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 g_mutex_lock (&self->drain_lock);
603 if (self->draining) {
604 self->draining = FALSE;
605 g_cond_broadcast (&self->drain_cond);
607 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
608 self->downstream_flow_ret = GST_FLOW_FLUSHING;
609 self->started = FALSE;
610 g_mutex_unlock (&self->drain_lock);
616 spf = klass->get_samples_per_frame (self, self->dec_out_port);
619 guint avail = gst_adapter_available (self->output_adapter);
622 /* On EOS we take the complete adapter content, no matter
623 * if it is a multiple of the codec frame size or not.
625 avail /= self->info.bpf;
626 nframes = (avail + spf - 1) / spf;
627 avail *= self->info.bpf;
630 outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
632 gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
637 g_mutex_lock (&self->drain_lock);
638 if (self->draining) {
639 GST_DEBUG_OBJECT (self, "Drained");
640 self->draining = FALSE;
641 g_cond_broadcast (&self->drain_cond);
642 flow_ret = GST_FLOW_OK;
643 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
645 GST_DEBUG_OBJECT (self, "Component signalled EOS");
646 flow_ret = GST_FLOW_EOS;
648 g_mutex_unlock (&self->drain_lock);
650 GST_AUDIO_DECODER_STREAM_LOCK (self);
651 self->downstream_flow_ret = flow_ret;
653 /* Here we fallback and pause the task for the EOS case */
654 if (flow_ret != GST_FLOW_OK)
657 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
664 if (flow_ret == GST_FLOW_EOS) {
665 GST_DEBUG_OBJECT (self, "EOS");
667 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
668 gst_event_new_eos ());
669 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
670 self->started = FALSE;
671 } else if (flow_ret < GST_FLOW_EOS) {
672 GST_ELEMENT_ERROR (self, STREAM, FAILED,
673 ("Internal data stream error."), ("stream stopped, reason %s",
674 gst_flow_get_name (flow_ret)));
676 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
677 gst_event_new_eos ());
678 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
679 self->started = FALSE;
680 } else if (flow_ret == GST_FLOW_FLUSHING) {
681 GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
682 g_mutex_lock (&self->drain_lock);
683 if (self->draining) {
684 self->draining = FALSE;
685 g_cond_broadcast (&self->drain_cond);
687 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
688 self->started = FALSE;
689 g_mutex_unlock (&self->drain_lock);
691 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
697 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
698 ("Unable to reconfigure output port"));
699 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
700 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
701 self->downstream_flow_ret = GST_FLOW_ERROR;
702 self->started = FALSE;
708 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
709 ("Invalid sized input buffer"));
710 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
711 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
712 self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
713 self->started = FALSE;
714 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
720 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
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 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
724 self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
725 self->started = FALSE;
730 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
731 ("Failed to relase output buffer to component: %s (0x%08x)",
732 gst_omx_error_to_string (err), err));
733 gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
734 gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
735 self->downstream_flow_ret = GST_FLOW_ERROR;
736 self->started = FALSE;
737 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
743 gst_omx_audio_dec_start (GstAudioDecoder * decoder)
745 GstOMXAudioDec *self;
747 self = GST_OMX_AUDIO_DEC (decoder);
749 self->last_upstream_ts = 0;
750 self->downstream_flow_ret = GST_FLOW_OK;
756 gst_omx_audio_dec_stop (GstAudioDecoder * decoder)
758 GstOMXAudioDec *self;
760 self = GST_OMX_AUDIO_DEC (decoder);
762 GST_DEBUG_OBJECT (self, "Stopping decoder");
764 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
765 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
767 gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
769 if (gst_omx_component_get_state (self->dec, 0) > OMX_StateIdle)
770 gst_omx_component_set_state (self->dec, OMX_StateIdle);
772 self->downstream_flow_ret = GST_FLOW_FLUSHING;
773 self->started = FALSE;
775 g_mutex_lock (&self->drain_lock);
776 self->draining = FALSE;
777 g_cond_broadcast (&self->drain_cond);
778 g_mutex_unlock (&self->drain_lock);
780 gst_adapter_flush (self->output_adapter,
781 gst_adapter_available (self->output_adapter));
783 gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
785 gst_buffer_replace (&self->codec_data, NULL);
787 GST_DEBUG_OBJECT (self, "Stopped decoder");
793 gst_omx_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
795 GstOMXAudioDec *self;
796 GstOMXAudioDecClass *klass;
798 const GValue *codec_data;
799 gboolean is_format_change = FALSE;
800 gboolean needs_disable = FALSE;
802 self = GST_OMX_AUDIO_DEC (decoder);
803 klass = GST_OMX_AUDIO_DEC_GET_CLASS (decoder);
805 GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
807 /* Check if the caps change is a real format change or if only irrelevant
808 * parts of the caps have changed or nothing at all.
810 if (klass->is_format_change)
811 is_format_change = klass->is_format_change (self, self->dec_in_port, caps);
814 gst_omx_component_get_state (self->dec,
815 GST_CLOCK_TIME_NONE) != OMX_StateLoaded;
816 /* If the component is not in Loaded state and a real format change happens
817 * we have to disable the port and re-allocate all buffers. If no real
818 * format change happened we can just exit here.
820 if (needs_disable && !is_format_change) {
821 GST_DEBUG_OBJECT (self,
822 "Already running and caps did not change the format");
826 if (needs_disable && is_format_change) {
827 GstOMXPort *out_port = self->dec_out_port;
829 GST_DEBUG_OBJECT (self, "Need to disable and drain decoder");
831 gst_omx_audio_dec_drain (self);
832 gst_omx_audio_dec_flush (decoder, FALSE);
833 gst_omx_port_set_flushing (out_port, 5 * GST_SECOND, TRUE);
835 if (klass->cdata.hacks & GST_OMX_HACK_NO_COMPONENT_RECONFIGURE) {
836 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
837 gst_omx_audio_dec_stop (GST_AUDIO_DECODER (self));
838 gst_omx_audio_dec_close (GST_AUDIO_DECODER (self));
839 GST_AUDIO_DECODER_STREAM_LOCK (self);
841 if (!gst_omx_audio_dec_open (GST_AUDIO_DECODER (self)))
843 needs_disable = FALSE;
845 if (gst_omx_port_set_enabled (self->dec_in_port, FALSE) != OMX_ErrorNone)
847 if (gst_omx_port_set_enabled (out_port, FALSE) != OMX_ErrorNone)
849 if (gst_omx_port_wait_buffers_released (self->dec_in_port,
850 5 * GST_SECOND) != OMX_ErrorNone)
852 if (gst_omx_port_wait_buffers_released (out_port,
853 1 * GST_SECOND) != OMX_ErrorNone)
855 if (gst_omx_port_deallocate_buffers (self->dec_in_port) != OMX_ErrorNone)
857 if (gst_omx_port_deallocate_buffers (self->dec_out_port) != OMX_ErrorNone)
859 if (gst_omx_port_wait_enabled (self->dec_in_port,
860 1 * GST_SECOND) != OMX_ErrorNone)
862 if (gst_omx_port_wait_enabled (out_port, 1 * GST_SECOND) != OMX_ErrorNone)
866 GST_DEBUG_OBJECT (self, "Decoder drained and disabled");
869 if (klass->set_format) {
870 if (!klass->set_format (self, self->dec_in_port, caps)) {
871 GST_ERROR_OBJECT (self, "Subclass failed to set the new format");
876 GST_DEBUG_OBJECT (self, "Updating outport port definition");
877 if (gst_omx_port_update_port_definition (self->dec_out_port,
878 NULL) != OMX_ErrorNone)
881 /* Get codec data from caps */
882 gst_buffer_replace (&self->codec_data, NULL);
883 s = gst_caps_get_structure (caps, 0);
884 codec_data = gst_structure_get_value (s, "codec_data");
886 /* Vorbis and some other codecs have multiple buffers in
887 * the stream-header field */
888 self->codec_data = gst_value_get_buffer (codec_data);
889 if (self->codec_data)
890 gst_buffer_ref (self->codec_data);
893 GST_DEBUG_OBJECT (self, "Enabling component");
896 if (gst_omx_port_set_enabled (self->dec_in_port, TRUE) != OMX_ErrorNone)
898 if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
901 if ((klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
902 if (gst_omx_port_set_enabled (self->dec_out_port, TRUE) != OMX_ErrorNone)
904 if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone)
907 if (gst_omx_port_wait_enabled (self->dec_out_port,
908 5 * GST_SECOND) != OMX_ErrorNone)
912 if (gst_omx_port_wait_enabled (self->dec_in_port,
913 5 * GST_SECOND) != OMX_ErrorNone)
915 if (gst_omx_port_mark_reconfigured (self->dec_in_port) != OMX_ErrorNone)
918 if (!(klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
919 /* Disable output port */
920 if (gst_omx_port_set_enabled (self->dec_out_port, FALSE) != OMX_ErrorNone)
923 if (gst_omx_port_wait_enabled (self->dec_out_port,
924 1 * GST_SECOND) != OMX_ErrorNone)
927 if (gst_omx_component_set_state (self->dec,
928 OMX_StateIdle) != OMX_ErrorNone)
931 /* Need to allocate buffers to reach Idle state */
932 if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
935 if (gst_omx_component_set_state (self->dec,
936 OMX_StateIdle) != OMX_ErrorNone)
939 /* Need to allocate buffers to reach Idle state */
940 if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
942 if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone)
946 if (gst_omx_component_get_state (self->dec,
947 GST_CLOCK_TIME_NONE) != OMX_StateIdle)
950 if (gst_omx_component_set_state (self->dec,
951 OMX_StateExecuting) != OMX_ErrorNone)
954 if (gst_omx_component_get_state (self->dec,
955 GST_CLOCK_TIME_NONE) != OMX_StateExecuting)
959 /* Unset flushing to allow ports to accept data again */
960 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
961 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
963 if (gst_omx_component_get_last_error (self->dec) != OMX_ErrorNone) {
964 GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)",
965 gst_omx_component_get_last_error_string (self->dec),
966 gst_omx_component_get_last_error (self->dec));
970 self->downstream_flow_ret = GST_FLOW_OK;
976 gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
978 GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
979 OMX_ERRORTYPE err = OMX_ErrorNone;
981 GST_DEBUG_OBJECT (self, "Flushing decoder");
983 if (gst_omx_component_get_state (self->dec, 0) == OMX_StateLoaded)
986 /* 0) Pause the components */
987 if (gst_omx_component_get_state (self->dec, 0) == OMX_StateExecuting) {
988 gst_omx_component_set_state (self->dec, OMX_StatePause);
989 gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
992 /* 1) Wait until the srcpad loop is stopped,
993 * unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
994 * caused by using this lock from inside the loop function */
995 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
996 gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
997 GST_DEBUG_OBJECT (self, "Flushing -- task stopped");
998 GST_AUDIO_DECODER_STREAM_LOCK (self);
1000 /* 2) Flush the ports */
1001 GST_DEBUG_OBJECT (self, "flushing ports");
1002 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
1003 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
1005 /* 3) Resume components */
1006 gst_omx_component_set_state (self->dec, OMX_StateExecuting);
1007 gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
1009 /* 4) Unset flushing to allow ports to accept data again */
1010 gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
1011 gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
1013 err = gst_omx_port_populate (self->dec_out_port);
1015 if (err != OMX_ErrorNone) {
1016 GST_WARNING_OBJECT (self, "Failed to populate output port: %s (0x%08x)",
1017 gst_omx_error_to_string (err), err);
1020 /* Reset our state */
1021 gst_adapter_flush (self->output_adapter,
1022 gst_adapter_available (self->output_adapter));
1023 self->last_upstream_ts = 0;
1024 self->downstream_flow_ret = GST_FLOW_OK;
1025 self->started = FALSE;
1026 GST_DEBUG_OBJECT (self, "Flush finished");
1029 static GstFlowReturn
1030 gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
1032 GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR;
1033 GstOMXAudioDec *self;
1036 GstBuffer *codec_data = NULL;
1038 GstClockTime timestamp, duration;
1042 self = GST_OMX_AUDIO_DEC (decoder);
1044 GST_DEBUG_OBJECT (self, "Handling frame");
1046 if (self->downstream_flow_ret != GST_FLOW_OK) {
1047 return self->downstream_flow_ret;
1050 if (!self->started) {
1051 GST_DEBUG_OBJECT (self, "Starting task");
1052 gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
1053 (GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL);
1057 return gst_omx_audio_dec_drain (self);
1059 /* Make sure to keep a reference to the input here,
1060 * it can be unreffed from the other thread if
1061 * finish_frame() is called */
1062 gst_buffer_ref (inbuf);
1064 timestamp = GST_BUFFER_TIMESTAMP (inbuf);
1065 duration = GST_BUFFER_DURATION (inbuf);
1067 port = self->dec_in_port;
1069 gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
1071 while (offset < minfo.size) {
1072 /* Make sure to release the base class stream lock, otherwise
1073 * _loop() can't call _finish_frame() and we might block forever
1074 * because no input buffers are released */
1075 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
1076 acq_ret = gst_omx_port_acquire_buffer (port, &buf);
1078 if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) {
1079 GST_AUDIO_DECODER_STREAM_LOCK (self);
1080 goto component_error;
1081 } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
1082 GST_AUDIO_DECODER_STREAM_LOCK (self);
1084 } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
1085 /* Reallocate all buffers */
1086 err = gst_omx_port_set_enabled (port, FALSE);
1087 if (err != OMX_ErrorNone) {
1088 GST_AUDIO_DECODER_STREAM_LOCK (self);
1089 goto reconfigure_error;
1092 err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
1093 if (err != OMX_ErrorNone) {
1094 GST_AUDIO_DECODER_STREAM_LOCK (self);
1095 goto reconfigure_error;
1098 err = gst_omx_port_deallocate_buffers (port);
1099 if (err != OMX_ErrorNone) {
1100 GST_AUDIO_DECODER_STREAM_LOCK (self);
1101 goto reconfigure_error;
1104 err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
1105 if (err != OMX_ErrorNone) {
1106 GST_AUDIO_DECODER_STREAM_LOCK (self);
1107 goto reconfigure_error;
1110 err = gst_omx_port_set_enabled (port, TRUE);
1111 if (err != OMX_ErrorNone) {
1112 GST_AUDIO_DECODER_STREAM_LOCK (self);
1113 goto reconfigure_error;
1116 err = gst_omx_port_allocate_buffers (port);
1117 if (err != OMX_ErrorNone) {
1118 GST_AUDIO_DECODER_STREAM_LOCK (self);
1119 goto reconfigure_error;
1122 err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
1123 if (err != OMX_ErrorNone) {
1124 GST_AUDIO_DECODER_STREAM_LOCK (self);
1125 goto reconfigure_error;
1128 err = gst_omx_port_mark_reconfigured (port);
1129 if (err != OMX_ErrorNone) {
1130 GST_AUDIO_DECODER_STREAM_LOCK (self);
1131 goto reconfigure_error;
1134 /* Now get a new buffer and fill it */
1135 GST_AUDIO_DECODER_STREAM_LOCK (self);
1138 GST_AUDIO_DECODER_STREAM_LOCK (self);
1140 g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
1142 if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) {
1143 gst_omx_port_release_buffer (port, buf);
1147 if (self->downstream_flow_ret != GST_FLOW_OK) {
1148 gst_omx_port_release_buffer (port, buf);
1152 if (self->codec_data) {
1153 GST_DEBUG_OBJECT (self, "Passing codec data to the component");
1155 codec_data = self->codec_data;
1157 if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <
1158 gst_buffer_get_size (codec_data)) {
1159 gst_omx_port_release_buffer (port, buf);
1160 goto too_large_codec_data;
1163 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_CODECCONFIG;
1164 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
1165 buf->omx_buf->nFilledLen = gst_buffer_get_size (codec_data);
1166 gst_buffer_extract (codec_data, 0,
1167 buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
1168 buf->omx_buf->nFilledLen);
1170 if (GST_CLOCK_TIME_IS_VALID (timestamp))
1171 buf->omx_buf->nTimeStamp =
1172 gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
1174 buf->omx_buf->nTimeStamp = 0;
1175 buf->omx_buf->nTickCount = 0;
1177 self->started = TRUE;
1178 err = gst_omx_port_release_buffer (port, buf);
1179 gst_buffer_replace (&self->codec_data, NULL);
1180 if (err != OMX_ErrorNone)
1182 /* Acquire new buffer for the actual frame */
1186 /* Now handle the frame */
1187 GST_DEBUG_OBJECT (self, "Passing frame offset %d to the component", offset);
1189 /* Copy the buffer content in chunks of size as requested
1191 buf->omx_buf->nFilledLen =
1192 MIN (minfo.size - offset,
1193 buf->omx_buf->nAllocLen - buf->omx_buf->nOffset);
1194 gst_buffer_extract (inbuf, offset,
1195 buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
1196 buf->omx_buf->nFilledLen);
1198 if (timestamp != GST_CLOCK_TIME_NONE) {
1199 buf->omx_buf->nTimeStamp =
1200 gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
1201 self->last_upstream_ts = timestamp;
1203 buf->omx_buf->nTimeStamp = 0;
1206 if (duration != GST_CLOCK_TIME_NONE && offset == 0) {
1207 buf->omx_buf->nTickCount =
1208 gst_util_uint64_scale (duration, OMX_TICKS_PER_SECOND, GST_SECOND);
1209 self->last_upstream_ts += duration;
1211 buf->omx_buf->nTickCount = 0;
1215 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_SYNCFRAME;
1218 * - OMX_BUFFERFLAG_DECODEONLY for buffers that are outside
1222 offset += buf->omx_buf->nFilledLen;
1224 if (offset == minfo.size)
1225 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
1227 self->started = TRUE;
1228 err = gst_omx_port_release_buffer (port, buf);
1229 if (err != OMX_ErrorNone)
1232 gst_buffer_unmap (inbuf, &minfo);
1233 gst_buffer_unref (inbuf);
1235 GST_DEBUG_OBJECT (self, "Passed frame to component");
1237 return self->downstream_flow_ret;
1241 gst_buffer_unmap (inbuf, &minfo);
1242 gst_buffer_unref (inbuf);
1244 GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
1245 ("Got OpenMAX buffer with no free space (%p, %u/%u)", buf,
1246 (guint) buf->omx_buf->nOffset, (guint) buf->omx_buf->nAllocLen));
1247 return GST_FLOW_ERROR;
1252 gst_buffer_unmap (inbuf, &minfo);
1253 gst_buffer_unref (inbuf);
1255 return self->downstream_flow_ret;
1258 too_large_codec_data:
1260 gst_buffer_unmap (inbuf, &minfo);
1261 gst_buffer_unref (inbuf);
1263 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
1264 ("codec_data larger than supported by OpenMAX port "
1265 "(%" G_GSIZE_FORMAT " > %u)", gst_buffer_get_size (codec_data),
1266 (guint) self->dec_in_port->port_def.nBufferSize));
1267 return GST_FLOW_ERROR;
1272 gst_buffer_unmap (inbuf, &minfo);
1273 gst_buffer_unref (inbuf);
1275 GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
1276 ("OpenMAX component in error state %s (0x%08x)",
1277 gst_omx_component_get_last_error_string (self->dec),
1278 gst_omx_component_get_last_error (self->dec)));
1279 return GST_FLOW_ERROR;
1284 gst_buffer_unmap (inbuf, &minfo);
1285 gst_buffer_unref (inbuf);
1287 GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
1288 return GST_FLOW_FLUSHING;
1292 gst_buffer_unmap (inbuf, &minfo);
1293 gst_buffer_unref (inbuf);
1295 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
1296 ("Unable to reconfigure input port"));
1297 return GST_FLOW_ERROR;
1301 gst_buffer_unmap (inbuf, &minfo);
1302 gst_buffer_unref (inbuf);
1304 GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
1305 ("Failed to relase input buffer to component: %s (0x%08x)",
1306 gst_omx_error_to_string (err), err));
1308 return GST_FLOW_ERROR;
1312 static GstFlowReturn
1313 gst_omx_audio_dec_drain (GstOMXAudioDec * self)
1315 GstOMXAudioDecClass *klass;
1317 GstOMXAcquireBufferReturn acq_ret;
1320 GST_DEBUG_OBJECT (self, "Draining component");
1322 klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
1324 if (!self->started) {
1325 GST_DEBUG_OBJECT (self, "Component not started yet");
1328 self->started = FALSE;
1330 if ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
1331 GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
1335 /* Make sure to release the base class stream lock, otherwise
1336 * _loop() can't call _finish_frame() and we might block forever
1337 * because no input buffers are released */
1338 GST_AUDIO_DECODER_STREAM_UNLOCK (self);
1340 /* Send an EOS buffer to the component and let the base
1341 * class drop the EOS event. We will send it later when
1342 * the EOS buffer arrives on the output port. */
1343 acq_ret = gst_omx_port_acquire_buffer (self->dec_in_port, &buf);
1344 if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) {
1345 GST_AUDIO_DECODER_STREAM_LOCK (self);
1346 GST_ERROR_OBJECT (self, "Failed to acquire buffer for draining: %d",
1348 return GST_FLOW_ERROR;
1351 g_mutex_lock (&self->drain_lock);
1352 self->draining = TRUE;
1353 buf->omx_buf->nFilledLen = 0;
1354 buf->omx_buf->nTimeStamp =
1355 gst_util_uint64_scale (self->last_upstream_ts, OMX_TICKS_PER_SECOND,
1357 buf->omx_buf->nTickCount = 0;
1358 buf->omx_buf->nFlags |= OMX_BUFFERFLAG_EOS;
1359 err = gst_omx_port_release_buffer (self->dec_in_port, buf);
1360 if (err != OMX_ErrorNone) {
1361 GST_ERROR_OBJECT (self, "Failed to drain component: %s (0x%08x)",
1362 gst_omx_error_to_string (err), err);
1363 g_mutex_unlock (&self->drain_lock);
1364 GST_AUDIO_DECODER_STREAM_LOCK (self);
1365 return GST_FLOW_ERROR;
1368 GST_DEBUG_OBJECT (self, "Waiting until component is drained");
1370 if (G_UNLIKELY (self->dec->hacks & GST_OMX_HACK_DRAIN_MAY_NOT_RETURN)) {
1371 gint64 wait_until = g_get_monotonic_time () + G_TIME_SPAN_SECOND / 2;
1373 if (!g_cond_wait_until (&self->drain_cond, &self->drain_lock, wait_until))
1374 GST_WARNING_OBJECT (self, "Drain timed out");
1376 GST_DEBUG_OBJECT (self, "Drained component");
1379 g_cond_wait (&self->drain_cond, &self->drain_lock);
1380 GST_DEBUG_OBJECT (self, "Drained component");
1383 g_mutex_unlock (&self->drain_lock);
1384 GST_AUDIO_DECODER_STREAM_LOCK (self);
1386 gst_adapter_flush (self->output_adapter,
1387 gst_adapter_available (self->output_adapter));
1388 self->started = FALSE;