Adding upstream version 0.37.0.upstream/0.37.0

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 401 insertions, 0 deletions

diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
new file mode 100644
index 0000000..7172908
--- /dev/null
+++ b/audio/out/ao_openal.c
@@ -0,0 +1,401 @@
+/*
+ * OpenAL audio output driver for MPlayer
+ *
+ * Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
+ *
+ * This file is part of mpv.
+ *
+ * mpv 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.
+ *
+ * mpv 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 mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <inttypes.h>
+#ifdef OPENAL_AL_H
+#include <OpenAL/alc.h>
+#include <OpenAL/al.h>
+#include <OpenAL/alext.h>
+#else
+#include <AL/alc.h>
+#include <AL/al.h>
+#include <AL/alext.h>
+#endif
+
+#include "common/msg.h"
+
+#include "ao.h"
+#include "internal.h"
+#include "audio/format.h"
+#include "osdep/timer.h"
+#include "options/m_option.h"
+
+#define MAX_CHANS MP_NUM_CHANNELS
+#define MAX_BUF 128
+#define MAX_SAMPLES 32768
+static ALuint buffers[MAX_BUF];
+static ALuint buffer_size[MAX_BUF];
+static ALuint source;
+
+static int cur_buf;
+static int unqueue_buf;
+
+static struct ao *ao_data;
+
+struct priv {
+    ALenum al_format;
+    int num_buffers;
+    int num_samples;
+    bool direct_channels;
+};
+
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    switch (cmd) {
+    case AOCONTROL_GET_VOLUME:
+    case AOCONTROL_SET_VOLUME: {
+        ALfloat volume;
+        float *vol = arg;
+        if (cmd == AOCONTROL_SET_VOLUME) {
+            volume = *vol / 100.0;
+            alListenerf(AL_GAIN, volume);
+        }
+        alGetListenerf(AL_GAIN, &volume);
+        *vol = volume * 100;
+        return CONTROL_TRUE;
+    }
+    case AOCONTROL_GET_MUTE:
+    case AOCONTROL_SET_MUTE: {
+        bool mute = *(bool *)arg;
+
+        // openal has no mute control, only gain.
+        // Thus reverse the muted state to get required gain
+        ALfloat al_mute = (ALfloat)(!mute);
+        if (cmd == AOCONTROL_SET_MUTE) {
+            alSourcef(source, AL_GAIN, al_mute);
+        }
+        alGetSourcef(source, AL_GAIN, &al_mute);
+        *(bool *)arg = !((bool)al_mute);
+        return CONTROL_TRUE;
+    }
+
+    }
+    return CONTROL_UNKNOWN;
+}
+
+static enum af_format get_supported_format(int format)
+{
+    switch (format) {
+    case AF_FORMAT_U8:
+        if (alGetEnumValue((ALchar*)"AL_FORMAT_MONO8"))
+            return AF_FORMAT_U8;
+        break;
+
+    case AF_FORMAT_S16:
+        if (alGetEnumValue((ALchar*)"AL_FORMAT_MONO16"))
+            return AF_FORMAT_S16;
+        break;
+
+    case AF_FORMAT_S32:
+        if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL)
+            return AF_FORMAT_S32;
+        break;
+
+    case AF_FORMAT_FLOAT:
+        if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE)
+            return AF_FORMAT_FLOAT;
+        break;
+    }
+    return AL_FALSE;
+}
+
+static ALenum get_supported_layout(int format, int channels)
+{
+    const char *channel_str[] = {
+        [1] = "MONO",
+        [2] = "STEREO",
+        [4] = "QUAD",
+        [6] = "51CHN",
+        [7] = "61CHN",
+        [8] = "71CHN",
+    };
+    const char *format_str[] = {
+        [AF_FORMAT_U8] = "8",
+        [AF_FORMAT_S16] = "16",
+        [AF_FORMAT_S32] = "32",
+        [AF_FORMAT_FLOAT] = "_FLOAT32",
+    };
+    if (channel_str[channels] == NULL || format_str[format] == NULL)
+        return AL_FALSE;
+
+    char enum_name[32];
+    // AF_FORMAT_FLOAT uses same enum name as AF_FORMAT_S32 for multichannel
+    // playback, while it is different for mono and stereo.
+    // OpenAL Soft does not support AF_FORMAT_S32 and seems to reuse the names.
+    if (channels > 2 && format == AF_FORMAT_FLOAT)
+        format = AF_FORMAT_S32;
+    snprintf(enum_name, sizeof(enum_name), "AL_FORMAT_%s%s", channel_str[channels],
+             format_str[format]);
+
+    if (alGetEnumValue((ALchar*)enum_name)) {
+        return alGetEnumValue((ALchar*)enum_name);
+    }
+    return AL_FALSE;
+}
+
+// close audio device
+static void uninit(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    alSourceStop(source);
+    alSourcei(source, AL_BUFFER, 0);
+
+    alDeleteBuffers(p->num_buffers, buffers);
+    alDeleteSources(1, &source);
+
+    ALCcontext *ctx = alcGetCurrentContext();
+    ALCdevice *dev = alcGetContextsDevice(ctx);
+    alcMakeContextCurrent(NULL);
+    alcDestroyContext(ctx);
+    alcCloseDevice(dev);
+    ao_data = NULL;
+}
+
+static int init(struct ao *ao)
+{
+    float position[3] = {0, 0, 0};
+    float direction[6] = {0, 0, -1, 0, 1, 0};
+    ALCdevice *dev = NULL;
+    ALCcontext *ctx = NULL;
+    ALCint freq = 0;
+    ALCint attribs[] = {ALC_FREQUENCY, ao->samplerate, 0, 0};
+    struct priv *p = ao->priv;
+    if (ao_data) {
+        MP_FATAL(ao, "Not reentrant!\n");
+        return -1;
+    }
+    ao_data = ao;
+    char *dev_name = ao->device;
+    dev = alcOpenDevice(dev_name && dev_name[0] ? dev_name : NULL);
+    if (!dev) {
+        MP_FATAL(ao, "could not open device\n");
+        goto err_out;
+    }
+    ctx = alcCreateContext(dev, attribs);
+    alcMakeContextCurrent(ctx);
+    alListenerfv(AL_POSITION, position);
+    alListenerfv(AL_ORIENTATION, direction);
+
+    alGenSources(1, &source);
+    if (p->direct_channels) {
+        if (alIsExtensionPresent("AL_SOFT_direct_channels_remix")) {
+            alSourcei(source,
+                alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT"),
+                alcGetEnumValue(dev, "AL_REMIX_UNMATCHED_SOFT"));
+        } else {
+            MP_WARN(ao, "Direct channels aren't supported by this version of OpenAL\n");
+        }
+    }
+
+    cur_buf = 0;
+    unqueue_buf = 0;
+    for (int i = 0; i < p->num_buffers; ++i) {
+        buffer_size[i] = 0;
+    }
+
+    alGenBuffers(p->num_buffers, buffers);
+
+    alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
+    if (alcGetError(dev) == ALC_NO_ERROR && freq)
+        ao->samplerate = freq;
+
+    // Check sample format
+    int try_formats[AF_FORMAT_COUNT + 1];
+    enum af_format sample_format = 0;
+    af_get_best_sample_formats(ao->format, try_formats);
+    for (int n = 0; try_formats[n]; n++) {
+        sample_format = get_supported_format(try_formats[n]);
+        if (sample_format != AF_FORMAT_UNKNOWN) {
+            ao->format = try_formats[n];
+            break;
+        }
+    }
+
+    if (sample_format == AF_FORMAT_UNKNOWN) {
+        MP_FATAL(ao, "Can't find appropriate sample format.\n");
+        uninit(ao);
+        goto err_out;
+    }
+
+    // Check if OpenAL driver supports the desired number of channels.
+    int num_channels = ao->channels.num;
+    do {
+        p->al_format = get_supported_layout(sample_format, num_channels);
+        if (p->al_format == AL_FALSE) {
+            num_channels = num_channels - 1;
+        }
+    } while (p->al_format == AL_FALSE && num_channels > 1);
+
+    // Request number of speakers for output from ao.
+    const struct mp_chmap possible_layouts[] = {
+        {0},                                        // empty
+        MP_CHMAP_INIT_MONO,                         // mono
+        MP_CHMAP_INIT_STEREO,                       // stereo
+        {0},                                        // 2.1
+        MP_CHMAP4(FL, FR, BL, BR),                  // 4.0
+        {0},                                        // 5.0
+        MP_CHMAP6(FL, FR, FC, LFE, BL, BR),         // 5.1
+        MP_CHMAP7(FL, FR, FC, LFE, SL, SR, BC),     // 6.1
+        MP_CHMAP8(FL, FR, FC, LFE, BL, BR, SL, SR), // 7.1
+    };
+    ao->channels = possible_layouts[num_channels];
+    if (!ao->channels.num)
+        mp_chmap_set_unknown(&ao->channels, num_channels);
+
+    if (p->al_format == AL_FALSE || !mp_chmap_is_valid(&ao->channels)) {
+        MP_FATAL(ao, "Can't find appropriate channel layout.\n");
+        uninit(ao);
+        goto err_out;
+    }
+
+    ao->device_buffer = p->num_buffers * p->num_samples;
+    return 0;
+
+err_out:
+    ao_data = NULL;
+    return -1;
+}
+
+static void unqueue_buffers(struct ao *ao)
+{
+    struct priv *q = ao->priv;
+    ALint p;
+    int till_wrap = q->num_buffers - unqueue_buf;
+    alGetSourcei(source, AL_BUFFERS_PROCESSED, &p);
+    if (p >= till_wrap) {
+        alSourceUnqueueBuffers(source, till_wrap, &buffers[unqueue_buf]);
+        unqueue_buf = 0;
+        p -= till_wrap;
+    }
+    if (p) {
+        alSourceUnqueueBuffers(source, p, &buffers[unqueue_buf]);
+        unqueue_buf += p;
+    }
+}
+
+static void reset(struct ao *ao)
+{
+    alSourceStop(source);
+    unqueue_buffers(ao);
+}
+
+static bool audio_set_pause(struct ao *ao, bool pause)
+{
+    if (pause) {
+        alSourcePause(source);
+    } else {
+        alSourcePlay(source);
+    }
+    return true;
+}
+
+static bool audio_write(struct ao *ao, void **data, int samples)
+{
+    struct priv *p = ao->priv;
+
+    int num = (samples + p->num_samples - 1) / p->num_samples;
+
+    for (int i = 0; i < num; i++) {
+        char *d = *data;
+        buffer_size[cur_buf] =
+            MPMIN(samples - i * p->num_samples, p->num_samples);
+        d += i * buffer_size[cur_buf] * ao->sstride;
+        alBufferData(buffers[cur_buf], p->al_format, d,
+            buffer_size[cur_buf] * ao->sstride, ao->samplerate);
+        alSourceQueueBuffers(source, 1, &buffers[cur_buf]);
+        cur_buf = (cur_buf + 1) % p->num_buffers;
+    }
+
+    return true;
+}
+
+static void audio_start(struct ao *ao)
+{
+    alSourcePlay(source);
+}
+
+static void get_state(struct ao *ao, struct mp_pcm_state *state)
+{
+    struct priv *p = ao->priv;
+
+    ALint queued;
+    unqueue_buffers(ao);
+    alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
+
+    double source_offset = 0;
+    if(alIsExtensionPresent("AL_SOFT_source_latency")) {
+        ALdouble offsets[2];
+        LPALGETSOURCEDVSOFT alGetSourcedvSOFT = alGetProcAddress("alGetSourcedvSOFT");
+        alGetSourcedvSOFT(source, AL_SEC_OFFSET_LATENCY_SOFT, offsets);
+        // Additional latency to the play buffer, the remaining seconds to be
+        // played minus the offset (seconds already played)
+        source_offset = offsets[1] - offsets[0];
+    } else {
+        float offset = 0;
+        alGetSourcef(source, AL_SEC_OFFSET, &offset);
+        source_offset = -offset;
+    }
+
+    int queued_samples = 0;
+    for (int i = 0, index = cur_buf; i < queued; ++i) {
+        queued_samples += buffer_size[index];
+        index = (index + 1) % p->num_buffers;
+    }
+
+    state->delay = queued_samples / (double)ao->samplerate + source_offset;
+
+    state->queued_samples = queued_samples;
+    state->free_samples = MPMAX(p->num_buffers - queued, 0) * p->num_samples;
+
+    ALint source_state = 0;
+    alGetSourcei(source, AL_SOURCE_STATE, &source_state);
+    state->playing = source_state == AL_PLAYING;
+}
+
+#define OPT_BASE_STRUCT struct priv
+
+const struct ao_driver audio_out_openal = {
+    .description = "OpenAL audio output",
+    .name      = "openal",
+    .init      = init,
+    .uninit    = uninit,
+    .control   = control,
+    .get_state = get_state,
+    .write     = audio_write,
+    .start     = audio_start,
+    .set_pause = audio_set_pause,
+    .reset     = reset,
+    .priv_size = sizeof(struct priv),
+    .priv_defaults = &(const struct priv) {
+        .num_buffers = 4,
+        .num_samples = 8192,
+        .direct_channels = true,
+    },
+    .options = (const struct m_option[]) {
+        {"num-buffers", OPT_INT(num_buffers), M_RANGE(2, MAX_BUF)},
+        {"num-samples", OPT_INT(num_samples), M_RANGE(256, MAX_SAMPLES)},
+        {"direct-channels", OPT_BOOL(direct_channels)},
+        {0}
+    },
+    .options_prefix = "openal",
+};