2 * Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
3 * Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
4 * Copyright (C) 2013, Collabora Ltd.
5 * Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
6 * Copyright (C) 2014, Sebastian Dröge <sebastian@centricular.com>
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation
11 * version 2.1 of the License.
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with this library; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
32 #include "gstomxaudiodec.h"
34 GST_DEBUG_CATEGORY_STATIC (gst_omx_audio_dec_debug_category);
35 #define GST_CAT_DEFAULT gst_omx_audio_dec_debug_category
38 static void gst_omx_audio_dec_finalize (GObject * object);
40 static GstStateChangeReturn
41 gst_omx_audio_dec_change_state (GstElement * element,
42 GstStateChange transition);
44 static gboolean gst_omx_audio_dec_open (GstAudioDecoder * decoder);
45 static gboolean gst_omx_audio_dec_close (GstAudioDecoder * decoder);
46 static gboolean gst_omx_audio_dec_start (GstAudioDecoder * decoder);
47 static gboolean gst_omx_audio_dec_stop (GstAudioDecoder * decoder);
48 static gboolean gst_omx_audio_dec_set_format (GstAudioDecoder * decoder,
50 static void gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard);
51 static GstFlowReturn gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder,
53 static GstFlowReturn gst_omx_audio_dec_drain (GstOMXAudioDec * self);
60 /* class initialization */
63 GST_DEBUG_CATEGORY_INIT (gst_omx_audio_dec_debug_category, "omxaudiodec", 0, \
64 "debug category for gst-omx audio decoder base class");
67 G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstOMXAudioDec, gst_omx_audio_dec,
68 GST_TYPE_AUDIO_DECODER, DEBUG_INIT);
71 gst_omx_audio_dec_class_init (GstOMXAudioDecClass * klass)
73 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
74 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
75 GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass);
77 gobject_class->finalize = gst_omx_audio_dec_finalize;
79 element_class->change_state =
80 GST_DEBUG_FUNCPTR (gst_omx_audio_dec_change_state);
82 audio_decoder_class->open = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_open);
83 audio_decoder_class->close = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_close);
84 audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_start);
85 audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_stop);
86 audio_decoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_flush);
87 audio_decoder_class->set_format =
88 GST_DEBUG_FUNCPTR (gst_omx_audio_dec_set_format);
89 audio_decoder_class->handle_frame =
90 GST_DEBUG_FUNCPTR (gst_omx_audio_dec_handle_frame);
92 klass->cdata.type = GST_OMX_COMPONENT_TYPE_FILTER;
93 klass->cdata.default_src_template_caps =
95 "rate = (int) [ 1, MAX ], "
96 "channels = (int) [ 1, " G_STRINGIFY (OMX_AUDIO_MAXCHANNELS) " ], "
97 "format = (string) " GST_AUDIO_FORMATS_ALL;
101 gst_omx_audio_dec_init (GstOMXAudioDec * self)
103 gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
104 gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
105 gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
107 GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (self));
109 g_mutex_init (&self->drain_lock);
110 g_cond_init (&self->drain_cond);
112 self->output_adapter = gst_adapter_new ();
116 gst_omx_audio_dec_open (GstAudioDecoder * decoder)
118 GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
119 GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
120 gint in_port_index, out_port_index;
122 GST_DEBUG_OBJECT (self, "Opening decoder");
125 gst_omx_component_new (GST_OBJECT_CAST (self), klass->cdata.core_name,
126 klass->cdata.component_name, klass->cdata.component_role,
128 self->started = FALSE;
133 if (gst_omx_component_get_state (self->dec,
134 GST_CLOCK_TIME_NONE) != OMX_StateLoaded)
137 in_port_index = klass->cdata.in_port_index;
138 out_port_index = klass->cdata.out_port_index;
140 if (in_port_index == -1 || out_port_index == -1) {
141 OMX_PORT_PARAM_TYPE param;
144 GST_OMX_INIT_STRUCT (¶m);
147 gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioInit,
149 if (err != OMX_ErrorNone) {
150 GST_WARNING_OBJECT (self, "Couldn't get port information: %s (0x%08x)",
151 gst_omx_error_to_string (err), err);
156 GST_DEBUG_OBJECT (self, "Detected %u ports, starting at %u",
157 (guint) param.nPorts, (guint) param.nStartPortNumber);
158 in_port_index = param.nStartPortNumber + 0;
159 out_port_index = param.nStartPortNumber + 1;
162 self->dec_in_port = gst_omx_component_add_port (self->dec, in_port_index);
163 self->dec_out_port = gst_omx_component_add_port (self->dec, out_port_index);
165 if (!self->dec_in_port || !self->dec_out_port)
168 GST_DEBUG_OBJECT (self, "Opened decoder");
174 gst_omx_audio_dec_shutdown (GstOMXAudioDec * self)
178 GST_DEBUG_OBJECT (self, "Shutting down decoder");
180 state = gst_omx_component_get_state (self->dec, 0);
181 if (state > OMX_StateLoaded || state == OMX_StateInvalid) {
182 if (state > OMX_StateIdle) {
183 gst_omx_component_set_state (self->dec, OMX_StateIdle);
184 gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
186 gst_omx_component_set_state (self->dec, OMX_StateLoaded);
187 gst_omx_port_deallocate_buffers (self->dec_in_port);
188 gst_omx_port_deallocate_buffers (self->dec_out_port);
189 if (state > OMX_StateLoaded)
190 gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
197 gst_omx_audio_dec_close (GstAudioDecoder * decoder)
199 GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
201 GST_DEBUG_OBJECT (self, "Closing decoder");
203 if (!gst_omx_audio_dec_shutdown (self))
206 self->dec_in_port = NULL;
207 self->dec_out_port = NULL;
209 gst_omx_component_free (self->dec);
212 self->started = FALSE;
214 GST_DEBUG_OBJECT (self, "Closed decoder");
220 gst_omx_audio_dec_finalize (GObject * object)
222 GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (object);
224 g_mutex_clear (&self->drain_lock);
225 g_cond_clear (&self->drain_cond);
227 if (self->output_adapter)
228 gst_object_unref (self->output_adapter);
229 self->output_adapter = NULL;
231 G_OBJECT_CLASS (gst_omx_audio_dec_parent_class)->finalize (object);
234 static GstStateChangeReturn
235 gst_omx_audio_dec_change_state (GstElement * element, GstStateChange transition)
237 GstOMXAudioDec *self;
238 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
240 g_return_val_if_fail (GST_IS_OMX_AUDIO_DEC (element),
241 GST_STATE_CHANGE_FAILURE);
242 self = GST_OMX_AUDIO_DEC (element);
244 switch (transition) {
245 case GST_STATE_CHANGE_NULL_TO_READY:
247 case GST_STATE_CHANGE_READY_TO_PAUSED:
248 self->downstream_flow_ret = GST_FLOW_OK;
249 self->draining = FALSE;
250 self->started = FALSE;
252 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
254 case GST_STATE_CHANGE_PAUSED_TO_READY:
255 if (self->dec_in_port)
256 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
257 if (self->dec_out_port)
258 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
260 g_mutex_lock (&self->drain_lock);
261 self->draining = FALSE;
262 g_cond_broadcast (&self->drain_cond);
263 g_mutex_unlock (&self->drain_lock);
270 GST_ELEMENT_CLASS (gst_omx_audio_dec_parent_class)->change_state
271 (element, transition);
273 if (ret == GST_STATE_CHANGE_FAILURE)
276 switch (transition) {
277 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
279 case GST_STATE_CHANGE_PAUSED_TO_READY:
280 self->downstream_flow_ret = GST_FLOW_FLUSHING;
281 self->started = FALSE;
283 if (!gst_omx_audio_dec_shutdown (self))
284 ret = GST_STATE_CHANGE_FAILURE;
286 case GST_STATE_CHANGE_READY_TO_NULL:
296 gst_omx_audio_dec_loop (GstOMXAudioDec * self)
298 GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
299 GstOMXPort *port = self->dec_out_port;
300 GstOMXBuffer *buf = NULL;
301 GstFlowReturn flow_ret = GST_FLOW_OK;
302 GstOMXAcquireBufferReturn acq_return;
306 acq_return = gst_omx_port_acquire_buffer (port, &buf);
307 if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) {
308 goto component_error;
309 } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
311 } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
315 if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (self)) ||
316 acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
317 OMX_PARAM_PORTDEFINITIONTYPE port_def;
318 OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
319 GstAudioChannelPosition omx_position[OMX_AUDIO_MAXCHANNELS];
320 GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
323 GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
325 /* Reallocate all buffers */
326 if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE
327 && gst_omx_port_is_enabled (port)) {
328 err = gst_omx_port_set_enabled (port, FALSE);
329 if (err != OMX_ErrorNone)
330 goto reconfigure_error;
332 err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
333 if (err != OMX_ErrorNone)
334 goto reconfigure_error;
336 err = gst_omx_port_deallocate_buffers (port);
337 if (err != OMX_ErrorNone)
338 goto reconfigure_error;
340 err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
341 if (err != OMX_ErrorNone)
342 goto reconfigure_error;
345 /* Just update caps */
346 GST_AUDIO_DECODER_STREAM_LOCK (self);
348 gst_omx_port_get_port_definition (port, &port_def);
349 g_assert (port_def.format.audio.eEncoding == OMX_AUDIO_CodingPCM);
351 GST_OMX_INIT_STRUCT (&pcm_param);
352 pcm_param.nPortIndex = self->dec_out_port->index;
354 gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioPcm,
356 if (err != OMX_ErrorNone) {
357 GST_ERROR_OBJECT (self, "Failed to get PCM parameters: %s (0x%08x)",
358 gst_omx_error_to_string (err), err);
362 g_assert (pcm_param.ePCMMode == OMX_AUDIO_PCMModeLinear);
363 g_assert (pcm_param.bInterleaved == OMX_TRUE);
365 gst_audio_info_init (&self->info);
367 for (i = 0; i < pcm_param.nChannels; i++) {
368 switch (pcm_param.eChannelMapping[i]) {
369 case OMX_AUDIO_ChannelLF:
370 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
372 case OMX_AUDIO_ChannelRF:
373 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
375 case OMX_AUDIO_ChannelCF:
376 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
378 case OMX_AUDIO_ChannelLS:
379 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
381 case OMX_AUDIO_ChannelRS:
382 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
384 case OMX_AUDIO_ChannelLFE:
385 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
387 case OMX_AUDIO_ChannelCS:
388 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
390 case OMX_AUDIO_ChannelLR:
391 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
393 case OMX_AUDIO_ChannelRR:
394 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
396 case OMX_AUDIO_ChannelNone:
398 /* This will break the outer loop too as the
399 * i == pcm_param.nChannels afterwards */
400 for (i = 0; i < pcm_param.nChannels; i++)
401 omx_position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
405 if (pcm_param.nChannels == 1
406 && omx_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER)
407 omx_position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
409 if (omx_position[0] == GST_AUDIO_CHANNEL_POSITION_NONE
410 && klass->get_channel_positions) {
411 GST_WARNING_OBJECT (self,
412 "Failed to get a valid channel layout, trying fallback");
413 klass->get_channel_positions (self, self->dec_out_port, omx_position);
416 memcpy (self->position, omx_position, sizeof (omx_position));
417 gst_audio_channel_positions_to_valid_order (self->position,
418 pcm_param.nChannels);
419 self->needs_reorder =
420 (memcmp (self->position, omx_position,
421 sizeof (GstAudioChannelPosition) * pcm_param.nChannels) != 0);
422 if (self->needs_reorder)
423 gst_audio_get_channel_reorder_map (pcm_param.nChannels, self->position,
424 omx_position, self->reorder_map);
426 gst_audio_info_set_format (&self->info,
427 gst_audio_format_build_integer (pcm_param.eNumData ==
428 OMX_NumericalDataSigned,
430 OMX_EndianLittle ? G_LITTLE_ENDIAN : G_BIG_ENDIAN,
431 pcm_param.nBitPerSample, pcm_param.nBitPerSample),
432 pcm_param.nSamplingRate, pcm_param.nChannels, self->position);
434 GST_DEBUG_OBJECT (self,
435 "Setting output state: format %s, rate %u, channels %u",
436 gst_audio_format_to_string (self->info.finfo->format),
437 (guint) pcm_param.nSamplingRate, (guint) pcm_param.nChannels);
439 if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self),
441 || !gst_audio_decoder_negotiate (GST_AUDIO_DECODER (self))) {
443 gst_omx_port_release_buffer (port, buf);
447 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
449 if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
450 err = gst_omx_port_set_enabled (port, TRUE);
451 if (err != OMX_ErrorNone)
452 goto reconfigure_error;
454 err = gst_omx_port_allocate_buffers (port);
455 if (err != OMX_ErrorNone)
456 goto reconfigure_error;
458 err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
459 if (err != OMX_ErrorNone)
460 goto reconfigure_error;
462 err = gst_omx_port_populate (port);
463 if (err != OMX_ErrorNone)
464 goto reconfigure_error;
466 err = gst_omx_port_mark_reconfigured (port);
467 if (err != OMX_ErrorNone)
468 goto reconfigure_error;
471 /* Now get a buffer */
472 if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) {
477 g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
479 g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
480 GST_AUDIO_DECODER_STREAM_LOCK (self);
484 /* This prevents a deadlock between the srcpad stream
485 * lock and the audiocodec stream lock, if ::reset()
486 * is called at the wrong time
488 if (gst_omx_port_is_flushing (port)) {
489 GST_DEBUG_OBJECT (self, "Flushing");
490 gst_omx_port_release_buffer (port, buf);
494 GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
495 (guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp);
497 GST_AUDIO_DECODER_STREAM_LOCK (self);
499 spf = klass->get_samples_per_frame (self, self->dec_out_port);
501 if (buf->omx_buf->nFilledLen > 0) {
505 GST_DEBUG_OBJECT (self, "Handling output data");
507 if (buf->omx_buf->nFilledLen % self->info.bpf != 0) {
508 gst_omx_port_release_buffer (port, buf);
513 gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
514 buf->omx_buf->nFilledLen);
516 gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE);
517 if (self->needs_reorder) {
518 gint i, n_samples, c, n_channels;
519 gint *reorder_map = self->reorder_map;
520 gint16 *dest, *source;
522 dest = (gint16 *) minfo.data;
523 source = (gint16 *) (buf->omx_buf->pBuffer + buf->omx_buf->nOffset);
524 n_samples = buf->omx_buf->nFilledLen / self->info.bpf;
525 n_channels = self->info.channels;
527 for (i = 0; i < n_samples; i++) {
528 for (c = 0; c < n_channels; c++) {
529 dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c];
533 memcpy (minfo.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
534 buf->omx_buf->nFilledLen);
536 gst_buffer_unmap (outbuf, &minfo);
539 gst_adapter_push (self->output_adapter, outbuf);
542 gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
546 GST_DEBUG_OBJECT (self, "Read frame from component");
550 guint avail = gst_adapter_available (self->output_adapter);
553 /* We take a multiple of codec frames and push
556 avail /= self->info.bpf;
557 nframes = avail / spf;
558 avail = nframes * spf;
559 avail *= self->info.bpf;
562 outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
564 gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
569 GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
572 err = gst_omx_port_release_buffer (port, buf);
573 if (err != OMX_ErrorNone)
577 self->downstream_flow_ret = flow_ret;
579 if (flow_ret != GST_FLOW_OK)
582 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
588 GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
589 ("OpenMAX component in error state %s (0x%08x)",
590 gst_omx_component_get_last_error_string (self->dec),
591 gst_omx_component_get_last_error (self->dec)));
592 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
593 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
594 self->downstream_flow_ret = GST_FLOW_ERROR;
595 self->started = FALSE;
601 GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
602 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
603 self->downstream_flow_ret = GST_FLOW_FLUSHING;
604 self->started = FALSE;
610 spf = klass->get_samples_per_frame (self, self->dec_out_port);
613 guint avail = gst_adapter_available (self->output_adapter);
616 /* On EOS we take the complete adapter content, no matter
617 * if it is a multiple of the codec frame size or not.
619 avail /= self->info.bpf;
620 nframes = (avail + spf - 1) / spf;
621 avail *= self->info.bpf;
624 outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
626 gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
631 g_mutex_lock (&self->drain_lock);
632 if (self->draining) {
633 GST_DEBUG_OBJECT (self, "Drained");
634 self->draining = FALSE;
635 g_cond_broadcast (&self->drain_cond);
636 flow_ret = GST_FLOW_OK;
637 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
639 GST_DEBUG_OBJECT (self, "Component signalled EOS");
640 flow_ret = GST_FLOW_EOS;
642 g_mutex_unlock (&self->drain_lock);
644 GST_AUDIO_DECODER_STREAM_LOCK (self);
645 self->downstream_flow_ret = flow_ret;
647 /* Here we fallback and pause the task for the EOS case */
648 if (flow_ret != GST_FLOW_OK)
651 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
658 if (flow_ret == GST_FLOW_EOS) {
659 GST_DEBUG_OBJECT (self, "EOS");
661 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
662 gst_event_new_eos ());
663 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
664 self->started = FALSE;
665 } else if (flow_ret < GST_FLOW_EOS) {
666 GST_ELEMENT_ERROR (self, STREAM, FAILED,
667 ("Internal data stream error."), ("stream stopped, reason %s",
668 gst_flow_get_name (flow_ret)));
670 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
671 gst_event_new_eos ());
672 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
673 self->started = FALSE;
674 } else if (flow_ret == GST_FLOW_FLUSHING) {
675 GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
676 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
677 self->started = FALSE;
679 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
685 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
686 ("Unable to reconfigure output port"));
687 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
688 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
689 self->downstream_flow_ret = GST_FLOW_ERROR;
690 self->started = FALSE;
696 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
697 ("Invalid sized input buffer"));
698 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
699 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
700 self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
701 self->started = FALSE;
702 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
708 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
709 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
710 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
711 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
712 self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
713 self->started = FALSE;
718 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
719 ("Failed to relase output buffer to component: %s (0x%08x)",
720 gst_omx_error_to_string (err), err));
721 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
722 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
723 self->downstream_flow_ret = GST_FLOW_ERROR;
724 self->started = FALSE;
725 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
731 gst_omx_audio_dec_start (GstAudioDecoder * decoder)
733 GstOMXAudioDec *self;
735 self = GST_OMX_AUDIO_DEC (decoder);
737 self->last_upstream_ts = 0;
738 self->downstream_flow_ret = GST_FLOW_OK;
744 gst_omx_audio_dec_stop (GstAudioDecoder * decoder)
746 GstOMXAudioDec *self;
748 self = GST_OMX_AUDIO_DEC (decoder);
750 GST_DEBUG_OBJECT (self, "Stopping decoder");
752 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
753 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
755 gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
757 if (gst_omx_component_get_state (self->dec, 0) > OMX_StateIdle)
758 gst_omx_component_set_state (self->dec, OMX_StateIdle);
760 self->downstream_flow_ret = GST_FLOW_FLUSHING;
761 self->started = FALSE;
763 g_mutex_lock (&self->drain_lock);
764 self->draining = FALSE;
765 g_cond_broadcast (&self->drain_cond);
766 g_mutex_unlock (&self->drain_lock);
768 gst_adapter_flush (self->output_adapter,
769 gst_adapter_available (self->output_adapter));
771 gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
773 gst_buffer_replace (&self->codec_data, NULL);
775 GST_DEBUG_OBJECT (self, "Stopped decoder");
781 gst_omx_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
783 GstOMXAudioDec *self;
784 GstOMXAudioDecClass *klass;
786 const GValue *codec_data;
787 gboolean is_format_change = FALSE;
788 gboolean needs_disable = FALSE;
790 self = GST_OMX_AUDIO_DEC (decoder);
791 klass = GST_OMX_AUDIO_DEC_GET_CLASS (decoder);
793 GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
795 /* Check if the caps change is a real format change or if only irrelevant
796 * parts of the caps have changed or nothing at all.
798 if (klass->is_format_change)
799 is_format_change = klass->is_format_change (self, self->dec_in_port, caps);
802 gst_omx_component_get_state (self->dec,
803 GST_CLOCK_TIME_NONE) != OMX_StateLoaded;
804 /* If the component is not in Loaded state and a real format change happens
805 * we have to disable the port and re-allocate all buffers. If no real
806 * format change happened we can just exit here.
808 if (needs_disable && !is_format_change) {
809 GST_DEBUG_OBJECT (self,
810 "Already running and caps did not change the format");
814 if (needs_disable && is_format_change) {
815 GstOMXPort *out_port = self->dec_out_port;
817 GST_DEBUG_OBJECT (self, "Need to disable and drain decoder");
819 gst_omx_audio_dec_drain (self);
820 gst_omx_audio_dec_flush (decoder, FALSE);
821 gst_omx_port_set_flushing (out_port, 5 * GST_SECOND, TRUE);
823 if (klass->cdata.hacks & GST_OMX_HACK_NO_COMPONENT_RECONFIGURE) {
824 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
825 gst_omx_audio_dec_stop (GST_AUDIO_DECODER (self));
826 gst_omx_audio_dec_close (GST_AUDIO_DECODER (self));
827 GST_AUDIO_DECODER_STREAM_LOCK (self);
829 if (!gst_omx_audio_dec_open (GST_AUDIO_DECODER (self)))
831 needs_disable = FALSE;
833 if (gst_omx_port_set_enabled (self->dec_in_port, FALSE) != OMX_ErrorNone)
835 if (gst_omx_port_set_enabled (out_port, FALSE) != OMX_ErrorNone)
837 if (gst_omx_port_wait_buffers_released (self->dec_in_port,
838 5 * GST_SECOND) != OMX_ErrorNone)
840 if (gst_omx_port_wait_buffers_released (out_port,
841 1 * GST_SECOND) != OMX_ErrorNone)
843 if (gst_omx_port_deallocate_buffers (self->dec_in_port) != OMX_ErrorNone)
845 if (gst_omx_port_deallocate_buffers (self->dec_out_port) != OMX_ErrorNone)
847 if (gst_omx_port_wait_enabled (self->dec_in_port,
848 1 * GST_SECOND) != OMX_ErrorNone)
850 if (gst_omx_port_wait_enabled (out_port, 1 * GST_SECOND) != OMX_ErrorNone)
854 GST_DEBUG_OBJECT (self, "Decoder drained and disabled");
857 if (klass->set_format) {
858 if (!klass->set_format (self, self->dec_in_port, caps)) {
859 GST_ERROR_OBJECT (self, "Subclass failed to set the new format");
864 GST_DEBUG_OBJECT (self, "Updating outport port definition");
865 if (gst_omx_port_update_port_definition (self->dec_out_port,
866 NULL) != OMX_ErrorNone)
869 /* Get codec data from caps */
870 gst_buffer_replace (&self->codec_data, NULL);
871 s = gst_caps_get_structure (caps, 0);
872 codec_data = gst_structure_get_value (s, "codec_data");
874 /* Vorbis and some other codecs have multiple buffers in
875 * the stream-header field */
876 self->codec_data = gst_value_get_buffer (codec_data);
877 if (self->codec_data)
878 gst_buffer_ref (self->codec_data);
881 GST_DEBUG_OBJECT (self, "Enabling component");
884 if (gst_omx_port_set_enabled (self->dec_in_port, TRUE) != OMX_ErrorNone)
886 if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
889 if ((klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
890 if (gst_omx_port_set_enabled (self->dec_out_port, TRUE) != OMX_ErrorNone)
892 if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone)
895 if (gst_omx_port_wait_enabled (self->dec_out_port,
896 5 * GST_SECOND) != OMX_ErrorNone)
900 if (gst_omx_port_wait_enabled (self->dec_in_port,
901 5 * GST_SECOND) != OMX_ErrorNone)
903 if (gst_omx_port_mark_reconfigured (self->dec_in_port) != OMX_ErrorNone)
906 if (!(klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
907 /* Disable output port */
908 if (gst_omx_port_set_enabled (self->dec_out_port, FALSE) != OMX_ErrorNone)
911 if (gst_omx_port_wait_enabled (self->dec_out_port,
912 1 * GST_SECOND) != OMX_ErrorNone)
915 if (gst_omx_component_set_state (self->dec,
916 OMX_StateIdle) != OMX_ErrorNone)
919 /* Need to allocate buffers to reach Idle state */
920 if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
923 if (gst_omx_component_set_state (self->dec,
924 OMX_StateIdle) != OMX_ErrorNone)
927 /* Need to allocate buffers to reach Idle state */
928 if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
930 if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone)
934 if (gst_omx_component_get_state (self->dec,
935 GST_CLOCK_TIME_NONE) != OMX_StateIdle)
938 if (gst_omx_component_set_state (self->dec,
939 OMX_StateExecuting) != OMX_ErrorNone)
942 if (gst_omx_component_get_state (self->dec,
943 GST_CLOCK_TIME_NONE) != OMX_StateExecuting)
947 /* Unset flushing to allow ports to accept data again */
948 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
949 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
951 if (gst_omx_component_get_last_error (self->dec) != OMX_ErrorNone) {
952 GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)",
953 gst_omx_component_get_last_error_string (self->dec),
954 gst_omx_component_get_last_error (self->dec));
958 self->downstream_flow_ret = GST_FLOW_OK;
964 gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
966 GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
967 OMX_ERRORTYPE err = OMX_ErrorNone;
969 GST_DEBUG_OBJECT (self, "Flushing decoder");
971 if (gst_omx_component_get_state (self->dec, 0) == OMX_StateLoaded)
974 /* 0) Pause the components */
975 if (gst_omx_component_get_state (self->dec, 0) == OMX_StateExecuting) {
976 gst_omx_component_set_state (self->dec, OMX_StatePause);
977 gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
980 /* 1) Wait until the srcpad loop is stopped,
981 * unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
982 * caused by using this lock from inside the loop function */
983 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
984 gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
985 GST_DEBUG_OBJECT (self, "Flushing -- task stopped");
986 GST_AUDIO_DECODER_STREAM_LOCK (self);
988 /* 2) Flush the ports */
989 GST_DEBUG_OBJECT (self, "flushing ports");
990 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
991 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
993 /* 3) Resume components */
994 gst_omx_component_set_state (self->dec, OMX_StateExecuting);
995 gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
997 /* 4) Unset flushing to allow ports to accept data again */
998 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
999 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
1001 err = gst_omx_port_populate (self->dec_out_port);
1003 if (err != OMX_ErrorNone) {
1004 GST_WARNING_OBJECT (self, "Failed to populate output port: %s (0x%08x)",
1005 gst_omx_error_to_string (err), err);
1008 /* Reset our state */
1009 gst_adapter_flush (self->output_adapter,
1010 gst_adapter_available (self->output_adapter));
1011 self->last_upstream_ts = 0;
1012 self->downstream_flow_ret = GST_FLOW_OK;
1013 self->started = FALSE;
1014 GST_DEBUG_OBJECT (self, "Flush finished");
1017 static GstFlowReturn
1018 gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
1020 GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR;
1021 GstOMXAudioDec *self;
1024 GstBuffer *codec_data = NULL;
1026 GstClockTime timestamp, duration;
1030 self = GST_OMX_AUDIO_DEC (decoder);
1032 GST_DEBUG_OBJECT (self, "Handling frame");
1034 if (self->downstream_flow_ret != GST_FLOW_OK) {
1035 return self->downstream_flow_ret;
1038 if (!self->started) {
1039 GST_DEBUG_OBJECT (self, "Starting task");
1040 gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
1041 (GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL);
1045 return gst_omx_audio_dec_drain (self);
1047 /* Make sure to keep a reference to the input here,
1048 * it can be unreffed from the other thread if
1049 * finish_frame() is called */
1050 gst_buffer_ref (inbuf);
1052 timestamp = GST_BUFFER_TIMESTAMP (inbuf);
1053 duration = GST_BUFFER_DURATION (inbuf);
1055 port = self->dec_in_port;
1057 gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
1059 while (offset < minfo.size) {
1060 /* Make sure to release the base class stream lock, otherwise
1061 * _loop() can't call _finish_frame() and we might block forever
1062 * because no input buffers are released */
1063 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
1064 acq_ret = gst_omx_port_acquire_buffer (port, &buf);
1066 if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) {
1067 GST_AUDIO_DECODER_STREAM_LOCK (self);
1068 goto component_error;
1069 } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
1070 GST_AUDIO_DECODER_STREAM_LOCK (self);
1072 } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
1073 /* Reallocate all buffers */
1074 err = gst_omx_port_set_enabled (port, FALSE);
1075 if (err != OMX_ErrorNone) {
1076 GST_AUDIO_DECODER_STREAM_LOCK (self);
1077 goto reconfigure_error;
1080 err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
1081 if (err != OMX_ErrorNone) {
1082 GST_AUDIO_DECODER_STREAM_LOCK (self);
1083 goto reconfigure_error;
1086 err = gst_omx_port_deallocate_buffers (port);
1087 if (err != OMX_ErrorNone) {
1088 GST_AUDIO_DECODER_STREAM_LOCK (self);
1089 goto reconfigure_error;
1092 err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
1093 if (err != OMX_ErrorNone) {
1094 GST_AUDIO_DECODER_STREAM_LOCK (self);
1095 goto reconfigure_error;
1098 err = gst_omx_port_set_enabled (port, TRUE);
1099 if (err != OMX_ErrorNone) {
1100 GST_AUDIO_DECODER_STREAM_LOCK (self);
1101 goto reconfigure_error;
1104 err = gst_omx_port_allocate_buffers (port);
1105 if (err != OMX_ErrorNone) {
1106 GST_AUDIO_DECODER_STREAM_LOCK (self);
1107 goto reconfigure_error;
1110 err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
1111 if (err != OMX_ErrorNone) {
1112 GST_AUDIO_DECODER_STREAM_LOCK (self);
1113 goto reconfigure_error;
1116 err = gst_omx_port_mark_reconfigured (port);
1117 if (err != OMX_ErrorNone) {
1118 GST_AUDIO_DECODER_STREAM_LOCK (self);
1119 goto reconfigure_error;
1122 /* Now get a new buffer and fill it */
1123 GST_AUDIO_DECODER_STREAM_LOCK (self);
1126 GST_AUDIO_DECODER_STREAM_LOCK (self);
1128 g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
1130 if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) {
1131 gst_omx_port_release_buffer (port, buf);
1135 if (self->downstream_flow_ret != GST_FLOW_OK) {
1136 gst_omx_port_release_buffer (port, buf);
1140 if (self->codec_data) {
1141 GST_DEBUG_OBJECT (self, "Passing codec data to the component");
1143 codec_data = self->codec_data;
1145 if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <
1146 gst_buffer_get_size (codec_data)) {
1147 gst_omx_port_release_buffer (port, buf);
1148 goto too_large_codec_data;
1151 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_CODECCONFIG;
1152 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
1153 buf->omx_buf->nFilledLen = gst_buffer_get_size (codec_data);
1154 gst_buffer_extract (codec_data, 0,
1155 buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
1156 buf->omx_buf->nFilledLen);
1158 if (GST_CLOCK_TIME_IS_VALID (timestamp))
1159 buf->omx_buf->nTimeStamp =
1160 gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
1162 buf->omx_buf->nTimeStamp = 0;
1163 buf->omx_buf->nTickCount = 0;
1165 self->started = TRUE;
1166 err = gst_omx_port_release_buffer (port, buf);
1167 gst_buffer_replace (&self->codec_data, NULL);
1168 if (err != OMX_ErrorNone)
1170 /* Acquire new buffer for the actual frame */
1174 /* Now handle the frame */
1175 GST_DEBUG_OBJECT (self, "Passing frame offset %d to the component", offset);
1177 /* Copy the buffer content in chunks of size as requested
1179 buf->omx_buf->nFilledLen =
1180 MIN (minfo.size - offset,
1181 buf->omx_buf->nAllocLen - buf->omx_buf->nOffset);
1182 gst_buffer_extract (inbuf, offset,
1183 buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
1184 buf->omx_buf->nFilledLen);
1186 if (timestamp != GST_CLOCK_TIME_NONE) {
1187 buf->omx_buf->nTimeStamp =
1188 gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
1189 self->last_upstream_ts = timestamp;
1191 buf->omx_buf->nTimeStamp = 0;
1194 if (duration != GST_CLOCK_TIME_NONE && offset == 0) {
1195 buf->omx_buf->nTickCount =
1196 gst_util_uint64_scale (duration, OMX_TICKS_PER_SECOND, GST_SECOND);
1197 self->last_upstream_ts += duration;
1199 buf->omx_buf->nTickCount = 0;
1203 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_SYNCFRAME;
1206 * - OMX_BUFFERFLAG_DECODEONLY for buffers that are outside
1210 offset += buf->omx_buf->nFilledLen;
1212 if (offset == minfo.size)
1213 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
1215 self->started = TRUE;
1216 err = gst_omx_port_release_buffer (port, buf);
1217 if (err != OMX_ErrorNone)
1220 gst_buffer_unmap (inbuf, &minfo);
1221 gst_buffer_unref (inbuf);
1223 GST_DEBUG_OBJECT (self, "Passed frame to component");
1225 return self->downstream_flow_ret;
1229 gst_buffer_unmap (inbuf, &minfo);
1230 gst_buffer_unref (inbuf);
1232 GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
1233 ("Got OpenMAX buffer with no free space (%p, %u/%u)", buf,
1234 (guint) buf->omx_buf->nOffset, (guint) buf->omx_buf->nAllocLen));
1235 return GST_FLOW_ERROR;
1240 gst_buffer_unmap (inbuf, &minfo);
1241 gst_buffer_unref (inbuf);
1243 return self->downstream_flow_ret;
1246 too_large_codec_data:
1248 gst_buffer_unmap (inbuf, &minfo);
1249 gst_buffer_unref (inbuf);
1251 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
1252 ("codec_data larger than supported by OpenMAX port "
1253 "(%" G_GSIZE_FORMAT " > %u)", gst_buffer_get_size (codec_data),
1254 (guint) self->dec_in_port->port_def.nBufferSize));
1255 return GST_FLOW_ERROR;
1260 gst_buffer_unmap (inbuf, &minfo);
1261 gst_buffer_unref (inbuf);
1263 GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
1264 ("OpenMAX component in error state %s (0x%08x)",
1265 gst_omx_component_get_last_error_string (self->dec),
1266 gst_omx_component_get_last_error (self->dec)));
1267 return GST_FLOW_ERROR;
1272 gst_buffer_unmap (inbuf, &minfo);
1273 gst_buffer_unref (inbuf);
1275 GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
1276 return GST_FLOW_FLUSHING;
1280 gst_buffer_unmap (inbuf, &minfo);
1281 gst_buffer_unref (inbuf);
1283 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
1284 ("Unable to reconfigure input port"));
1285 return GST_FLOW_ERROR;
1289 gst_buffer_unmap (inbuf, &minfo);
1290 gst_buffer_unref (inbuf);
1292 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
1293 ("Failed to relase input buffer to component: %s (0x%08x)",
1294 gst_omx_error_to_string (err), err));
1296 return GST_FLOW_ERROR;
1300 static GstFlowReturn
1301 gst_omx_audio_dec_drain (GstOMXAudioDec * self)
1303 GstOMXAudioDecClass *klass;
1305 GstOMXAcquireBufferReturn acq_ret;
1308 GST_DEBUG_OBJECT (self, "Draining component");
1310 klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
1312 if (!self->started) {
1313 GST_DEBUG_OBJECT (self, "Component not started yet");
1316 self->started = FALSE;
1318 if ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
1319 GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
1323 /* Make sure to release the base class stream lock, otherwise
1324 * _loop() can't call _finish_frame() and we might block forever
1325 * because no input buffers are released */
1326 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
1328 /* Send an EOS buffer to the component and let the base
1329 * class drop the EOS event. We will send it later when
1330 * the EOS buffer arrives on the output port. */
1331 acq_ret = gst_omx_port_acquire_buffer (self->dec_in_port, &buf);
1332 if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) {
1333 GST_AUDIO_DECODER_STREAM_LOCK (self);
1334 GST_ERROR_OBJECT (self, "Failed to acquire buffer for draining: %d",
1336 return GST_FLOW_ERROR;
1339 g_mutex_lock (&self->drain_lock);
1340 self->draining = TRUE;
1341 buf->omx_buf->nFilledLen = 0;
1342 buf->omx_buf->nTimeStamp =
1343 gst_util_uint64_scale (self->last_upstream_ts, OMX_TICKS_PER_SECOND,
1345 buf->omx_buf->nTickCount = 0;
1346 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_EOS;
1347 err = gst_omx_port_release_buffer (self->dec_in_port, buf);
1348 if (err != OMX_ErrorNone) {
1349 GST_ERROR_OBJECT (self, "Failed to drain component: %s (0x%08x)",
1350 gst_omx_error_to_string (err), err);
1351 g_mutex_unlock (&self->drain_lock);
1352 GST_AUDIO_DECODER_STREAM_LOCK (self);
1353 return GST_FLOW_ERROR;
1356 GST_DEBUG_OBJECT (self, "Waiting until component is drained");
1358 if (G_UNLIKELY (self->dec->hacks & GST_OMX_HACK_DRAIN_MAY_NOT_RETURN)) {
1359 gint64 wait_until = g_get_monotonic_time () + G_TIME_SPAN_SECOND / 2;
1361 if (!g_cond_wait_until (&self->drain_cond, &self->drain_lock, wait_until))
1362 GST_WARNING_OBJECT (self, "Drain timed out");
1364 GST_DEBUG_OBJECT (self, "Drained component");
1367 g_cond_wait (&self->drain_cond, &self->drain_lock);
1368 GST_DEBUG_OBJECT (self, "Drained component");
1371 g_mutex_unlock (&self->drain_lock);
1372 GST_AUDIO_DECODER_STREAM_LOCK (self);
1374 gst_adapter_flush (self->output_adapter,
1375 gst_adapter_available (self->output_adapter));
1376 self->started = FALSE;