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mpv/audio/out/ao_opensles.c
Tom Yan d1e9f4a159 ao_opensles: add guards for sample rate to use
Upstream "Wilhelm" (the Android OpenSLES implementation) supports
only 8000 <= rate <= 192000. Make sure mpv resamples the audio
when necessary.
2021-11-19 14:27:52 +01:00

267 lines
7.9 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;
void *buf;
int bytes_per_enqueue;
pthread_mutex_t buffer_lock;
double audio_latency;
int frames_per_enqueue;
int buffer_size_in_ms;
};
#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->buf);
p->buf = NULL;
}
#undef DESTROY
static void buffer_callback(SLBufferQueueItf buffer_queue, void *context)
{
struct ao *ao = context;
struct priv *p = ao->priv;
SLresult res;
double delay;
pthread_mutex_lock(&p->buffer_lock);
delay = p->frames_per_enqueue / (double)ao->samplerate;
delay += p->audio_latency;
ao_read_data(ao, &p->buf, p->frames_per_enqueue,
mp_time_us() + 1000000LL * delay);
res = (*buffer_queue)->Enqueue(buffer_queue, p->buf, p->bytes_per_enqueue);
if (res != SL_RESULT_SUCCESS)
MP_ERR(ao, "Failed to Enqueue: %d\n", res);
pthread_mutex_unlock(&p->buffer_lock);
}
#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;
SLAndroidDataFormat_PCM_EX pcm;
SLDataSource audio_source;
SLDataSink audio_sink;
// This AO only supports two channels at the moment
mp_chmap_from_channels(&ao->channels, 2);
// Upstream "Wilhelm" supports only 8000 <= rate <= 192000
ao->samplerate = MPCLAMP(ao->samplerate, 8000, 192000);
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 = 8;
if (af_fmt_is_int(ao->format)) {
// Be future-proof
if (af_fmt_to_bytes(ao->format) > 2)
ao->format = AF_FORMAT_S32;
else
ao->format = af_fmt_from_planar(ao->format);
pcm.formatType = SL_DATAFORMAT_PCM;
} else {
ao->format = AF_FORMAT_FLOAT;
pcm.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
pcm.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
}
pcm.numChannels = ao->channels.num;
pcm.containerSize = pcm.bitsPerSample = 8 * af_fmt_to_bytes(ao->format);
pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
pcm.sampleRate = ao->samplerate * 1000;
if (p->buffer_size_in_ms) {
ao->device_buffer = ao->samplerate * p->buffer_size_in_ms / 1000;
// As the purpose of buffer_size_in_ms is to request a specific
// soft buffer size:
ao->def_buffer = 0;
}
// But it does not make sense if it is smaller than the enqueue size:
if (p->frames_per_enqueue) {
ao->device_buffer = MPMAX(ao->device_buffer, p->frames_per_enqueue);
} else {
if (ao->device_buffer) {
p->frames_per_enqueue = ao->device_buffer;
} else if (ao->def_buffer) {
p->frames_per_enqueue = ao->def_buffer * ao->samplerate;
} else {
MP_ERR(ao, "Enqueue size is not set and can neither be derived\n");
goto error;
}
}
p->bytes_per_enqueue = p->frames_per_enqueue * ao->channels.num *
af_fmt_to_bytes(ao->format);
p->buf = calloc(1, p->bytes_per_enqueue);
if (!p->buf) {
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;
SLInterfaceID iid_array[] = { SL_IID_BUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION };
SLboolean required[] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE };
CHK((*p->engine)->CreateAudioPlayer(p->engine, &p->player, &audio_source,
&audio_sink, 2, 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));
CHK((*p->play)->SetPlayState(p->play, SL_PLAYSTATE_PLAYING));
SLAndroidConfigurationItf android_config;
SLuint32 audio_latency = 0, value_size = sizeof(SLuint32);
SLint32 get_interface_result = (*p->player)->GetInterface(
p->player,
SL_IID_ANDROIDCONFIGURATION,
&android_config
);
if (get_interface_result == SL_RESULT_SUCCESS) {
SLint32 get_configuration_result = (*android_config)->GetConfiguration(
android_config,
(const SLchar *)"androidGetAudioLatency",
&value_size,
&audio_latency
);
if (get_configuration_result == SL_RESULT_SUCCESS) {
p->audio_latency = (double)audio_latency / 1000.0;
MP_INFO(ao, "Device latency is %f\n", p->audio_latency);
}
}
return 1;
error:
uninit(ao);
return -1;
}
#undef CHK
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
(*p->buffer_queue)->Clear(p->buffer_queue);
}
static void resume(struct ao *ao)
{
struct priv *p = ao->priv;
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,
.start = resume,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.buffer_size_in_ms = 250,
},
.options = (const struct m_option[]) {
{"frames-per-enqueue", OPT_INT(frames_per_enqueue),
M_RANGE(1, 96000)},
{"buffer-size-in-ms", OPT_INT(buffer_size_in_ms),
M_RANGE(0, 500)},
{0}
},
.options_prefix = "opensles",
};