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-rw-r--r--audio/out/ao_openal.c401
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diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
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+++ b/audio/out/ao_openal.c
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+/*
+ * 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",
+};
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