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gstreamer/sys/opensles/openslesringbuffer.c
Sebastian Dröge 0acdcc1b37 openslesringbuffer: Warn if the position reported by OpenSL is higher than what we queued up so far 
This would hint at wrong position reporting, and apparently sometimes happens
after a seek.
2016-03-16 12:03:53 +02:00

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/* GStreamer
* Copyright (C) 2012 Fluendo S.A. <support@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <string.h>
#include "opensles.h"
#include "openslesringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (opensles_ringbuffer_debug);
#define GST_CAT_DEFAULT opensles_ringbuffer_debug
#define _do_init \
GST_DEBUG_CATEGORY_INIT (opensles_ringbuffer_debug, \
"opensles_ringbuffer", 0, "OpenSL ES ringbuffer");
#define parent_class gst_opensles_ringbuffer_parent_class
G_DEFINE_TYPE_WITH_CODE (GstOpenSLESRingBuffer, gst_opensles_ringbuffer,
GST_TYPE_AUDIO_RING_BUFFER, _do_init);
/*
* Some generic helper functions
*/
static inline SLuint32
_opensles_sample_rate (guint rate)
{
switch (rate) {
case 8000:
return SL_SAMPLINGRATE_8;
case 11025:
return SL_SAMPLINGRATE_11_025;
case 12000:
return SL_SAMPLINGRATE_12;
case 16000:
return SL_SAMPLINGRATE_16;
case 22050:
return SL_SAMPLINGRATE_22_05;
case 24000:
return SL_SAMPLINGRATE_24;
case 32000:
return SL_SAMPLINGRATE_32;
case 44100:
return SL_SAMPLINGRATE_44_1;
case 48000:
return SL_SAMPLINGRATE_48;
case 64000:
return SL_SAMPLINGRATE_64;
case 88200:
return SL_SAMPLINGRATE_88_2;
case 96000:
return SL_SAMPLINGRATE_96;
case 192000:
return SL_SAMPLINGRATE_192;
default:
return 0;
}
}
static inline SLuint32
_opensles_channel_mask (GstAudioRingBufferSpec * spec)
{
switch (spec->info.channels) {
case 1:
return (SL_SPEAKER_FRONT_CENTER);
case 2:
return (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT);
default:
return 0;
}
}
static inline void
_opensles_format (GstAudioRingBufferSpec * spec, SLDataFormat_PCM * format)
{
format->formatType = SL_DATAFORMAT_PCM;
format->numChannels = spec->info.channels;
format->samplesPerSec = _opensles_sample_rate (spec->info.rate);
format->bitsPerSample = spec->info.finfo->depth;
format->containerSize = spec->info.finfo->width;
format->channelMask = _opensles_channel_mask (spec);
format->endianness =
((spec->info.finfo->endianness ==
G_BIG_ENDIAN) ? SL_BYTEORDER_BIGENDIAN : SL_BYTEORDER_LITTLEENDIAN);
}
/*
* Recorder related functions
*/
static gboolean
_opensles_recorder_acquire (GstAudioRingBuffer * rb,
GstAudioRingBufferSpec * spec)
{
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
SLDataFormat_PCM format;
SLAndroidConfigurationItf config;
/* Configure audio source */
SLDataLocator_IODevice loc_dev = {
SL_DATALOCATOR_IODEVICE, SL_IODEVICE_AUDIOINPUT,
SL_DEFAULTDEVICEID_AUDIOINPUT, NULL
};
SLDataSource audioSrc = { &loc_dev, NULL };
/* Configure audio sink */
SLDataLocator_AndroidSimpleBufferQueue loc_bq = {
SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2
};
SLDataSink audioSink = { &loc_bq, &format };
/* Required optional interfaces */
const SLInterfaceID ids[2] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
SL_IID_ANDROIDCONFIGURATION
};
const SLboolean req[2] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE };
/* Define the audio format in OpenSL ES terminology */
_opensles_format (spec, &format);
/* Create the audio recorder object (requires the RECORD_AUDIO permission) */
result = (*thiz->engineEngine)->CreateAudioRecorder (thiz->engineEngine,
&thiz->recorderObject, &audioSrc, &audioSink, 2, ids, req);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "engine.CreateAudioRecorder failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Set the recording preset if we have one */
if (thiz->preset != GST_OPENSLES_RECORDING_PRESET_NONE) {
SLint32 preset = gst_to_opensles_recording_preset (thiz->preset);
result = (*thiz->recorderObject)->GetInterface (thiz->recorderObject,
SL_IID_ANDROIDCONFIGURATION, &config);
if (result == SL_RESULT_SUCCESS) {
result = (*config)->SetConfiguration (config,
SL_ANDROID_KEY_RECORDING_PRESET, &preset, sizeof (preset));
if (result != SL_RESULT_SUCCESS) {
GST_WARNING_OBJECT (thiz, "Failed to set playback stream type (0x%08x)",
(guint32) result);
}
} else {
GST_WARNING_OBJECT (thiz,
"Could not get configuration interface 0x%08x", (guint32) result);
}
}
/* Realize the audio recorder object */
result =
(*thiz->recorderObject)->Realize (thiz->recorderObject, SL_BOOLEAN_FALSE);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "recorder.Realize failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Get the record interface */
result = (*thiz->recorderObject)->GetInterface (thiz->recorderObject,
SL_IID_RECORD, &thiz->recorderRecord);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "recorder.GetInterface(Record) failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Get the buffer queue interface */
result =
(*thiz->recorderObject)->GetInterface (thiz->recorderObject,
SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &thiz->bufferQueue);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "recorder.GetInterface(BufferQueue) failed(0x%08x)",
(guint32) result);
goto failed;
}
return TRUE;
failed:
return FALSE;
}
/* This callback function is executed when the ringbuffer is started to preroll
* the output buffer queue with empty buffers, from app thread, and each time
* there's a filled buffer, from audio device processing thread,
* the callback behaviour.
*/
static void
_opensles_recorder_cb (SLAndroidSimpleBufferQueueItf bufferQueue, void *context)
{
GstAudioRingBuffer *rb = GST_AUDIO_RING_BUFFER_CAST (context);
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
guint8 *ptr;
gint seg;
gint len;
/* Advance only when we are called by the callback function */
if (bufferQueue) {
gst_audio_ring_buffer_advance (rb, 1);
}
/* Get a segment form the GStreamer ringbuffer to write in */
if (!gst_audio_ring_buffer_prepare_read (rb, &seg, &ptr, &len)) {
GST_WARNING_OBJECT (rb, "No segment available");
return;
}
GST_LOG_OBJECT (thiz, "enqueue: %p size %d segment: %d", ptr, len, seg);
/* Enqueue the sefment as buffer to be written */
result = (*thiz->bufferQueue)->Enqueue (thiz->bufferQueue, ptr, len);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.Enqueue failed(0x%08x)",
(guint32) result);
return;
}
}
static gboolean
_opensles_recorder_start (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
/* Register callback on the buffer queue */
if (!thiz->is_queue_callback_registered) {
result = (*thiz->bufferQueue)->RegisterCallback (thiz->bufferQueue,
_opensles_recorder_cb, rb);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.RegisterCallback failed(0x%08x)",
(guint32) result);
return FALSE;
}
thiz->is_queue_callback_registered = TRUE;
}
/* Preroll one buffer */
_opensles_recorder_cb (NULL, rb);
/* Start recording */
result =
(*thiz->recorderRecord)->SetRecordState (thiz->recorderRecord,
SL_RECORDSTATE_RECORDING);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "recorder.SetRecordState failed(0x%08x)",
(guint32) result);
return FALSE;
}
return TRUE;
}
static gboolean
_opensles_recorder_stop (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
/* Stop recording */
result =
(*thiz->recorderRecord)->SetRecordState (thiz->recorderRecord,
SL_RECORDSTATE_STOPPED);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "recorder.SetRecordState failed(0x%08x)",
(guint32) result);
return FALSE;
}
/* Unregister callback on the buffer queue */
result = (*thiz->bufferQueue)->RegisterCallback (thiz->bufferQueue,
NULL, NULL);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.RegisterCallback failed(0x%08x)",
(guint32) result);
return FALSE;
}
thiz->is_queue_callback_registered = FALSE;
/* Reset the queue */
result = (*thiz->bufferQueue)->Clear (thiz->bufferQueue);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.Clear failed(0x%08x)",
(guint32) result);
return FALSE;
}
return TRUE;
}
/*
* Player related functions
*/
static gboolean
_opensles_player_change_volume (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
SLresult result;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
if (thiz->playerVolume) {
gint millibel = (1.0 - thiz->volume) * -5000.0;
result =
(*thiz->playerVolume)->SetVolumeLevel (thiz->playerVolume, millibel);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.SetVolumeLevel failed(0x%08x)",
(guint32) result);
return FALSE;
}
GST_DEBUG_OBJECT (thiz, "changed volume to %d", millibel);
}
return TRUE;
}
static gboolean
_opensles_player_change_mute (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
SLresult result;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
if (thiz->playerVolume) {
result = (*thiz->playerVolume)->SetMute (thiz->playerVolume, thiz->mute);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.SetMute failed(0x%08x)",
(guint32) result);
return FALSE;
}
GST_DEBUG_OBJECT (thiz, "changed mute to %d", thiz->mute);
}
return TRUE;
}
/* This is a callback function invoked by the playback device thread and
* it's used to monitor position changes */
static void
_opensles_player_event_cb (SLPlayItf caller, void *context, SLuint32 event)
{
if (event & SL_PLAYEVENT_HEADATNEWPOS) {
SLmillisecond position;
(*caller)->GetPosition (caller, &position);
GST_LOG_OBJECT (context, "at position=%u ms", (guint) position);
}
}
static gboolean
_opensles_player_acquire (GstAudioRingBuffer * rb,
GstAudioRingBufferSpec * spec)
{
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
SLDataFormat_PCM format;
SLAndroidConfigurationItf config;
/* Configure audio source
* 4 buffers is the "typical" size as optimized inside Android's
* OpenSL ES, see frameworks/wilhelm/src/itfstruct.h BUFFER_HEADER_TYPICAL
*
* Also only use half of our segment size to make sure that there's always
* some more queued up in our ringbuffer and we don't start to read silence.
*/
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, MIN (4, MAX (spec->segtotal >> 1,
1))
};
SLDataSource audioSrc = { &loc_bufq, &format };
/* Configure audio sink */
SLDataLocator_OutputMix loc_outmix = {
SL_DATALOCATOR_OUTPUTMIX, thiz->outputMixObject
};
SLDataSink audioSink = { &loc_outmix, NULL };
/* Define the required interfaces */
const SLInterfaceID ids[3] = { SL_IID_BUFFERQUEUE, SL_IID_VOLUME,
SL_IID_ANDROIDCONFIGURATION
};
const SLboolean req[3] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
SL_BOOLEAN_FALSE
};
/* Define the format in OpenSL ES terminology */
_opensles_format (spec, &format);
/* Create the player object */
result = (*thiz->engineEngine)->CreateAudioPlayer (thiz->engineEngine,
&thiz->playerObject, &audioSrc, &audioSink, 3, ids, req);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "engine.CreateAudioPlayer failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Set the stream type if we have one */
if (thiz->stream_type != GST_OPENSLES_STREAM_TYPE_NONE) {
SLint32 stream_type = gst_to_opensles_stream_type (thiz->stream_type);
result = (*thiz->playerObject)->GetInterface (thiz->playerObject,
SL_IID_ANDROIDCONFIGURATION, &config);
if (result == SL_RESULT_SUCCESS) {
result = (*config)->SetConfiguration (config,
SL_ANDROID_KEY_STREAM_TYPE, &stream_type, sizeof (stream_type));
if (result != SL_RESULT_SUCCESS) {
GST_WARNING_OBJECT (thiz, "Failed to set playback stream type (0x%08x)",
(guint32) result);
}
} else {
GST_WARNING_OBJECT (thiz,
"Could not get configuration interface 0x%08x", (guint32) result);
}
}
/* Realize the player object */
result =
(*thiz->playerObject)->Realize (thiz->playerObject, SL_BOOLEAN_FALSE);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.Realize failed(0x%08x)", (guint32) result);
goto failed;
}
/* Get the play interface */
result = (*thiz->playerObject)->GetInterface (thiz->playerObject,
SL_IID_PLAY, &thiz->playerPlay);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.GetInterface(Play) failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Get the buffer queue interface */
result = (*thiz->playerObject)->GetInterface (thiz->playerObject,
SL_IID_BUFFERQUEUE, &thiz->bufferQueue);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.GetInterface(BufferQueue) failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Get the volume interface */
result = (*thiz->playerObject)->GetInterface (thiz->playerObject,
SL_IID_VOLUME, &thiz->playerVolume);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.GetInterface(Volume) failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Request position update events at each 20 ms */
result = (*thiz->playerPlay)->SetPositionUpdatePeriod (thiz->playerPlay, 20);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.SetPositionUpdatePeriod failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Define the event mask to be monitorized */
result = (*thiz->playerPlay)->SetCallbackEventsMask (thiz->playerPlay,
SL_PLAYEVENT_HEADATNEWPOS);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.SetCallbackEventsMask failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Register a callback to process the events */
result = (*thiz->playerPlay)->RegisterCallback (thiz->playerPlay,
_opensles_player_event_cb, thiz);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.RegisterCallback(event_cb) failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Configure the volume and mute state */
_opensles_player_change_volume (rb);
_opensles_player_change_mute (rb);
/* Allocate the queue associated ringbuffer memory */
thiz->data_segtotal = loc_bufq.numBuffers;
thiz->data_size = spec->segsize * thiz->data_segtotal;
thiz->data = g_malloc0 (thiz->data_size);
g_atomic_int_set (&thiz->segqueued, 0);
g_atomic_int_set (&thiz->is_prerolled, 0);
thiz->cursor = 0;
return TRUE;
failed:
return FALSE;
}
/* This callback function is executed when the ringbuffer is started to preroll
* the input buffer queue with few buffers, from app thread, and each time
* that rendering of one buffer finishes, from audio device processing thread,
* the callback behaviour.
*
* We wrap the queue behaviour with an appropriate chunk of memory (queue len *
* ringbuffer segment size) which is used to hold the audio data while it's
* being processed in the queue. The memory region is used whit a ringbuffer
* behaviour.
*/
static void
_opensles_player_cb (SLAndroidSimpleBufferQueueItf bufferQueue, void *context)
{
GstAudioRingBuffer *rb = GST_AUDIO_RING_BUFFER_CAST (context);
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
guint8 *ptr, *cur;
gint seg;
gint len;
/* Get a segment form the GStreamer ringbuffer to read some samples */
if (!gst_audio_ring_buffer_prepare_read (rb, &seg, &ptr, &len)) {
GST_WARNING_OBJECT (rb, "No segment available");
return;
}
/* copy the segment data to our queue associated ringbuffer memory */
cur = thiz->data + (thiz->cursor * rb->spec.segsize);
memcpy (cur, ptr, len);
g_atomic_int_inc (&thiz->segqueued);
GST_LOG_OBJECT (thiz, "enqueue: %p size %d segment: %d in queue[%d]",
cur, len, seg, thiz->cursor);
/* advance the cursor in our queue associated ringbuffer */
thiz->cursor = (thiz->cursor + 1) % thiz->data_segtotal;
/* Enqueue the buffer to be rendered */
result = (*thiz->bufferQueue)->Enqueue (thiz->bufferQueue, cur, len);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.Enqueue failed(0x%08x)",
(guint32) result);
return;
}
/* Fill with silence samples the segment of the GStreamer ringbuffer */
gst_audio_ring_buffer_clear (rb, seg);
/* Make the segment reusable */
gst_audio_ring_buffer_advance (rb, 1);
}
static gboolean
_opensles_player_start (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
/* Register callback on the buffer queue */
if (!thiz->is_queue_callback_registered) {
result = (*thiz->bufferQueue)->RegisterCallback (thiz->bufferQueue,
_opensles_player_cb, rb);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.RegisterCallback failed(0x%08x)",
(guint32) result);
return FALSE;
}
thiz->is_queue_callback_registered = TRUE;
}
/* Fill the queue by enqueing a buffer */
if (!g_atomic_int_get (&thiz->is_prerolled)) {
_opensles_player_cb (NULL, rb);
g_atomic_int_set (&thiz->is_prerolled, 1);
}
/* Change player state into PLAYING */
result =
(*thiz->playerPlay)->SetPlayState (thiz->playerPlay,
SL_PLAYSTATE_PLAYING);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.SetPlayState failed(0x%08x)",
(guint32) result);
return FALSE;
}
return TRUE;
}
static gboolean
_opensles_player_pause (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
result =
(*thiz->playerPlay)->SetPlayState (thiz->playerPlay, SL_PLAYSTATE_PAUSED);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.SetPlayState failed(0x%08x)",
(guint32) result);
return FALSE;
}
return TRUE;
}
static gboolean
_opensles_player_stop (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
SLresult result;
/* Change player state into STOPPED */
result =
(*thiz->playerPlay)->SetPlayState (thiz->playerPlay,
SL_PLAYSTATE_STOPPED);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "player.SetPlayState failed(0x%08x)",
(guint32) result);
return FALSE;
}
/* Unregister callback on the buffer queue */
result = (*thiz->bufferQueue)->RegisterCallback (thiz->bufferQueue,
NULL, NULL);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.RegisterCallback failed(0x%08x)",
(guint32) result);
return FALSE;
}
thiz->is_queue_callback_registered = FALSE;
/* Reset the queue */
result = (*thiz->bufferQueue)->Clear (thiz->bufferQueue);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "bufferQueue.Clear failed(0x%08x)",
(guint32) result);
return FALSE;
}
/* Reset our state */
g_atomic_int_set (&thiz->segqueued, 0);
thiz->cursor = 0;
return TRUE;
}
/*
* OpenSL ES ringbuffer wrapper
*/
GstAudioRingBuffer *
gst_opensles_ringbuffer_new (RingBufferMode mode)
{
GstOpenSLESRingBuffer *thiz;
g_return_val_if_fail (mode > RB_MODE_NONE && mode < RB_MODE_LAST, NULL);
thiz = g_object_new (GST_TYPE_OPENSLES_RING_BUFFER, NULL);
if (thiz) {
thiz->mode = mode;
if (mode == RB_MODE_SRC) {
thiz->acquire = _opensles_recorder_acquire;
thiz->start = _opensles_recorder_start;
thiz->pause = _opensles_recorder_stop;
thiz->stop = _opensles_recorder_stop;
thiz->change_volume = NULL;
} else if (mode == RB_MODE_SINK_PCM) {
thiz->acquire = _opensles_player_acquire;
thiz->start = _opensles_player_start;
thiz->pause = _opensles_player_pause;
thiz->stop = _opensles_player_stop;
thiz->change_volume = _opensles_player_change_volume;
}
}
GST_DEBUG_OBJECT (thiz, "ringbuffer created");
return GST_AUDIO_RING_BUFFER (thiz);
}
void
gst_opensles_ringbuffer_set_volume (GstAudioRingBuffer * rb, gfloat volume)
{
GstOpenSLESRingBuffer *thiz;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
thiz->volume = volume;
if (thiz->change_volume) {
thiz->change_volume (rb);
}
}
void
gst_opensles_ringbuffer_set_mute (GstAudioRingBuffer * rb, gboolean mute)
{
GstOpenSLESRingBuffer *thiz;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
thiz->mute = mute;
if (thiz->change_mute) {
thiz->change_mute (rb);
}
}
static gboolean
gst_opensles_ringbuffer_open_device (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
SLresult result;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
/* Create and realize the engine object */
thiz->engineObject = gst_opensles_get_engine ();
if (!thiz->engineObject) {
GST_ERROR_OBJECT (thiz, "Failed to get engine object");
goto failed;
}
/* Get the engine interface, which is needed in order to create other objects */
result = (*thiz->engineObject)->GetInterface (thiz->engineObject,
SL_IID_ENGINE, &thiz->engineEngine);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "engine.GetInterface(Engine) failed(0x%08x)",
(guint32) result);
goto failed;
}
if (thiz->mode == RB_MODE_SINK_PCM) {
SLOutputMixItf outputMix;
/* Create an output mixer object */
result = (*thiz->engineEngine)->CreateOutputMix (thiz->engineEngine,
&thiz->outputMixObject, 0, NULL, NULL);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "engine.CreateOutputMix failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Realize the output mixer object */
result = (*thiz->outputMixObject)->Realize (thiz->outputMixObject,
SL_BOOLEAN_FALSE);
if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (thiz, "outputMix.Realize failed(0x%08x)",
(guint32) result);
goto failed;
}
/* Get the mixer interface */
result = (*thiz->outputMixObject)->GetInterface (thiz->outputMixObject,
SL_IID_OUTPUTMIX, &outputMix);
if (result != SL_RESULT_SUCCESS) {
GST_WARNING_OBJECT (thiz, "outputMix.GetInterface failed(0x%08x)",
(guint32) result);
} else {
SLint32 numDevices = MAX_NUMBER_OUTPUT_DEVICES;
SLuint32 deviceIDs[MAX_NUMBER_OUTPUT_DEVICES];
gint i;
/* Query the list of output devices */
(*outputMix)->GetDestinationOutputDeviceIDs (outputMix, &numDevices,
deviceIDs);
GST_DEBUG_OBJECT (thiz, "Found %d output devices", (gint) numDevices);
for (i = 0; i < numDevices; i++) {
GST_DEBUG_OBJECT (thiz, " DeviceID: %08x", (guint) deviceIDs[i]);
}
}
}
GST_DEBUG_OBJECT (thiz, "device opened");
return TRUE;
failed:
return FALSE;
}
static gboolean
gst_opensles_ringbuffer_close_device (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
/* Destroy the output mix object */
if (thiz->outputMixObject) {
(*thiz->outputMixObject)->Destroy (thiz->outputMixObject);
thiz->outputMixObject = NULL;
}
/* Destroy the engine object and invalidate all associated interfaces */
if (thiz->engineObject) {
gst_opensles_release_engine (thiz->engineObject);
thiz->engineObject = NULL;
thiz->engineEngine = NULL;
}
thiz->bufferQueue = NULL;
GST_DEBUG_OBJECT (thiz, "device closed");
return TRUE;
}
static gboolean
gst_opensles_ringbuffer_acquire (GstAudioRingBuffer * rb,
GstAudioRingBufferSpec * spec)
{
GstOpenSLESRingBuffer *thiz;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
/* Instantiate and configure the OpenSL ES interfaces */
if (!thiz->acquire (rb, spec)) {
return FALSE;
}
/* Initialize our ringbuffer memory region */
rb->size = spec->segtotal * spec->segsize;
rb->memory = g_malloc0 (rb->size);
GST_DEBUG_OBJECT (thiz, "ringbuffer acquired");
return TRUE;
}
static gboolean
gst_opensles_ringbuffer_release (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
thiz = GST_OPENSLES_RING_BUFFER (rb);
/* XXX: We need to sleep a bit before destroying the player object
* because of a bug in Android in versions < 4.2.
*
* OpenSLES is using AudioTrack for rendering the sound. AudioTrack
* has a thread that pulls raw audio from the buffer queue and then
* passes it forward to AudioFlinger (AudioTrack::processAudioBuffer()).
* This thread is calling various callbacks on events, e.g. when
* an underrun happens or to request data. OpenSLES sets this callback
* on AudioTrack (audioTrack_callBack_pullFromBuffQueue() from
* android_AudioPlayer.cpp). Among other things this is taking a lock
* on the player interface.
*
* Now if we destroy the player interface object, it will first of all
* take the player interface lock (IObject_Destroy()). Then it destroys
* the audio player instance (android_audioPlayer_destroy()) which then
* calls stop() on the AudioTrack and deletes it. Now the destructor of
* AudioTrack will wait until the rendering thread (AudioTrack::processAudioBuffer())
* has finished.
*
* If all this happens with bad timing it can happen that the rendering
* thread is currently e.g. handling underrun but did not lock the player
* interface object yet. Then destroying happens and takes the lock and waits
* for the thread to finish. Then the thread tries to take the lock and waits
* forever.
*
* We wait a bit before destroying the player object to make sure that
* the rendering thread finished whatever it was doing, and then stops
* (note: we called gst_opensles_ringbuffer_stop() before this already).
*/
/* Destroy audio player object, and invalidate all associated interfaces */
if (thiz->playerObject) {
g_usleep (50000);
(*thiz->playerObject)->Destroy (thiz->playerObject);
thiz->playerObject = NULL;
thiz->playerPlay = NULL;
thiz->playerVolume = NULL;
}
/* Destroy audio recorder object, and invalidate all associated interfaces */
if (thiz->recorderObject) {
g_usleep (50000);
(*thiz->recorderObject)->Destroy (thiz->recorderObject);
thiz->recorderObject = NULL;
thiz->recorderRecord = NULL;
}
if (thiz->data) {
g_free (thiz->data);
thiz->data = NULL;
}
if (rb->memory) {
g_free (rb->memory);
rb->memory = NULL;
rb->size = 0;
}
GST_DEBUG_OBJECT (thiz, "ringbuffer released");
return TRUE;
}
static gboolean
gst_opensles_ringbuffer_start (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
gboolean res;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
res = thiz->start (rb);
GST_DEBUG_OBJECT (thiz, "ringbuffer %s started", (res ? "" : "not"));
return res;
}
static gboolean
gst_opensles_ringbuffer_pause (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
gboolean res;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
res = thiz->pause (rb);
GST_DEBUG_OBJECT (thiz, "ringbuffer %s paused", (res ? "" : "not"));
return res;
}
static gboolean
gst_opensles_ringbuffer_stop (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
gboolean res;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
res = thiz->stop (rb);
GST_DEBUG_OBJECT (thiz, "ringbuffer %s stopped", (res ? " " : "not"));
return res;
}
static guint
gst_opensles_ringbuffer_delay (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
guint res = 0;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
if (thiz->playerPlay) {
SLuint32 state;
SLmillisecond position;
guint64 playedpos = 0, queuedpos = 0;
(*thiz->playerPlay)->GetPlayState (thiz->playerPlay, &state);
if (state == SL_PLAYSTATE_PLAYING) {
(*thiz->playerPlay)->GetPosition (thiz->playerPlay, &position);
playedpos =
gst_util_uint64_scale_round (position, rb->spec.info.rate, 1000);
queuedpos = g_atomic_int_get (&thiz->segqueued) * rb->samples_per_seg;
if (queuedpos < playedpos) {
res = 0;
GST_ERROR_OBJECT (thiz,
"Queued position smaller than playback position (%" G_GUINT64_FORMAT
" < %" G_GUINT64_FORMAT ")", queuedpos, playedpos);
} else {
res = queuedpos - playedpos;
}
}
GST_LOG_OBJECT (thiz, "queued samples %" G_GUINT64_FORMAT " position %u ms "
"(%" G_GUINT64_FORMAT " samples) delay %u samples",
queuedpos, (guint) position, playedpos, res);
}
return res;
}
static void
gst_opensles_ringbuffer_clear_all (GstAudioRingBuffer * rb)
{
GstOpenSLESRingBuffer *thiz;
thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
if (thiz->data) {
SLresult result;
memset (thiz->data, 0, thiz->data_size);
g_atomic_int_set (&thiz->segqueued, 0);
thiz->cursor = 0;
/* Reset the queue */
result = (*thiz->bufferQueue)->Clear (thiz->bufferQueue);
if (result != SL_RESULT_SUCCESS) {
GST_WARNING_OBJECT (thiz, "bufferQueue.Clear failed(0x%08x)",
(guint32) result);
}
g_atomic_int_set (&thiz->is_prerolled, 0);
}
GST_CALL_PARENT (GST_AUDIO_RING_BUFFER_CLASS, clear_all, (rb));
}
static void
gst_opensles_ringbuffer_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_opensles_ringbuffer_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_opensles_ringbuffer_class_init (GstOpenSLESRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstAudioRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstringbuffer_class = (GstAudioRingBufferClass *) klass;
gobject_class->dispose = gst_opensles_ringbuffer_dispose;
gobject_class->finalize = gst_opensles_ringbuffer_finalize;
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_release);
gstringbuffer_class->start =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_start);
gstringbuffer_class->pause =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_pause);
gstringbuffer_class->resume =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_stop);
gstringbuffer_class->delay =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_delay);
gstringbuffer_class->clear_all =
GST_DEBUG_FUNCPTR (gst_opensles_ringbuffer_clear_all);
}
static void
gst_opensles_ringbuffer_init (GstOpenSLESRingBuffer * thiz)
{
thiz->mode = RB_MODE_NONE;
thiz->engineObject = NULL;
thiz->outputMixObject = NULL;
thiz->playerObject = NULL;
thiz->recorderObject = NULL;
thiz->is_queue_callback_registered = FALSE;
}
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