mpv/audio/out/ao_opensles.c

251 lines
7.2 KiB
C

/*
* OpenSL ES audio output driver.
* Copyright (C) 2016 Ilya Zhuravlev <whatever@xyz.is>
*
* 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 "ao.h"
#include "internal.h"
#include "common/msg.h"
#include "audio/format.h"
#include "options/m_option.h"
#include "osdep/timer.h"
#include <SLES/OpenSLES.h>
#include <SLES/OpenSLES_Android.h>
#include <pthread.h>
struct priv {
SLObjectItf sl, output_mix, player;
SLBufferQueueItf buffer_queue;
SLEngineItf engine;
SLPlayItf play;
char *curbuf, *buf1, *buf2;
size_t buffer_size;
pthread_mutex_t buffer_lock;
int cfg_frames_per_buffer;
int cfg_sample_rate;
};
static const int fmtmap[][2] = {
{ AF_FORMAT_U8, SL_PCMSAMPLEFORMAT_FIXED_8 },
{ AF_FORMAT_S16, SL_PCMSAMPLEFORMAT_FIXED_16 },
{ AF_FORMAT_S32, SL_PCMSAMPLEFORMAT_FIXED_32 },
{ 0 }
};
#define DESTROY(thing) \
if (p->thing) { \
(*p->thing)->Destroy(p->thing); \
p->thing = NULL; \
}
static void uninit(struct ao *ao)
{
struct priv *p = ao->priv;
DESTROY(player);
DESTROY(output_mix);
DESTROY(sl);
p->buffer_queue = NULL;
p->engine = NULL;
p->play = NULL;
pthread_mutex_destroy(&p->buffer_lock);
free(p->buf1);
free(p->buf2);
p->curbuf = p->buf1 = p->buf2 = NULL;
p->buffer_size = 0;
}
#undef DESTROY
static void buffer_callback(SLBufferQueueItf buffer_queue, void *context)
{
struct ao *ao = context;
struct priv *p = ao->priv;
SLresult res;
void *data[1];
double delay;
pthread_mutex_lock(&p->buffer_lock);
data[0] = p->curbuf;
delay = 2 * p->buffer_size / (double)ao->bps;
ao_read_data(ao, data, p->buffer_size / ao->sstride,
mp_time_us() + 1000000LL * delay);
res = (*buffer_queue)->Enqueue(buffer_queue, p->curbuf, p->buffer_size);
if (res != SL_RESULT_SUCCESS)
MP_ERR(ao, "Failed to Enqueue: %d\n", res);
else
p->curbuf = (p->curbuf == p->buf1) ? p->buf2 : p->buf1;
pthread_mutex_unlock(&p->buffer_lock);
}
#define DEFAULT_BUFFER_SIZE_MS 50
#define CHK(stmt) \
{ \
SLresult res = stmt; \
if (res != SL_RESULT_SUCCESS) { \
MP_ERR(ao, "%s: %d\n", #stmt, res); \
goto error; \
} \
}
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
SLDataLocator_BufferQueue locator_buffer_queue;
SLDataLocator_OutputMix locator_output_mix;
SLDataFormat_PCM pcm;
SLDataSource audio_source;
SLDataSink audio_sink;
// This AO only supports two channels at the moment
mp_chmap_from_channels(&ao->channels, 2);
CHK(slCreateEngine(&p->sl, 0, NULL, 0, NULL, NULL));
CHK((*p->sl)->Realize(p->sl, SL_BOOLEAN_FALSE));
CHK((*p->sl)->GetInterface(p->sl, SL_IID_ENGINE, (void*)&p->engine));
CHK((*p->engine)->CreateOutputMix(p->engine, &p->output_mix, 0, NULL, NULL));
CHK((*p->output_mix)->Realize(p->output_mix, SL_BOOLEAN_FALSE));
locator_buffer_queue.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
locator_buffer_queue.numBuffers = 2;
pcm.formatType = SL_DATAFORMAT_PCM;
pcm.numChannels = 2;
int compatible_formats[AF_FORMAT_COUNT];
af_get_best_sample_formats(ao->format, compatible_formats);
pcm.bitsPerSample = 0;
for (int i = 0; compatible_formats[i] && !pcm.bitsPerSample; ++i)
for (int j = 0; fmtmap[j][0]; ++j)
if (compatible_formats[i] == fmtmap[j][0]) {
ao->format = fmtmap[j][0];
pcm.bitsPerSample = fmtmap[j][1];
break;
}
if (!pcm.bitsPerSample) {
MP_ERR(ao, "Cannot find compatible audio format\n");
goto error;
}
pcm.containerSize = 8 * af_fmt_to_bytes(ao->format);
pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
if (p->cfg_sample_rate)
ao->samplerate = p->cfg_sample_rate;
// samplesPerSec is misnamed, actually it's samples per ms
pcm.samplesPerSec = ao->samplerate * 1000;
if (p->cfg_frames_per_buffer)
ao->device_buffer = p->cfg_frames_per_buffer;
else
ao->device_buffer = ao->samplerate * DEFAULT_BUFFER_SIZE_MS / 1000;
p->buffer_size = ao->device_buffer * ao->channels.num *
af_fmt_to_bytes(ao->format);
p->buf1 = calloc(1, p->buffer_size);
p->buf2 = calloc(1, p->buffer_size);
p->curbuf = p->buf1;
if (!p->buf1 || !p->buf2) {
MP_ERR(ao, "Failed to allocate device buffer\n");
goto error;
}
int r = pthread_mutex_init(&p->buffer_lock, NULL);
if (r) {
MP_ERR(ao, "Failed to initialize the mutex: %d\n", r);
goto error;
}
audio_source.pFormat = (void*)&pcm;
audio_source.pLocator = (void*)&locator_buffer_queue;
locator_output_mix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
locator_output_mix.outputMix = p->output_mix;
audio_sink.pLocator = (void*)&locator_output_mix;
audio_sink.pFormat = NULL;
SLboolean required[] = { SL_BOOLEAN_TRUE };
SLInterfaceID iid_array[] = { SL_IID_BUFFERQUEUE };
CHK((*p->engine)->CreateAudioPlayer(p->engine, &p->player, &audio_source,
&audio_sink, 1, iid_array, required));
CHK((*p->player)->Realize(p->player, SL_BOOLEAN_FALSE));
CHK((*p->player)->GetInterface(p->player, SL_IID_PLAY, (void*)&p->play));
CHK((*p->player)->GetInterface(p->player, SL_IID_BUFFERQUEUE,
(void*)&p->buffer_queue));
CHK((*p->buffer_queue)->RegisterCallback(p->buffer_queue,
buffer_callback, ao));
return 1;
error:
uninit(ao);
return -1;
}
#undef CHK
static void set_play_state(struct ao *ao, SLuint32 state)
{
struct priv *p = ao->priv;
SLresult res = (*p->play)->SetPlayState(p->play, state);
if (res != SL_RESULT_SUCCESS)
MP_ERR(ao, "Failed to SetPlayState(%d): %d\n", state, res);
}
static void reset(struct ao *ao)
{
set_play_state(ao, SL_PLAYSTATE_STOPPED);
}
static void resume(struct ao *ao)
{
struct priv *p = ao->priv;
set_play_state(ao, SL_PLAYSTATE_PLAYING);
// enqueue two buffers
buffer_callback(p->buffer_queue, ao);
buffer_callback(p->buffer_queue, ao);
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_opensles = {
.description = "OpenSL ES audio output",
.name = "opensles",
.init = init,
.uninit = uninit,
.reset = reset,
.resume = resume,
.priv_size = sizeof(struct priv),
.options = (const struct m_option[]) {
OPT_INTRANGE("frames-per-buffer", cfg_frames_per_buffer, 0, 1, 10000),
OPT_INTRANGE("sample-rate", cfg_sample_rate, 0, 1000, 100000),
{0}
},
};