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Diffstat (limited to 'audio/out/ao_opensles.c')
-rw-r--r--audio/out/ao_opensles.c98
1 files changed, 52 insertions, 46 deletions
diff --git a/audio/out/ao_opensles.c b/audio/out/ao_opensles.c
index ea48de892e..8c24320406 100644
--- a/audio/out/ao_opensles.c
+++ b/audio/out/ao_opensles.c
@@ -35,18 +35,13 @@ struct priv {
SLBufferQueueItf buffer_queue;
SLEngineItf engine;
SLPlayItf play;
- char *buf;
- size_t buffer_size;
+ void *buf;
+ int bytes_per_enqueue;
pthread_mutex_t buffer_lock;
double audio_latency;
- int cfg_frames_per_buffer;
-};
-
-static const int fmtmap[][2] = {
- { AF_FORMAT_U8, SL_PCMSAMPLEFORMAT_FIXED_8 },
- { AF_FORMAT_S16, SL_PCMSAMPLEFORMAT_FIXED_16 },
- { 0 }
+ int frames_per_enqueue;
+ int buffer_size_in_ms;
};
#define DESTROY(thing) \
@@ -71,7 +66,6 @@ static void uninit(struct ao *ao)
free(p->buf);
p->buf = NULL;
- p->buffer_size = 0;
}
#undef DESTROY
@@ -81,26 +75,22 @@ 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->buf;
- delay = 2 * p->buffer_size / (double)ao->bps;
+ delay = 2 * p->frames_per_enqueue / (double)ao->samplerate;
delay += p->audio_latency;
- ao_read_data(ao, data, p->buffer_size / ao->sstride,
+ ao_read_data(ao, &p->buf, p->frames_per_enqueue,
mp_time_us() + 1000000LL * delay);
- res = (*buffer_queue)->Enqueue(buffer_queue, p->buf, p->buffer_size);
+ 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 DEFAULT_BUFFER_SIZE_MS 250
-
#define CHK(stmt) \
{ \
SLresult res = stmt; \
@@ -115,7 +105,7 @@ static int init(struct ao *ao)
struct priv *p = ao->priv;
SLDataLocator_BufferQueue locator_buffer_queue;
SLDataLocator_OutputMix locator_output_mix;
- SLDataFormat_PCM pcm;
+ SLAndroidDataFormat_PCM_EX pcm;
SLDataSource audio_source;
SLDataSink audio_sink;
@@ -129,43 +119,55 @@ static int init(struct ao *ao)
CHK((*p->output_mix)->Realize(p->output_mix, SL_BOOLEAN_FALSE));
locator_buffer_queue.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
- locator_buffer_queue.numBuffers = 1;
-
- pcm.formatType = SL_DATAFORMAT_PCM;
- pcm.numChannels = 2;
-
- int compatible_formats[AF_FORMAT_COUNT + 1];
- 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;
+ 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.containerSize = 8 * af_fmt_to_bytes(ao->format);
+ 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;
+ }
- // samplesPerSec is misnamed, actually it's samples per ms
- pcm.samplesPerSec = ao->samplerate * 1000;
+ // 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;
+ }
+ }
- 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 *
+ p->bytes_per_enqueue = p->frames_per_enqueue * ao->channels.num *
af_fmt_to_bytes(ao->format);
- p->buf = calloc(1, p->buffer_size);
+ 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);
@@ -248,8 +250,12 @@ const struct ao_driver audio_out_opensles = {
.resume = resume,
.priv_size = sizeof(struct priv),
+ .priv_defaults = &(const struct priv) {
+ .buffer_size_in_ms = 250,
+ },
.options = (const struct m_option[]) {
- OPT_INTRANGE("frames-per-buffer", cfg_frames_per_buffer, 0, 1, 96000),
+ OPT_INTRANGE("frames-per-enqueue", frames_per_enqueue, 0, 1, 96000),
+ OPT_INTRANGE("buffer-size-in-ms", buffer_size_in_ms, 0, 0, 500),
{0}
},
.options_prefix = "opensles",