mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-06-05 23:18:52 +00:00
opensles: sprinkle comments and cosmetic fixes
This commit is contained in:
Josep Torra
Sebastian Dröge
parent
78e3b9f428
commit
920354eb0d
4 changed files with 112 additions and 43 deletions
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@ -40,7 +40,9 @@ GST_BOILERPLATE_FULL (GstOpenSLESRingBuffer, gst_opensles_ringbuffer,
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#define RECORDER_QUEUE_SIZE 2
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/* Some generic helper functions */
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/*
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* Some generic helper functions
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*/
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static inline SLuint32
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_opensles_sample_rate (guint rate)
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@ -80,7 +82,6 @@ _opensles_sample_rate (guint rate)
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static inline SLuint32
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_opensles_channel_mask (GstRingBufferSpec * spec)
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{
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/* FIXME: handle more than two channels */
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switch (spec->channels) {
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case 1:
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return (SL_SPEAKER_FRONT_CENTER);
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@ -104,7 +105,9 @@ _opensles_format (GstRingBufferSpec * spec, SLDataFormat_PCM * format)
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(spec->bigend ? SL_BYTEORDER_BIGENDIAN : SL_BYTEORDER_LITTLEENDIAN);
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}
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/* Recorder related functions */
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/*
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* Recorder related functions
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*/
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static gboolean
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_opensles_recorder_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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@ -126,13 +129,14 @@ _opensles_recorder_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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};
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SLDataSink audioSink = { &loc_bq, &format };
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/* Required optional interfaces */
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const SLInterfaceID id[1] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
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const SLboolean req[1] = { SL_BOOLEAN_TRUE };
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/* Define the format in OpenSL ES terms */
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/* Define the audio format in OpenSL ES terminology */
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_opensles_format (spec, &format);
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/* Create audio recorder (requires the RECORD_AUDIO permission) */
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/* Create the audio recorder object (requires the RECORD_AUDIO permission) */
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result = (*thiz->engineEngine)->CreateAudioRecorder (thiz->engineEngine,
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&thiz->recorderObject, &audioSrc, &audioSink, 1, id, req);
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if (result != SL_RESULT_SUCCESS) {
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@ -141,7 +145,7 @@ _opensles_recorder_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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goto failed;
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}
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/* Realize the audio recorder */
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/* Realize the audio recorder object */
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result =
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(*thiz->recorderObject)->Realize (thiz->recorderObject, SL_BOOLEAN_FALSE);
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if (result != SL_RESULT_SUCCESS) {
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@ -179,6 +183,11 @@ failed:
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return FALSE;
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}
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/* This callback function is executed when the ringbuffer is started to preroll
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* the output buffer queue with empty buffers, from app thread, and each time
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* there's a filled buffer, from audio device processing thread,
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* the callback behaviour.
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*/
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static void
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_opensles_recorder_cb (SLAndroidSimpleBufferQueueItf bufferQueue, void *context)
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{
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@ -189,21 +198,24 @@ _opensles_recorder_cb (SLAndroidSimpleBufferQueueItf bufferQueue, void *context)
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gint seg;
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gint len;
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/* Get a segment form the GStreamer ringbuffer to write in */
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if (!gst_ring_buffer_prepare_read (rb, &seg, &ptr, &len)) {
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GST_WARNING_OBJECT (rb, "No segment available");
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return;
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}
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/* Enqueue a buffer */
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GST_LOG_OBJECT (thiz, "enqueue: %p size %d segment: %d", ptr, len, seg);
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result = (*thiz->bufferQueue)->Enqueue (thiz->bufferQueue, ptr, len);
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/* Enqueue the sefment as buffer to be written */
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result = (*thiz->bufferQueue)->Enqueue (thiz->bufferQueue, ptr, len);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (thiz, "bufferQueue.Enqueue failed(0x%08x)",
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(guint32) result);
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return;
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}
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/* FIXME: we advance here and behaviour might be racy with the reading
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* thread */
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gst_ring_buffer_advance (rb, 1);
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}
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@ -226,7 +238,7 @@ _opensles_recorder_start (GstRingBuffer * rb)
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thiz->is_queue_callback_registered = TRUE;
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}
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/* Fill the queue by enqueing buffers */
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/* Preroll the buffer queue by enqueing segments */
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for (i = 0; i < RECORDER_QUEUE_SIZE; i++) {
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_opensles_recorder_cb (NULL, rb);
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}
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@ -240,6 +252,7 @@ _opensles_recorder_start (GstRingBuffer * rb)
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(guint32) result);
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return FALSE;
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}
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return TRUE;
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}
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@ -280,7 +293,9 @@ _opensles_recorder_stop (GstRingBuffer * rb)
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return TRUE;
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}
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/* Player related functions */
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/*
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* Player related functions
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*/
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static gboolean
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_opensles_player_change_volume (GstRingBuffer * rb)
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@ -326,6 +341,8 @@ _opensles_player_change_mute (GstRingBuffer * rb)
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return TRUE;
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}
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/* This is a callback function invoked by the playback device thread and
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* it's used to monitor position changes */
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static void
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_opensles_player_event_cb (SLPlayItf caller, void *context, SLuint32 event)
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{
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@ -338,8 +355,6 @@ _opensles_player_event_cb (SLPlayItf caller, void *context, SLuint32 event)
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(*caller)->GetPosition (caller, &position);
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GST_LOG_OBJECT (thiz, "at position=%u ms", (guint) position);
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} else if (event & SL_PLAYEVENT_HEADSTALLED) {
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GST_WARNING_OBJECT (thiz, "head stalled");
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}
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}
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@ -363,13 +378,14 @@ _opensles_player_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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};
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SLDataSink audioSink = { &loc_outmix, NULL };
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/* Create an audio player */
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/* Define the required interfaces */
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const SLInterfaceID ids[2] = { SL_IID_BUFFERQUEUE, SL_IID_VOLUME };
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const SLboolean req[2] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
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/* Define the format in OpenSL ES terms */
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/* Define the format in OpenSL ES terminology */
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_opensles_format (spec, &format);
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/* Create the player object */
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result = (*thiz->engineEngine)->CreateAudioPlayer (thiz->engineEngine,
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&thiz->playerObject, &audioSrc, &audioSink, 2, ids, req);
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if (result != SL_RESULT_SUCCESS) {
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@ -378,7 +394,7 @@ _opensles_player_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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goto failed;
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}
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/* Realize the player */
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/* Realize the player object */
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result =
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(*thiz->playerObject)->Realize (thiz->playerObject, SL_BOOLEAN_FALSE);
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if (result != SL_RESULT_SUCCESS) {
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@ -413,8 +429,8 @@ _opensles_player_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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goto failed;
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}
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/* Request position update events at each 10 ms */
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result = (*thiz->playerPlay)->SetPositionUpdatePeriod (thiz->playerPlay, 10);
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/* Request position update events at each 20 ms */
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result = (*thiz->playerPlay)->SetPositionUpdatePeriod (thiz->playerPlay, 20);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (thiz, "player.SetPositionUpdatePeriod failed(0x%08x)",
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(guint32) result);
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@ -423,7 +439,7 @@ _opensles_player_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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/* Define the event mask to be monitorized */
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result = (*thiz->playerPlay)->SetCallbackEventsMask (thiz->playerPlay,
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SL_PLAYEVENT_HEADATNEWPOS | SL_PLAYEVENT_HEADSTALLED);
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SL_PLAYEVENT_HEADATNEWPOS);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (thiz, "player.SetCallbackEventsMask failed(0x%08x)",
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(guint32) result);
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@ -443,7 +459,7 @@ _opensles_player_acquire (GstRingBuffer * rb, GstRingBufferSpec * spec)
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_opensles_player_change_volume (rb);
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_opensles_player_change_mute (rb);
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/* Define our queue data buffer */
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/* Allocate the queue associated ringbuffer memory */
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thiz->data_segtotal = loc_bufq.numBuffers;
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thiz->data = g_malloc (spec->segsize * thiz->data_segtotal);
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thiz->cursor = 0;
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@ -454,6 +470,16 @@ failed:
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return FALSE;
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}
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/* This callback function is executed when the ringbuffer is started to preroll
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* the input buffer queue with few buffers, from app thread, and each time
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* that rendering of one buffer finishes, from audio device processing thread,
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* the callback behaviour.
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*
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* We wrap the queue behaviour with an appropriate chunk of memory (queue len *
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* ringbuffer segment size) which is used to hold the audio data while it's
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* being processed in the queue. The memory region is used whit a ringbuffer
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* behaviour.
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*/
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static void
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_opensles_player_cb (SLAndroidSimpleBufferQueueItf bufferQueue, void *context)
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{
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@ -464,29 +490,33 @@ _opensles_player_cb (SLAndroidSimpleBufferQueueItf bufferQueue, void *context)
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gint seg;
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gint len;
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/* Get a segment form the GStreamer ringbuffer to read some samples */
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if (!gst_ring_buffer_prepare_read (rb, &seg, &ptr, &len)) {
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GST_WARNING_OBJECT (rb, "No segment available");
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return;
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}
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/* copy data to our queue ringbuffer */
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/* copy the segment data to our queue associated ringbuffer memory */
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cur = thiz->data + (thiz->cursor * rb->spec.segsize);
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memcpy (cur, ptr, len);
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g_atomic_int_inc (&thiz->segqueued);
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/* Enqueue a buffer */
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GST_LOG_OBJECT (thiz, "enqueue: %p size %d segment: %d in queue[%d]",
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cur, len, seg, thiz->cursor);
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result = (*thiz->bufferQueue)->Enqueue (thiz->bufferQueue, cur, len);
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/* advance the cursor in our queue associated ringbuffer */
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thiz->cursor = (thiz->cursor + 1) % thiz->data_segtotal;
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/* Enqueue the buffer to be rendered */
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result = (*thiz->bufferQueue)->Enqueue (thiz->bufferQueue, cur, len);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (thiz, "bufferQueue.Enqueue failed(0x%08x)",
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(guint32) result);
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return;
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}
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/* Fill with silence samples the segment of the GStreamer ringbuffer */
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gst_ring_buffer_clear (rb, seg);
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/* Make the segment reusable */
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gst_ring_buffer_advance (rb, 1);
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}
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@ -540,6 +570,7 @@ _opensles_player_pause (GstRingBuffer * rb)
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(guint32) result);
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return FALSE;
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}
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return TRUE;
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}
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@ -553,7 +584,6 @@ _opensles_player_stop (GstRingBuffer * rb)
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result =
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(*thiz->playerPlay)->SetPlayState (thiz->playerPlay,
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SL_PLAYSTATE_STOPPED);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (thiz, "player.SetPlayState failed(0x%08x)",
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(guint32) result);
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@ -578,13 +608,17 @@ _opensles_player_stop (GstRingBuffer * rb)
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return FALSE;
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}
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/* Reset our state */
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g_atomic_int_set (&thiz->segqueued, 0);
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thiz->cursor = 0;
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return TRUE;
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}
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/* OpenSL ES ring buffer wrapper */
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/*
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* OpenSL ES ringbuffer wrapper
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*/
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GstRingBuffer *
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gst_opensles_ringbuffer_new (RingBufferMode mode)
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@ -613,6 +647,7 @@ gst_opensles_ringbuffer_new (RingBufferMode mode)
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}
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GST_DEBUG_OBJECT (thiz, "ringbuffer created");
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return GST_RING_BUFFER (thiz);
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}
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@ -652,14 +687,14 @@ gst_opensles_ringbuffer_open_device (GstRingBuffer * rb)
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thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
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/* Create engine */
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/* Create the engine object */
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result = slCreateEngine (&thiz->engineObject, 0, NULL, 0, NULL, NULL);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (thiz, "slCreateEngine failed(0x%08x)", (guint32) result);
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goto failed;
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}
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/* Realize the engine */
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/* Realize the engine object */
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result = (*thiz->engineObject)->Realize (thiz->engineObject,
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SL_BOOLEAN_FALSE);
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if (result != SL_RESULT_SUCCESS) {
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@ -667,8 +702,7 @@ gst_opensles_ringbuffer_open_device (GstRingBuffer * rb)
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goto failed;
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}
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/* Get the engine interface, which is needed in order to
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* create other objects */
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/* Get the engine interface, which is needed in order to create other objects */
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result = (*thiz->engineObject)->GetInterface (thiz->engineObject,
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SL_IID_ENGINE, &thiz->engineEngine);
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if (result != SL_RESULT_SUCCESS) {
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@ -680,7 +714,7 @@ gst_opensles_ringbuffer_open_device (GstRingBuffer * rb)
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if (thiz->mode == RB_MODE_SINK_PCM) {
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SLOutputMixItf outputMix;
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/* Create an output mixer */
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/* Create an output mixer object */
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result = (*thiz->engineEngine)->CreateOutputMix (thiz->engineEngine,
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&thiz->outputMixObject, 0, NULL, NULL);
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if (result != SL_RESULT_SUCCESS) {
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@ -689,7 +723,7 @@ gst_opensles_ringbuffer_open_device (GstRingBuffer * rb)
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goto failed;
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}
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/* Realize the output mixer */
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/* Realize the output mixer object */
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result = (*thiz->outputMixObject)->Realize (thiz->outputMixObject,
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SL_BOOLEAN_FALSE);
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if (result != SL_RESULT_SUCCESS) {
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@ -698,16 +732,18 @@ gst_opensles_ringbuffer_open_device (GstRingBuffer * rb)
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goto failed;
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}
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/* Check for output device options */
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/* Get the mixer interface */
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result = (*thiz->outputMixObject)->GetInterface (thiz->outputMixObject,
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SL_IID_OUTPUTMIX, &outputMix);
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if (result != SL_RESULT_SUCCESS) {
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GST_WARNING_OBJECT (thiz, "outputMix.GetInterface failed(0x%08x)",
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(guint32) result);
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} else {
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SLint32 numDevices;
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SLuint32 deviceIDs[16];
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SLint32 numDevices = 0;
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SLuint32 deviceIDs[MAX_NUMBER_OUTPUT_DEVICES];
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gint i;
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/* Query the list of output devices */
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(*outputMix)->GetDestinationOutputDeviceIDs (outputMix, &numDevices,
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deviceIDs);
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GST_DEBUG_OBJECT (thiz, "Found %d output devices", (gint) numDevices);
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@ -731,13 +767,13 @@ gst_opensles_ringbuffer_close_device (GstRingBuffer * rb)
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thiz = GST_OPENSLES_RING_BUFFER_CAST (rb);
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/* Destroy output mix object */
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/* Destroy the output mix object */
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if (thiz->outputMixObject) {
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(*thiz->outputMixObject)->Destroy (thiz->outputMixObject);
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thiz->outputMixObject = NULL;
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}
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/* Destroy engine object, and invalidate all associated interfaces */
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/* Destroy the engine object and invalidate all associated interfaces */
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if (thiz->engineObject) {
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(*thiz->engineObject)->Destroy (thiz->engineObject);
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thiz->engineObject = NULL;
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@ -801,6 +837,7 @@ gst_opensles_ringbuffer_release (GstRingBuffer * rb)
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gst_buffer_unref (rb->data);
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rb->data = NULL;
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}
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GST_DEBUG_OBJECT (thiz, "ringbuffer released");
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return TRUE;
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}
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@ -28,6 +28,8 @@
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G_BEGIN_DECLS
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#define MAX_NUMBER_OUTPUT_DEVICES 16
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#define GST_TYPE_OPENSLES_RING_BUFFER \
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(gst_opensles_ringbuffer_get_type())
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#define GST_OPENSLES_RING_BUFFER(obj) \
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@ -17,6 +17,21 @@
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-openslessink
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* @see_also: openslessrc
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*
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* This element renders raw audio samples using the OpenSL ES API in Android OS.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -v filesrc location=music.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! opeslessink
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* ]| Play an Ogg/Vorbis file.
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* </refsect2>
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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|
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@ -40,6 +55,9 @@ enum
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/* According to Android's NDK doc the following are the supported rates */
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#define RATES "8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100"
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/* 48000 Hz is also claimed to be supported but the AudioFlinger downsampling
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* doesn't seems to work properly so we relay GStreamer audioresample element
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* to cope with this samplerate. */
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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|
|
@ -106,11 +124,6 @@ gst_opensles_sink_create_ringbuffer (GstBaseAudioSink * base)
|
|||
(gint) (aod)->maxSampleRate, (gint) (aod)->isFreqRangeContinuous, \
|
||||
(gint) (aod)->maxChannels
|
||||
|
||||
/* Next it's not defined in Android */
|
||||
#ifndef MAX_NUMBER_OUTPUT_DEVICES
|
||||
#define MAX_NUMBER_OUTPUT_DEVICES 16
|
||||
#endif
|
||||
|
||||
static gboolean
|
||||
_opensles_query_capabilities (GstOpenSLESSink * sink)
|
||||
{
|
||||
|
|
@ -136,8 +149,7 @@ _opensles_query_capabilities (GstOpenSLESSink * sink)
|
|||
goto beach;
|
||||
}
|
||||
|
||||
/* Get the engine interface, which is needed in order to
|
||||
* create other objects */
|
||||
/* Get the engine interface, which is needed in order to create other objects */
|
||||
result = (*engineObject)->GetInterface (engineObject,
|
||||
SL_IID_AUDIOIODEVICECAPABILITIES, &audioIODeviceCapabilities);
|
||||
if (result != SL_RESULT_SUCCESS) {
|
||||
|
|
@ -147,6 +159,7 @@ _opensles_query_capabilities (GstOpenSLESSink * sink)
|
|||
goto beach;
|
||||
}
|
||||
|
||||
/* Query the list of available audio outputs */
|
||||
result = (*audioIODeviceCapabilities)->GetAvailableAudioOutputs
|
||||
(audioIODeviceCapabilities, &numOutputs, outputDeviceIDs);
|
||||
if (result != SL_RESULT_SUCCESS) {
|
||||
|
|
@ -182,7 +195,7 @@ _opensles_query_capabilities (GstOpenSLESSink * sink)
|
|||
|
||||
res = TRUE;
|
||||
beach:
|
||||
/* Destroy engine object */
|
||||
/* Destroy the engine object */
|
||||
if (engineObject) {
|
||||
(*engineObject)->Destroy (engineObject);
|
||||
}
|
||||
|
|
@ -269,6 +282,8 @@ gst_opensles_sink_init (GstOpenSLESSink * sink, GstOpenSLESSinkClass * gclass)
|
|||
_opensles_query_capabilities (sink);
|
||||
|
||||
gst_base_audio_sink_set_provide_clock (GST_BASE_AUDIO_SINK (sink), TRUE);
|
||||
/* Override some default values to fit on the AudioFlinger behaviour of
|
||||
* processing 20ms buffers as minimum buffer size. */
|
||||
GST_BASE_AUDIO_SINK (sink)->buffer_time = 400000;
|
||||
GST_BASE_AUDIO_SINK (sink)->latency_time = 20000;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -17,6 +17,21 @@
|
|||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
/**
|
||||
* SECTION:element-openslessrc
|
||||
* @see_also: openslessink
|
||||
*
|
||||
* This element reads data from default audio input using the OpenSL ES API in Android OS.
|
||||
*
|
||||
* <refsect2>
|
||||
* <title>Example pipelines</title>
|
||||
* |[
|
||||
* gst-launch -v openslessrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=recorded.ogg
|
||||
* ]| Record from default audio input and encode to Ogg/Vorbis.
|
||||
* </refsect2>
|
||||
*
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
# include <config.h>
|
||||
#endif
|
||||
|
|
|
|||
Loading…
Reference in a new issue