappings, idx) { + if (!am->sink) + continue; + + if (nd->profile && + nd->profile->output_mappings && + pa_idxset_get_by_data(nd->profile->output_mappings, am, NULL)) + continue; + + sink_inputs = pa_sink_move_all_start(am->sink, sink_inputs); + pa_alsa_sink_free(am->sink); + am->sink = NULL; + } + + if (od->profile && od->profile->input_mappings) + PA_IDXSET_FOREACH(am, od->profile->input_mappings, idx) { + if (!am->source) + continue; + + if (nd->profile && + nd->profile->input_mappings && + pa_idxset_get_by_data(nd->profile->input_mappings, am, NULL)) + continue; + + source_outputs = pa_source_move_all_start(am->source, source_outputs); + pa_alsa_source_free(am->source); + am->source = NULL; + } + + /* if UCM is available for this card then update the verb */ + if (u->use_ucm) { + if (pa_alsa_ucm_set_profile(&u->ucm, c, nd->profile ? nd->profile->name : NULL, + od->profile ? od->profile->name : NULL) < 0) { + ret = -1; + goto finish; + } + } + + if (nd->profile && nd->profile->output_mappings) + PA_IDXSET_FOREACH(am, nd->profile->output_mappings, idx) { + + if (!am->sink) + am->sink = pa_alsa_sink_new(c->module, u->modargs, __FILE__, c, am); + + if (sink_inputs && am->sink) { + pa_sink_move_all_finish(am->sink, sink_inputs, false); + sink_inputs = NULL; + } + } + + if (nd->profile && nd->profile->input_mappings) + PA_IDXSET_FOREACH(am, nd->profile->input_mappings, idx) { + + if (!am->source) + am->source = pa_alsa_source_new(c->module, u->modargs, __FILE__, c, am); + + if (source_outputs && am->source) { + pa_source_move_all_finish(am->source, source_outputs, false); + source_outputs = NULL; + } + } + +finish: + if (sink_inputs) + pa_sink_move_all_fail(sink_inputs); + + if (source_outputs) + pa_source_move_all_fail(source_outputs); + + return ret; +} + +static void init_profile(struct userdata *u) { + uint32_t idx; + pa_alsa_mapping *am; + struct profile_data *d; + pa_alsa_ucm_config *ucm = &u->ucm; + + pa_assert(u); + + d = PA_CARD_PROFILE_DATA(u->card->active_profile); + + if (d->profile && u->use_ucm) { + /* Set initial verb */ + if (pa_alsa_ucm_set_profile(ucm, u->card, d->profile->name, NULL) < 0) { + pa_log("Failed to set ucm profile %s", d->profile->name); + return; + } + } + + if (d->profile && d->profile->output_mappings) + PA_IDXSET_FOREACH(am, d->profile->output_mappings, idx) + am->sink = pa_alsa_sink_new(u->module, u->modargs, __FILE__, u->card, am); + + if (d->profile && d->profile->input_mappings) + PA_IDXSET_FOREACH(am, d->profile->input_mappings, idx) + am->source = pa_alsa_source_new(u->module, u->modargs, __FILE__, u->card, am); +} + +static pa_available_t calc_port_state(pa_device_port *p, struct userdata *u) { + void *state; + pa_alsa_jack *jack; + pa_available_t pa = PA_AVAILABLE_UNKNOWN; + pa_device_port *port; + + PA_HASHMAP_FOREACH(jack, u->jacks, state) { + pa_available_t cpa; + + if (u->use_ucm) + port = pa_hashmap_get(u->card->ports, jack->name); + else { + if (jack->path) + port = jack->path->port; + else + continue; + } + + if (p != port) + continue; + + cpa = jack->plugged_in ? jack->state_plugged : jack->state_unplugged; + + if (cpa == PA_AVAILABLE_NO) { + /* If a plugged-in jack causes the availability to go to NO, it + * should override all other availability information (like a + * blacklist) so set and bail */ + if (jack->plugged_in) { + pa = cpa; + break; + } + + /* If the current availablility is unknown go the more precise no, + * but otherwise don't change state */ + if (pa == PA_AVAILABLE_UNKNOWN) + pa = cpa; + } else if (cpa == PA_AVAILABLE_YES) { + /* Output is available through at least one jack, so go to that + * level of availability. We still need to continue iterating through + * the jacks in case a jack is plugged in that forces the state to no + */ + pa = cpa; + } + } + return pa; +} + +struct temp_port_avail { + pa_device_port *port; + pa_available_t avail; +}; + +static int report_jack_state(snd_mixer_elem_t *melem, unsigned int mask) { + struct userdata *u = snd_mixer_elem_get_callback_private(melem); + snd_hctl_elem_t *elem = snd_mixer_elem_get_private(melem); + snd_ctl_elem_value_t *elem_value; + bool plugged_in; + void *state; + pa_alsa_jack *jack; + struct temp_port_avail *tp, *tports; + pa_card_profile *profile; + pa_available_t active_available = PA_AVAILABLE_UNKNOWN; + + pa_assert(u); + + /* Changing the jack state may cause a port change, and a port change will + * make the sink or source change the mixer settings. If there are multiple + * users having pulseaudio running, the mixer changes done by inactive + * users may mess up the volume settings for the active users, because when + * the inactive users change the mixer settings, those changes are picked + * up by the active user's pulseaudio instance and the changes are + * interpreted as if the active user changed the settings manually e.g. + * with alsamixer. Even single-user systems suffer from this, because gdm + * runs its own pulseaudio instance. + * + * We rerun this function when being unsuspended to catch up on jack state + * changes */ + if (u->card->suspend_cause & PA_SUSPEND_SESSION) + return 0; + + if (mask == SND_CTL_EVENT_MASK_REMOVE) + return 0; + + snd_ctl_elem_value_alloca(&elem_value); + if (snd_hctl_elem_read(elem, elem_value) < 0) { + pa_log_warn("Failed to read jack detection from '%s'", pa_strnull(snd_hctl_elem_get_name(elem))); + return 0; + } + + plugged_in = !!snd_ctl_elem_value_get_boolean(elem_value, 0); + + pa_log_debug("Jack '%s' is now %s", pa_strnull(snd_hctl_elem_get_name(elem)), plugged_in ? "plugged in" : "unplugged"); + + tports = tp = pa_xnew0(struct temp_port_avail, pa_hashmap_size(u->jacks)+1); + + PA_HASHMAP_FOREACH(jack, u->jacks, state) + if (jack->melem == melem) { + pa_alsa_jack_set_plugged_in(jack, plugged_in); + + if (u->use_ucm) { + /* When using UCM, pa_alsa_jack_set_plugged_in() maps the jack + * state to port availability. */ + continue; + } + + /* When not using UCM, we have to do the jack state -> port + * availability mapping ourselves. */ + pa_assert_se(tp->port = jack->path->port); + tp->avail = calc_port_state(tp->port, u); + tp++; + } + + /* Report available ports before unavailable ones: in case port 1 becomes available when port 2 becomes unavailable, + this prevents an unnecessary switch port 1 -> port 3 -> port 2 */ + + for (tp = tports; tp->port; tp++) + if (tp->avail != PA_AVAILABLE_NO) + pa_device_port_set_available(tp->port, tp->avail); + for (tp = tports; tp->port; tp++) + if (tp->avail == PA_AVAILABLE_NO) + pa_device_port_set_available(tp->port, tp->avail); + + for (tp = tports; tp->port; tp++) { + pa_alsa_port_data *data; + pa_sink *sink; + uint32_t idx; + + data = PA_DEVICE_PORT_DATA(tp->port); + + if (!data->suspend_when_unavailable) + continue; + + PA_IDXSET_FOREACH(sink, u->core->sinks, idx) { + if (sink->active_port == tp->port) + pa_sink_suspend(sink, tp->avail == PA_AVAILABLE_NO, PA_SUSPEND_UNAVAILABLE); + } + } + + /* Update profile availabilities. Ideally we would mark all profiles + * unavailable that contain unavailable devices. We can't currently do that + * in all cases, because if there are multiple sinks in a profile, and the + * profile contains a mix of available and unavailable ports, we don't know + * how the ports are distributed between the different sinks. It's possible + * that some sinks contain only unavailable ports, in which case we should + * mark the profile as unavailable, but it's also possible that all sinks + * contain at least one available port, in which case we should mark the + * profile as available. Until the data structures are improved so that we + * can distinguish between these two cases, we mark the problematic cases + * as available (well, "unknown" to be precise, but there's little + * practical difference). + * + * When all output ports are unavailable, we know that all sinks are + * unavailable, and therefore the profile is marked unavailable as well. + * The same applies to input ports as well, of course. + * + * If there are no output ports at all, but the profile contains at least + * one sink, then the output is considered to be available. */ + if (u->card->active_profile) + active_available = u->card->active_profile->available; + PA_HASHMAP_FOREACH(profile, u->card->profiles, state) { + pa_device_port *port; + void *state2; + bool has_input_port = false; + bool has_output_port = false; + bool found_available_input_port = false; + bool found_available_output_port = false; + pa_available_t available = PA_AVAILABLE_UNKNOWN; + + PA_HASHMAP_FOREACH(port, u->card->ports, state2) { + if (!pa_hashmap_get(port->profiles, profile->name)) + continue; + + if (port->direction == PA_DIRECTION_INPUT) { + has_input_port = true; + + if (port->available != PA_AVAILABLE_NO) + found_available_input_port = true; + } else { + has_output_port = true; + + if (port->available != PA_AVAILABLE_NO) + found_available_output_port = true; + } + } + + if ((has_input_port && !found_available_input_port) || (has_output_port && !found_available_output_port)) + available = PA_AVAILABLE_NO; + + /* We want to update the active profile's status last, so logic that + * may change the active profile based on profile availability status + * has an updated view of all profiles' availabilities. */ + if (profile == u->card->active_profile) + active_available = available; + else + pa_card_profile_set_available(profile, available); + } + + if (u->card->active_profile) + pa_card_profile_set_available(u->card->active_profile, active_available); + + pa_xfree(tports); + return 0; +} + +static pa_device_port* find_port_with_eld_device(struct userdata *u, int device) { + void *state; + pa_device_port *p; + + if (u->use_ucm) { + PA_HASHMAP_FOREACH(p, u->card->ports, state) { + pa_alsa_ucm_port_data *data = PA_DEVICE_PORT_DATA(p); + pa_assert(data->eld_mixer_device_name); + if (device == data->eld_device) + return p; + } + } else { + PA_HASHMAP_FOREACH(p, u->card->ports, state) { + pa_alsa_port_data *data = PA_DEVICE_PORT_DATA(p); + pa_assert(data->path); + if (device == data->path->eld_device) + return p; + } + } + return NULL; +} + +static int hdmi_eld_changed(snd_mixer_elem_t *melem, unsigned int mask) { + struct userdata *u = snd_mixer_elem_get_callback_private(melem); + snd_hctl_elem_t *elem = snd_mixer_elem_get_private(melem); + int device = snd_hctl_elem_get_device(elem); + const char *old_monitor_name; + pa_device_port *p; + pa_hdmi_eld eld; + bool changed = false; + + if (mask == SND_CTL_EVENT_MASK_REMOVE) + return 0; + + p = find_port_with_eld_device(u, device); + if (p == NULL) { + pa_log_error("Invalid device changed in ALSA: %d", device); + return 0; + } + + if (pa_alsa_get_hdmi_eld(elem, &eld) < 0) + memset(&eld, 0, sizeof(eld)); + + old_monitor_name = pa_proplist_gets(p->proplist, PA_PROP_DEVICE_PRODUCT_NAME); + if (eld.monitor_name[0] == '\0') { + changed |= old_monitor_name != NULL; + pa_proplist_unset(p->proplist, PA_PROP_DEVICE_PRODUCT_NAME); + } else { + changed |= (old_monitor_name == NULL) || (strcmp(old_monitor_name, eld.monitor_name) != 0); + pa_proplist_sets(p->proplist, PA_PROP_DEVICE_PRODUCT_NAME, eld.monitor_name); + } + + if (changed && mask != 0) + pa_subscription_post(u->core, PA_SUBSCRIPTION_EVENT_CARD|PA_SUBSCRIPTION_EVENT_CHANGE, u->card->index); + + return 0; +} + +static void init_eld_ctls(struct userdata *u) { + void *state; + pa_device_port *port; + + /* The code in this function expects ports to have a pa_alsa_port_data + * struct as their data, but in UCM mode ports don't have any data. Hence, + * the ELD controls can't currently be used in UCM mode. */ + PA_HASHMAP_FOREACH(port, u->card->ports, state) { + snd_mixer_t *mixer_handle; + snd_mixer_elem_t* melem; + int device; + + if (u->use_ucm) { + pa_alsa_ucm_port_data *data = PA_DEVICE_PORT_DATA(port); + device = data->eld_device; + if (device < 0 || !data->eld_mixer_device_name) + continue; + + mixer_handle = pa_alsa_open_mixer_by_name(u->mixers, data->eld_mixer_device_name, true); + } else { + pa_alsa_port_data *data = PA_DEVICE_PORT_DATA(port); + + pa_assert(data->path); + + device = data->path->eld_device; + if (device < 0) + continue; + + mixer_handle = pa_alsa_open_mixer(u->mixers, u->alsa_card_index, true); + } + + if (!mixer_handle) + continue; + + melem = pa_alsa_mixer_find_pcm(mixer_handle, "ELD", device); + if (melem) { + pa_alsa_mixer_set_fdlist(u->mixers, mixer_handle, u->core->mainloop); + snd_mixer_elem_set_callback(melem, hdmi_eld_changed); + snd_mixer_elem_set_callback_private(melem, u); + hdmi_eld_changed(melem, 0); + pa_log_info("ELD device found for port %s (%d).", port->name, device); + } + else + pa_log_debug("No ELD device found for port %s (%d).", port->name, device); + } +} + +static void init_jacks(struct userdata *u) { + void *state; + pa_alsa_path* path; + pa_alsa_jack* jack; + char buf[64]; + + u->jacks = pa_hashmap_new(pa_idxset_trivial_hash_func, pa_idxset_trivial_compare_func); + + if (u->use_ucm) { + PA_LLIST_FOREACH(jack, u->ucm.jacks) + if (jack->has_control) + pa_hashmap_put(u->jacks, jack, jack); + } else { + /* See if we have any jacks */ + if (u->profile_set->output_paths) + PA_HASHMAP_FOREACH(path, u->profile_set->output_paths, state) + PA_LLIST_FOREACH(jack, path->jacks) + if (jack->has_control) + pa_hashmap_put(u->jacks, jack, jack); + + if (u->profile_set->input_paths) + PA_HASHMAP_FOREACH(path, u->profile_set->input_paths, state) + PA_LLIST_FOREACH(jack, path->jacks) + if (jack->has_control) + pa_hashmap_put(u->jacks, jack, jack); + } + + pa_log_debug("Found %d jacks.", pa_hashmap_size(u->jacks)); + + if (pa_hashmap_size(u->jacks) == 0) + return; + + PA_HASHMAP_FOREACH(jack, u->jacks, state) { + if (!jack->mixer_device_name) { + jack->mixer_handle = pa_alsa_open_mixer(u->mixers, u->alsa_card_index, false); + if (!jack->mixer_handle) { + pa_log("Failed to open mixer for card %d for jack detection", u->alsa_card_index); + continue; + } + } else { + jack->mixer_handle = pa_alsa_open_mixer_by_name(u->mixers, jack->mixer_device_name, false); + if (!jack->mixer_handle) { + pa_log("Failed to open mixer '%s' for jack detection", jack->mixer_device_name); + continue; + } + } + pa_alsa_mixer_set_fdlist(u->mixers, jack->mixer_handle, u->core->mainloop); + jack->melem = pa_alsa_mixer_find_card(jack->mixer_handle, &jack->alsa_id, 0); + if (!jack->melem) { + pa_alsa_mixer_id_to_string(buf, sizeof(buf), &jack->alsa_id); + pa_log_warn("Jack %s seems to have disappeared.", buf); + pa_alsa_jack_set_has_control(jack, false); + continue; + } + snd_mixer_elem_set_callback(jack->melem, report_jack_state); + snd_mixer_elem_set_callback_private(jack->melem, u); + report_jack_state(jack->melem, 0); + } +} + +static void prune_singleton_availability_groups(pa_hashmap *ports) { + pa_device_port *p; + pa_hashmap *group_counts; + void *state, *count; + const char *group; + + /* Collect groups and erase those that don't have more than 1 path */ + group_counts = pa_hashmap_new(pa_idxset_string_hash_func, pa_idxset_string_compare_func); + + PA_HASHMAP_FOREACH(p, ports, state) { + if (p->availability_group) { + count = pa_hashmap_get(group_counts, p->availability_group); + pa_hashmap_remove(group_counts, p->availability_group); + pa_hashmap_put(group_counts, p->availability_group, PA_UINT_TO_PTR(PA_PTR_TO_UINT(count) + 1)); + } + } + + /* Now we have an availability_group -> count map, let's drop all groups + * that have only one member */ + PA_HASHMAP_FOREACH_KV(group, count, group_counts, state) { + if (count == PA_UINT_TO_PTR(1)) + pa_hashmap_remove(group_counts, group); + } + + PA_HASHMAP_FOREACH(p, ports, state) { + if (p->availability_group && !pa_hashmap_get(group_counts, p->availability_group)) { + pa_log_debug("Pruned singleton availability group %s from port %s", p->availability_group, p->name); + + pa_xfree(p->availability_group); + p->availability_group = NULL; + } + } + + pa_hashmap_free(group_counts); +} + +static void set_card_name(pa_card_new_data *data, pa_modargs *ma, const char *device_id) { + char *t; + const char *n; + + pa_assert(data); + pa_assert(ma); + pa_assert(device_id); + + if ((n = pa_modargs_get_value(ma, "card_name", NULL))) { + pa_card_new_data_set_name(data, n); + data->namereg_fail = true; + return; + } + + if ((n = pa_modargs_get_value(ma, "name", NULL))) + data->namereg_fail = true; + else { + n = device_id; + data->namereg_fail = false; + } + + t = pa_sprintf_malloc("alsa_card.%s", n); + pa_card_new_data_set_name(data, t); + pa_xfree(t); +} + +static pa_hook_result_t card_suspend_changed(pa_core *c, pa_card *card, struct userdata *u) { + void *state; + pa_alsa_jack *jack; + + if (card->suspend_cause == 0) { + /* We were unsuspended, update jack state in case it changed while we were suspended */ + PA_HASHMAP_FOREACH(jack, u->jacks, state) { + if (jack->melem) + report_jack_state(jack->melem, 0); + } + } + + return PA_HOOK_OK; +} + +static pa_hook_result_t sink_input_put_hook_callback(pa_core *c, pa_sink_input *sink_input, struct userdata *u) { + const char *role; + pa_sink *sink = sink_input->sink; + + pa_assert(sink); + + role = pa_proplist_gets(sink_input->proplist, PA_PROP_MEDIA_ROLE); + + /* new sink input linked to sink of this card */ + if (role && sink->card == u->card) + pa_alsa_ucm_roled_stream_begin(&u->ucm, role, PA_DIRECTION_OUTPUT); + + return PA_HOOK_OK; +} + +static pa_hook_result_t source_output_put_hook_callback(pa_core *c, pa_source_output *source_output, struct userdata *u) { + const char *role; + pa_source *source = source_output->source; + + pa_assert(source); + + role = pa_proplist_gets(source_output->proplist, PA_PROP_MEDIA_ROLE); + + /* new source output linked to source of this card */ + if (role && source->card == u->card) + pa_alsa_ucm_roled_stream_begin(&u->ucm, role, PA_DIRECTION_INPUT); + + return PA_HOOK_OK; +} + +static pa_hook_result_t sink_input_unlink_hook_callback(pa_core *c, pa_sink_input *sink_input, struct userdata *u) { + const char *role; + pa_sink *sink = sink_input->sink; + + pa_assert(sink); + + role = pa_proplist_gets(sink_input->proplist, PA_PROP_MEDIA_ROLE); + + /* new sink input unlinked from sink of this card */ + if (role && sink->card == u->card) + pa_alsa_ucm_roled_stream_end(&u->ucm, role, PA_DIRECTION_OUTPUT); + + return PA_HOOK_OK; +} + +static pa_hook_result_t source_output_unlink_hook_callback(pa_core *c, pa_source_output *source_output, struct userdata *u) { + const char *role; + pa_source *source = source_output->source; + + pa_assert(source); + + role = pa_proplist_gets(source_output->proplist, PA_PROP_MEDIA_ROLE); + + /* new source output unlinked from source of this card */ + if (role && source->card == u->card) + pa_alsa_ucm_roled_stream_end(&u->ucm, role, PA_DIRECTION_INPUT); + + return PA_HOOK_OK; +} + +int pa__init(pa_module *m) { + pa_card_new_data data; + bool ignore_dB = false; + struct userdata *u; + pa_reserve_wrapper *reserve = NULL; + const char *description; + const char *profile_str = NULL; + char *fn = NULL; + bool namereg_fail = false; + int err = -PA_MODULE_ERR_UNSPECIFIED, rval; + + pa_alsa_refcnt_inc(); + + pa_assert(m); + + m->userdata = u = pa_xnew0(struct userdata, 1); + u->core = m->core; + u->module = m; + u->use_ucm = true; + u->ucm.core = m->core; + + u->mixers = pa_hashmap_new_full(pa_idxset_string_hash_func, pa_idxset_string_compare_func, + pa_xfree, (pa_free_cb_t) pa_alsa_mixer_free); + u->ucm.mixers = u->mixers; /* alias */ + + if (!(u->modargs = pa_modargs_new(m->argument, valid_modargs))) { + pa_log("Failed to parse module arguments."); + goto fail; + } + + u->device_id = pa_xstrdup(pa_modargs_get_value(u->modargs, "device_id", DEFAULT_DEVICE_ID)); + + if ((u->alsa_card_index = snd_card_get_index(u->device_id)) < 0) { + pa_log("Card '%s' doesn't exist: %s", u->device_id, pa_alsa_strerror(u->alsa_card_index)); + goto fail; + } + + if (pa_modargs_get_value_boolean(u->modargs, "ignore_dB", &ignore_dB) < 0) { + pa_log("Failed to parse ignore_dB argument."); + goto fail; + } + + if (!pa_in_system_mode()) { + char *rname; + + if ((rname = pa_alsa_get_reserve_name(u->device_id))) { + reserve = pa_reserve_wrapper_get(m->core, rname); + pa_xfree(rname); + + if (!reserve) + goto fail; + } + } + + if (pa_modargs_get_value_boolean(u->modargs, "use_ucm", &u->use_ucm) < 0) { + pa_log("Failed to parse use_ucm argument."); + goto fail; + } + + /* Force ALSA to reread its configuration. This matters if our device + * was hot-plugged after ALSA has already read its configuration - see + * https://bugs.freedesktop.org/show_bug.cgi?id=54029 + */ + + snd_config_update_free_global(); + + rval = u->use_ucm ? pa_alsa_ucm_query_profiles(&u->ucm, u->alsa_card_index) : -1; + if (rval == -PA_ALSA_ERR_UCM_LINKED) { + err = -PA_MODULE_ERR_SKIP; + goto fail; + } + if (rval == 0) { + pa_log_info("Found UCM profiles"); + + u->profile_set = pa_alsa_ucm_add_profile_set(&u->ucm, &u->core->default_channel_map); + + /* hook start of sink input/source output to enable modifiers */ + /* A little bit later than module-role-cork */ + pa_module_hook_connect(m, &m->core->hooks[PA_CORE_HOOK_SINK_INPUT_PUT], PA_HOOK_LATE+10, + (pa_hook_cb_t) sink_input_put_hook_callback, u); + pa_module_hook_connect(m, &m->core->hooks[PA_CORE_HOOK_SOURCE_OUTPUT_PUT], PA_HOOK_LATE+10, + (pa_hook_cb_t) source_output_put_hook_callback, u); + + /* hook end of sink input/source output to disable modifiers */ + /* A little bit later than module-role-cork */ + pa_module_hook_connect(m, &m->core->hooks[PA_CORE_HOOK_SINK_INPUT_UNLINK], PA_HOOK_LATE+10, + (pa_hook_cb_t) sink_input_unlink_hook_callback, u); + pa_module_hook_connect(m, &m->core->hooks[PA_CORE_HOOK_SOURCE_OUTPUT_UNLINK], PA_HOOK_LATE+10, + (pa_hook_cb_t) source_output_unlink_hook_callback, u); + } + else { + u->use_ucm = false; +#ifdef HAVE_UDEV + fn = pa_udev_get_property(u->alsa_card_index, "PULSE_PROFILE_SET"); +#endif + + if (pa_modargs_get_value(u->modargs, "profile_set", NULL)) { + pa_xfree(fn); + fn = pa_xstrdup(pa_modargs_get_value(u->modargs, "profile_set", NULL)); + } + + u->profile_set = pa_alsa_profile_set_new(fn, &u->core->default_channel_map); + pa_xfree(fn); + } + + if (!u->profile_set) + goto fail; + + u->profile_set->ignore_dB = ignore_dB; + + pa_alsa_profile_set_probe(u->profile_set, u->mixers, u->device_id, &m->core->default_sample_spec, m->core->default_n_fragments, m->core->default_fragment_size_msec); + pa_alsa_profile_set_dump(u->profile_set); + + pa_card_new_data_init(&data); + data.driver = __FILE__; + data.module = m; + + pa_alsa_init_proplist_card(m->core, data.proplist, u->alsa_card_index); + + pa_proplist_sets(data.proplist, PA_PROP_DEVICE_STRING, u->device_id); + pa_alsa_init_description(data.proplist, NULL); + set_card_name(&data, u->modargs, u->device_id); + + /* We need to give pa_modargs_get_value_boolean() a pointer to a local + * variable instead of using &data.namereg_fail directly, because + * data.namereg_fail is a bitfield and taking the address of a bitfield + * variable is impossible. */ + namereg_fail = data.namereg_fail; + if (pa_modargs_get_value_boolean(u->modargs, "namereg_fail", &namereg_fail) < 0) { + pa_log("Failed to parse namereg_fail argument."); + pa_card_new_data_done(&data); + goto fail; + } + data.namereg_fail = namereg_fail; + + if (reserve) + if ((description = pa_proplist_gets(data.proplist, PA_PROP_DEVICE_DESCRIPTION))) + pa_reserve_wrapper_set_application_device_name(reserve, description); + + add_profiles(u, data.profiles, data.ports); + + if (pa_hashmap_isempty(data.profiles)) { + pa_log("Failed to find a working profile."); + pa_card_new_data_done(&data); + goto fail; + } + + add_disabled_profile(data.profiles); + prune_singleton_availability_groups(data.ports); + + if (pa_modargs_get_proplist(u->modargs, "card_properties", data.proplist, PA_UPDATE_REPLACE) < 0) { + pa_log("Invalid properties"); + pa_card_new_data_done(&data); + goto fail; + } + + /* The Intel HDMI LPE driver needs some special handling. When the HDMI + * cable is not plugged in, trying to play audio doesn't work. Any written + * audio is immediately discarded and an underrun is reported, and that + * results in an infinite loop of "fill buffer, handle underrun". To work + * around this issue, the suspend_when_unavailable flag is used to stop + * playback when the HDMI cable is unplugged. */ + if (!u->use_ucm && + pa_safe_streq(pa_proplist_gets(data.proplist, "alsa.driver_name"), "snd_hdmi_lpe_audio")) { + pa_device_port *port; + void *state; + + PA_HASHMAP_FOREACH(port, data.ports, state) { + pa_alsa_port_data *port_data; + + port_data = PA_DEVICE_PORT_DATA(port); + port_data->suspend_when_unavailable = true; + } + } + + u->card = pa_card_new(m->core, &data); + pa_card_new_data_done(&data); + + if (!u->card) + goto fail; + + u->card->userdata = u; + u->card->set_profile = card_set_profile; + + pa_module_hook_connect(m, &m->core->hooks[PA_CORE_HOOK_CARD_SUSPEND_CHANGED], PA_HOOK_NORMAL, + (pa_hook_cb_t) card_suspend_changed, u); + + init_jacks(u); + + pa_card_choose_initial_profile(u->card); + + /* If the "profile" modarg is given, we have to override whatever the usual + * policy chose in pa_card_choose_initial_profile(). */ + profile_str = pa_modargs_get_value(u->modargs, "profile", NULL); + if (profile_str) { + pa_card_profile *profile; + + profile = pa_hashmap_get(u->card->profiles, profile_str); + if (!profile) { + pa_log("No such profile: %s", profile_str); + goto fail; + } + + pa_card_set_profile(u->card, profile, false); + } + + pa_card_put(u->card); + + init_profile(u); + init_eld_ctls(u); + + /* Remove all probe only mixers */ + if (u->mixers) { + const char *devname; + pa_alsa_mixer *pm; + void *state; + PA_HASHMAP_FOREACH_KV(devname, pm, u->mixers, state) + if (pm->used_for_probe_only) + pa_hashmap_remove_and_free(u->mixers, devname); + } + + if (reserve) + pa_reserve_wrapper_unref(reserve); + + if (!pa_hashmap_isempty(u->profile_set->decibel_fixes)) + pa_log_warn("Card %s uses decibel fixes (i.e. overrides the decibel information for some alsa volume elements). " + "Please note that this feature is meant just as a help for figuring out the correct decibel values. " + "PulseAudio is not the correct place to maintain the decibel mappings! The fixed decibel values " + "should be sent to ALSA developers so that they can fix the driver. If it turns out that this feature " + "is abused (i.e. fixes are not pushed to ALSA), the decibel fix feature may be removed in some future " + "PulseAudio version.", u->card->name); + + return 0; + +fail: + if (reserve) + pa_reserve_wrapper_unref(reserve); + + pa__done(m); + + return err; +} + +int pa__get_n_used(pa_module *m) { + struct userdata *u; + int n = 0; + uint32_t idx; + pa_sink *sink; + pa_source *source; + + pa_assert(m); + pa_assert_se(u = m->userdata); + pa_assert(u->card); + + PA_IDXSET_FOREACH(sink, u->card->sinks, idx) + n += pa_sink_linked_by(sink); + + PA_IDXSET_FOREACH(source, u->card->sources, idx) + n += pa_source_linked_by(source); + + return n; +} + +void pa__done(pa_module*m) { + struct userdata *u; + + pa_assert(m); + + if (!(u = m->userdata)) + goto finish; + + if (u->mixers) + pa_hashmap_free(u->mixers); + if (u->jacks) + pa_hashmap_free(u->jacks); + + if (u->card && u->card->sinks) + pa_idxset_remove_all(u->card->sinks, (pa_free_cb_t) pa_alsa_sink_free); + + if (u->card && u->card->sources) + pa_idxset_remove_all(u->card->sources, (pa_free_cb_t) pa_alsa_source_free); + + if (u->card) + pa_card_free(u->card); + + if (u->modargs) + pa_modargs_free(u->modargs); + + if (u->profile_set) + pa_alsa_profile_set_free(u->profile_set); + + pa_alsa_ucm_free(&u->ucm); + + pa_xfree(u->device_id); + pa_xfree(u); + +finish: + pa_alsa_refcnt_dec(); +} diff --git a/src/modules/alsa/module-alsa-sink.c b/src/modules/alsa/module-alsa-sink.c new file mode 100644 index 0000000..a90c5e4 --- /dev/null +++ b/src/modules/alsa/module-alsa-sink.c @@ -0,0 +1,137 @@ +/*** + This file is part of PulseAudio. + + Copyright 2004-2008 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <pulsecore/module.h> +#include <pulsecore/sink.h> +#include <pulsecore/modargs.h> + +#include "alsa-util.h" +#include "alsa-sink.h" + +PA_MODULE_AUTHOR("Lennart Poettering"); +PA_MODULE_DESCRIPTION("ALSA Sink"); +PA_MODULE_VERSION(PACKAGE_VERSION); +PA_MODULE_LOAD_ONCE(false); +PA_MODULE_USAGE( + "name=<name of the sink, to be prefixed> " + "sink_name=<name for the sink> " + "sink_properties=<properties for the sink> " + "namereg_fail=<when false attempt to synthesise new sink_name if it is already taken> " + "device=<ALSA device> " + "device_id=<ALSA card index> " + "format=<sample format> " + "rate=<sample rate> " + "alternate_rate=<alternate sample rate> " + "channels=<number of channels> " + "channel_map=<channel map> " + "fragments=<number of fragments> " + "fragment_size=<fragment size> " + "mmap=<enable memory mapping?> " + "tsched=<enable system timer based scheduling mode?> " + "tsched_buffer_size=<buffer size when using timer based scheduling> " + "tsched_buffer_watermark=<lower fill watermark> " + "ignore_dB=<ignore dB information from the device?> " + "control=<name of mixer control, or name and index separated by a comma> " + "rewind_safeguard=<number of bytes that cannot be rewound> " + "deferred_volume=<Synchronize software and hardware volume changes to avoid momentary jumps?> " + "deferred_volume_safety_margin=<usec adjustment depending on volume direction> " + "deferred_volume_extra_delay=<usec adjustment to HW volume changes> " + "fixed_latency_range=<disable latency range changes on underrun?>"); + +static const char* const valid_modargs[] = { + "name", + "sink_name", + "sink_properties", + "namereg_fail", + "device", + "device_id", + "format", + "rate", + "alternate_rate", + "channels", + "channel_map", + "fragments", + "fragment_size", + "mmap", + "tsched", + "tsched_buffer_size", + "tsched_buffer_watermark", + "ignore_dB", + "control", + "rewind_safeguard", + "deferred_volume", + "deferred_volume_safety_margin", + "deferred_volume_extra_delay", + "fixed_latency_range", + NULL +}; + +int pa__init(pa_module*m) { + pa_modargs *ma = NULL; + + pa_assert(m); + + pa_alsa_refcnt_inc(); + + if (!(ma = pa_modargs_new(m->argument, valid_modargs))) { + pa_log("Failed to parse module arguments"); + goto fail; + } + + if (!(m->userdata = pa_alsa_sink_new(m, ma, __FILE__, NULL, NULL))) + goto fail; + + pa_modargs_free(ma); + + return 0; + +fail: + + if (ma) + pa_modargs_free(ma); + + pa__done(m); + + return -1; +} + +int pa__get_n_used(pa_module *m) { + pa_sink *sink; + + pa_assert(m); + pa_assert_se(sink = m->userdata); + + return pa_sink_linked_by(sink); +} + +void pa__done(pa_module*m) { + pa_sink *sink; + + pa_assert(m); + + if ((sink = m->userdata)) + pa_alsa_sink_free(sink); + + pa_alsa_refcnt_dec(); +} diff --git a/src/modules/alsa/module-alsa-source.c b/src/modules/alsa/module-alsa-source.c new file mode 100644 index 0000000..d152283 --- /dev/null +++ b/src/modules/alsa/module-alsa-source.c @@ -0,0 +1,144 @@ +/*** + This file is part of PulseAudio. + + Copyright 2004-2008 Lennart Poettering + Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB + + PulseAudio is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as published + by the Free Software Foundation; either version 2.1 of the License, + or (at your option) any later version. + + PulseAudio is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU Lesser General Public License + along with PulseAudio; if not, see <http://www.gnu.org/licenses/>. +***/ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <stdio.h> + +#include <alsa/asoundlib.h> + +#ifdef HAVE_VALGRIND_MEMCHECK_H +#include <valgrind/memcheck.h> +#endif + +#include <pulsecore/module.h> +#include <pulsecore/modargs.h> +#include <pulsecore/log.h> +#include <pulsecore/macro.h> + +#include "alsa-util.h" +#include "alsa-source.h" + +PA_MODULE_AUTHOR("Lennart Poettering"); +PA_MODULE_DESCRIPTION("ALSA Source"); +PA_MODULE_VERSION(PACKAGE_VERSION); +PA_MODULE_LOAD_ONCE(false); +PA_MODULE_USAGE( + "name=<name for the source, to be prefixed> " + "source_name=<name for the source> " + "source_properties=<properties for the source> " + "namereg_fail=<when false attempt to synthesise new source_name if it is already taken> " + "device=<ALSA device> " + "device_id=<ALSA card index> " + "format=<sample format> " + "rate=<sample rate> " + "alternate_rate=<alternate sample rate> " + "channels=<number of channels> " + "channel_map=<channel map> " + "fragments=<number of fragments> " + "fragment_size=<fragment size> " + "mmap=<enable memory mapping?> " + "tsched=<enable system timer based scheduling mode?> " + "tsched_buffer_size=<buffer size when using timer based scheduling> " + "tsched_buffer_watermark=<upper fill watermark> " + "ignore_dB=<ignore dB information from the device?> " + "control=<name of mixer control, or name and index separated by a comma>" + "deferred_volume=<Synchronize software and hardware volume changes to avoid momentary jumps?> " + "deferred_volume_safety_margin=<usec adjustment depending on volume direction> " + "deferred_volume_extra_delay=<usec adjustment to HW volume changes> " + "fixed_latency_range=<disable latency range changes on overrun?>"); + +static const char* const valid_modargs[] = { + "name", + "source_name", + "source_properties", + "namereg_fail", + "device", + "device_id", + "format", + "rate", + "alternate_rate", + "channels", + "channel_map", + "fragments", + "fragment_size", + "mmap", + "tsched", + "tsched_buffer_size", + "tsched_buffer_watermark", + "ignore_dB", + "control", + "deferred_volume", + "deferred_volume_safety_margin", + "deferred_volume_extra_delay", + "fixed_latency_range", + NULL +}; + +int pa__init(pa_module*m) { + pa_modargs *ma = NULL; + + pa_assert(m); + + pa_alsa_refcnt_inc(); + + if (!(ma = pa_modargs_new(m->argument, valid_modargs))) { + pa_log("Failed to parse module arguments"); + goto fail; + } + + if (!(m->userdata = pa_alsa_source_new(m, ma, __FILE__, NULL, NULL))) + goto fail; + + pa_modargs_free(ma); + + return 0; + +fail: + + if (ma) + pa_modargs_free(ma); + + pa__done(m); + + return -1; +} + +int pa__get_n_used(pa_module *m) { + pa_source *source; + + pa_assert(m); + pa_assert_se(source = m->userdata); + + return pa_source_linked_by(source); +} + +void pa__done(pa_module*m) { + pa_source *source; + + pa_assert(m); + + if ((source = m->userdata)) + pa_alsa_source_free(source); + + pa_alsa_refcnt_dec(); +} |