mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-24 02:31:03 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-omx/omx/gstomxaudiodec.c

1404 lines
44 KiB
C
Raw Blame History

/*
* Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
* Copyright (C) 2013, Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
* Copyright (C) 2014, Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation
* version 2.1 of the License.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <string.h>
#include "gstomxaudiodec.h"
GST_DEBUG_CATEGORY_STATIC (gst_omx_audio_dec_debug_category);
#define GST_CAT_DEFAULT gst_omx_audio_dec_debug_category
/* prototypes */
static void gst_omx_audio_dec_finalize (GObject * object);
static GstStateChangeReturn
gst_omx_audio_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_omx_audio_dec_open (GstAudioDecoder * decoder);
static gboolean gst_omx_audio_dec_close (GstAudioDecoder * decoder);
static gboolean gst_omx_audio_dec_start (GstAudioDecoder * decoder);
static gboolean gst_omx_audio_dec_stop (GstAudioDecoder * decoder);
static gboolean gst_omx_audio_dec_set_format (GstAudioDecoder * decoder,
GstCaps * caps);
static void gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard);
static GstFlowReturn gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder,
GstBuffer * buffer);
static GstFlowReturn gst_omx_audio_dec_drain (GstOMXAudioDec * self);
enum
{
PROP_0
};
/* class initialization */
#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (gst_omx_audio_dec_debug_category, "omxaudiodec", 0, \
"debug category for gst-omx audio decoder base class");
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstOMXAudioDec, gst_omx_audio_dec,
GST_TYPE_AUDIO_DECODER, DEBUG_INIT);
static void
gst_omx_audio_dec_class_init (GstOMXAudioDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass);
gobject_class->finalize = gst_omx_audio_dec_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_omx_audio_dec_change_state);
audio_decoder_class->open = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_open);
audio_decoder_class->close = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_close);
audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_start);
audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_stop);
audio_decoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_flush);
audio_decoder_class->set_format =
GST_DEBUG_FUNCPTR (gst_omx_audio_dec_set_format);
audio_decoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_omx_audio_dec_handle_frame);
klass->cdata.type = GST_OMX_COMPONENT_TYPE_FILTER;
klass->cdata.default_src_template_caps =
"audio/x-raw, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, " G_STRINGIFY (OMX_AUDIO_MAXCHANNELS) " ], "
"format = (string) " GST_AUDIO_FORMATS_ALL;
}
static void
gst_omx_audio_dec_init (GstOMXAudioDec * self)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(self), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (self));
g_mutex_init (&self->drain_lock);
g_cond_init (&self->drain_cond);
self->output_adapter = gst_adapter_new ();
}
static gboolean
gst_omx_audio_dec_open (GstAudioDecoder * decoder)
{
GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
gint in_port_index, out_port_index;
GST_DEBUG_OBJECT (self, "Opening decoder");
self->dec =
gst_omx_component_new (GST_OBJECT_CAST (self), klass->cdata.core_name,
klass->cdata.component_name, klass->cdata.component_role,
klass->cdata.hacks);
self->started = FALSE;
if (!self->dec)
return FALSE;
if (gst_omx_component_get_state (self->dec,
GST_CLOCK_TIME_NONE) != OMX_StateLoaded)
return FALSE;
in_port_index = klass->cdata.in_port_index;
out_port_index = klass->cdata.out_port_index;
if (in_port_index == -1 || out_port_index == -1) {
OMX_PORT_PARAM_TYPE param;
OMX_ERRORTYPE err;
GST_OMX_INIT_STRUCT (&param);
err =
gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioInit,
&param);
if (err != OMX_ErrorNone) {
GST_WARNING_OBJECT (self, "Couldn't get port information: %s (0x%08x)",
gst_omx_error_to_string (err), err);
/* Fallback */
in_port_index = 0;
out_port_index = 1;
} else {
GST_DEBUG_OBJECT (self, "Detected %u ports, starting at %u",
(guint) param.nPorts, (guint) param.nStartPortNumber);
in_port_index = param.nStartPortNumber + 0;
out_port_index = param.nStartPortNumber + 1;
}
}
self->dec_in_port = gst_omx_component_add_port (self->dec, in_port_index);
self->dec_out_port = gst_omx_component_add_port (self->dec, out_port_index);
if (!self->dec_in_port || !self->dec_out_port)
return FALSE;
GST_DEBUG_OBJECT (self, "Opened decoder");
return TRUE;
}
static gboolean
gst_omx_audio_dec_shutdown (GstOMXAudioDec * self)
{
OMX_STATETYPE state;
GST_DEBUG_OBJECT (self, "Shutting down decoder");
state = gst_omx_component_get_state (self->dec, 0);
if (state > OMX_StateLoaded || state == OMX_StateInvalid) {
if (state > OMX_StateIdle) {
gst_omx_component_set_state (self->dec, OMX_StateIdle);
gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
}
gst_omx_component_set_state (self->dec, OMX_StateLoaded);
gst_omx_port_deallocate_buffers (self->dec_in_port);
gst_omx_port_deallocate_buffers (self->dec_out_port);
if (state > OMX_StateLoaded)
gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
}
return TRUE;
}
static gboolean
gst_omx_audio_dec_close (GstAudioDecoder * decoder)
{
GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Closing decoder");
if (!gst_omx_audio_dec_shutdown (self))
return FALSE;
self->dec_in_port = NULL;
self->dec_out_port = NULL;
if (self->dec)
gst_omx_component_unref (self->dec);
self->dec = NULL;
self->started = FALSE;
GST_DEBUG_OBJECT (self, "Closed decoder");
return TRUE;
}
static void
gst_omx_audio_dec_finalize (GObject * object)
{
GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (object);
g_mutex_clear (&self->drain_lock);
g_cond_clear (&self->drain_cond);
if (self->output_adapter)
gst_object_unref (self->output_adapter);
self->output_adapter = NULL;
G_OBJECT_CLASS (gst_omx_audio_dec_parent_class)->finalize (object);
}
static GstStateChangeReturn
gst_omx_audio_dec_change_state (GstElement * element, GstStateChange transition)
{
GstOMXAudioDec *self;
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
g_return_val_if_fail (GST_IS_OMX_AUDIO_DEC (element),
GST_STATE_CHANGE_FAILURE);
self = GST_OMX_AUDIO_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
self->downstream_flow_ret = GST_FLOW_OK;
self->draining = FALSE;
self->started = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (self->dec_in_port)
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
if (self->dec_out_port)
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
break;
default:
break;
}
ret =
GST_ELEMENT_CLASS (gst_omx_audio_dec_parent_class)->change_state
(element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
if (!gst_omx_audio_dec_shutdown (self))
ret = GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
gst_omx_audio_dec_loop (GstOMXAudioDec * self)
{
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
GstOMXPort *port = self->dec_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
gint spf;
acq_return = gst_omx_port_acquire_buffer (port, &buf, GST_OMX_WAIT);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) {
goto component_error;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
goto flushing;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
goto eos;
}
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (self)) ||
acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
OMX_PARAM_PORTDEFINITIONTYPE port_def;
OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
GstAudioChannelPosition omx_position[OMX_AUDIO_MAXCHANNELS];
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
gint i;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
/* Reallocate all buffers */
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE
&& gst_omx_port_is_enabled (port)) {
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Just update caps */
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_omx_port_get_port_definition (port, &port_def);
g_assert (port_def.format.audio.eEncoding == OMX_AUDIO_CodingPCM);
GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = self->dec_out_port->index;
err =
gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioPcm,
&pcm_param);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to get PCM parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
goto caps_failed;
}
g_assert (pcm_param.ePCMMode == OMX_AUDIO_PCMModeLinear);
g_assert (pcm_param.bInterleaved == OMX_TRUE);
gst_audio_info_init (&self->info);
for (i = 0; i < pcm_param.nChannels; i++) {
switch (pcm_param.eChannelMapping[i]) {
case OMX_AUDIO_ChannelLF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
break;
case OMX_AUDIO_ChannelRF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case OMX_AUDIO_ChannelCF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
break;
case OMX_AUDIO_ChannelLS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
break;
case OMX_AUDIO_ChannelRS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
break;
case OMX_AUDIO_ChannelLFE:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
break;
case OMX_AUDIO_ChannelCS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
break;
case OMX_AUDIO_ChannelLR:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
break;
case OMX_AUDIO_ChannelRR:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
case OMX_AUDIO_ChannelNone:
default:
/* This will break the outer loop too as the
* i == pcm_param.nChannels afterwards */
for (i = 0; i < pcm_param.nChannels; i++)
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
break;
}
}
if (pcm_param.nChannels == 1
&& omx_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER)
omx_position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
if (omx_position[0] == GST_AUDIO_CHANNEL_POSITION_NONE
&& klass->get_channel_positions) {
GST_WARNING_OBJECT (self,
"Failed to get a valid channel layout, trying fallback");
klass->get_channel_positions (self, self->dec_out_port, omx_position);
}
memcpy (self->position, omx_position, sizeof (omx_position));
gst_audio_channel_positions_to_valid_order (self->position,
pcm_param.nChannels);
self->needs_reorder =
(memcmp (self->position, omx_position,
sizeof (GstAudioChannelPosition) * pcm_param.nChannels) != 0);
if (self->needs_reorder)
gst_audio_get_channel_reorder_map (pcm_param.nChannels, self->position,
omx_position, self->reorder_map);
gst_audio_info_set_format (&self->info,
gst_audio_format_build_integer (pcm_param.eNumData ==
OMX_NumericalDataSigned,
pcm_param.eEndian ==
OMX_EndianLittle ? G_LITTLE_ENDIAN : G_BIG_ENDIAN,
pcm_param.nBitPerSample, pcm_param.nBitPerSample),
pcm_param.nSamplingRate, pcm_param.nChannels, self->position);
GST_DEBUG_OBJECT (self,
"Setting output state: format %s, rate %u, channels %u",
gst_audio_format_to_string (self->info.finfo->format),
(guint) pcm_param.nSamplingRate, (guint) pcm_param.nChannels);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self),
&self->info)
|| !gst_audio_decoder_negotiate (GST_AUDIO_DECODER (self))) {
if (buf)
gst_omx_port_release_buffer (port, buf);
goto caps_failed;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_populate (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Now get a buffer */
if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) {
return;
}
}
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
if (!buf) {
g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto eos;
}
/* This prevents a deadlock between the srcpad stream
* lock and the audiocodec stream lock, if ::reset()
* is called at the wrong time
*/
if (gst_omx_port_is_flushing (port)) {
GST_DEBUG_OBJECT (self, "Flushing");
gst_omx_port_release_buffer (port, buf);
goto flushing;
}
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
(guint) buf->omx_buf->nFlags,
(guint64) GST_OMX_GET_TICKS (buf->omx_buf->nTimeStamp));
GST_AUDIO_DECODER_STREAM_LOCK (self);
spf = klass->get_samples_per_frame (self, self->dec_out_port);
if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
GstMapInfo minfo;
GST_DEBUG_OBJECT (self, "Handling output data");
if (buf->omx_buf->nFilledLen % self->info.bpf != 0) {
gst_omx_port_release_buffer (port, buf);
goto invalid_buffer;
}
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buf->omx_buf->nFilledLen);
gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE);
if (self->needs_reorder) {
gint i, n_samples, c, n_channels;
gint *reorder_map = self->reorder_map;
gint16 *dest, *source;
dest = (gint16 *) minfo.data;
source = (gint16 *) (buf->omx_buf->pBuffer + buf->omx_buf->nOffset);
n_samples = buf->omx_buf->nFilledLen / self->info.bpf;
n_channels = self->info.channels;
for (i = 0; i < n_samples; i++) {
for (c = 0; c < n_channels; c++) {
dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c];
}
}
} else {
memcpy (minfo.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
}
gst_buffer_unmap (outbuf, &minfo);
if (spf != -1) {
gst_adapter_push (self->output_adapter, outbuf);
} else {
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
}
}
GST_DEBUG_OBJECT (self, "Read frame from component");
if (spf != -1) {
GstBuffer *outbuf;
guint avail = gst_adapter_available (self->output_adapter);
guint nframes;
/* We take a multiple of codec frames and push
* them downstream
*/
avail /= self->info.bpf;
nframes = avail / spf;
avail = nframes * spf;
avail *= self->info.bpf;
if (avail > 0) {
outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
}
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
if (buf) {
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
component_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->dec),
gst_omx_component_get_last_error (self->dec)));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
g_mutex_lock (&self->drain_lock);
if (self->draining) {
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
}
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
g_mutex_unlock (&self->drain_lock);
return;
}
eos:
{
spf = klass->get_samples_per_frame (self, self->dec_out_port);
if (spf != -1) {
GstBuffer *outbuf;
guint avail = gst_adapter_available (self->output_adapter);
guint nframes;
/* On EOS we take the complete adapter content, no matter
* if it is a multiple of the codec frame size or not.
*/
avail /= self->info.bpf;
nframes = (avail + spf - 1) / spf;
avail *= self->info.bpf;
if (avail > 0) {
outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
}
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
flow_ret = GST_FLOW_OK;
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
} else {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
self->downstream_flow_ret = flow_ret;
/* Here we fallback and pause the task for the EOS case */
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
("Internal data stream error."), ("stream stopped, reason %s",
gst_flow_get_name (flow_ret)));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
g_mutex_lock (&self->drain_lock);
if (self->draining) {
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
}
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
g_mutex_unlock (&self->drain_lock);
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
invalid_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Invalid sized input buffer"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
caps_failed:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
release_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase output buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
}
static gboolean
gst_omx_audio_dec_start (GstAudioDecoder * decoder)
{
GstOMXAudioDec *self;
self = GST_OMX_AUDIO_DEC (decoder);
self->last_upstream_ts = 0;
self->downstream_flow_ret = GST_FLOW_OK;
return TRUE;
}
static gboolean
gst_omx_audio_dec_stop (GstAudioDecoder * decoder)
{
GstOMXAudioDec *self;
self = GST_OMX_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Stopping decoder");
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
if (gst_omx_component_get_state (self->dec, 0) > OMX_StateIdle)
gst_omx_component_set_state (self->dec, OMX_StateIdle);
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
gst_buffer_replace (&self->codec_data, NULL);
GST_DEBUG_OBJECT (self, "Stopped decoder");
return TRUE;
}
static gboolean
gst_omx_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstOMXAudioDec *self;
GstOMXAudioDecClass *klass;
GstStructure *s;
const GValue *codec_data;
gboolean is_format_change = FALSE;
gboolean needs_disable = FALSE;
self = GST_OMX_AUDIO_DEC (decoder);
klass = GST_OMX_AUDIO_DEC_GET_CLASS (decoder);
GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
/* Check if the caps change is a real format change or if only irrelevant
* parts of the caps have changed or nothing at all.
*/
if (klass->is_format_change)
is_format_change = klass->is_format_change (self, self->dec_in_port, caps);
needs_disable =
gst_omx_component_get_state (self->dec,
GST_CLOCK_TIME_NONE) != OMX_StateLoaded;
/* If the component is not in Loaded state and a real format change happens
* we have to disable the port and re-allocate all buffers. If no real
* format change happened we can just exit here.
*/
if (needs_disable && !is_format_change) {
GST_DEBUG_OBJECT (self,
"Already running and caps did not change the format");
return TRUE;
}
if (needs_disable && is_format_change) {
GstOMXPort *out_port = self->dec_out_port;
GST_DEBUG_OBJECT (self, "Need to disable and drain decoder");
gst_omx_audio_dec_drain (self);
gst_omx_audio_dec_flush (decoder, FALSE);
gst_omx_port_set_flushing (out_port, 5 * GST_SECOND, TRUE);
if (klass->cdata.hacks & GST_OMX_HACK_NO_COMPONENT_RECONFIGURE) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_omx_audio_dec_stop (GST_AUDIO_DECODER (self));
gst_omx_audio_dec_close (GST_AUDIO_DECODER (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (!gst_omx_audio_dec_open (GST_AUDIO_DECODER (self)))
return FALSE;
needs_disable = FALSE;
} else {
/* Disabling at the same time input port and output port is only
* required when a buffer is shared between the ports. This cannot
* be the case for a decoder because its input and output buffers
* are of different nature. So let's disable ports sequencially.
* Starting from IL 1.2.0, this point has been clarified.
* OMX_SendCommand will return an error if the IL client attempts to
* call it when there is already an on-going command being processed.
* The exception is for buffer sharing above and the event
* OMX_EventPortNeedsDisable will be sent to request disabling the
* other port at the same time. */
if (gst_omx_port_set_enabled (self->dec_in_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (self->dec_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->dec_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->dec_in_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_set_enabled (out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (out_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (out_port, 1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
}
GST_DEBUG_OBJECT (self, "Decoder drained and disabled");
}
if (klass->set_format) {
if (!klass->set_format (self, self->dec_in_port, caps)) {
GST_ERROR_OBJECT (self, "Subclass failed to set the new format");
return FALSE;
}
}
GST_DEBUG_OBJECT (self, "Updating outport port definition");
if (gst_omx_port_update_port_definition (self->dec_out_port,
NULL) != OMX_ErrorNone)
return FALSE;
/* Get codec data from caps */
gst_buffer_replace (&self->codec_data, NULL);
s = gst_caps_get_structure (caps, 0);
codec_data = gst_structure_get_value (s, "codec_data");
if (codec_data) {
/* Vorbis and some other codecs have multiple buffers in
* the stream-header field */
self->codec_data = gst_value_get_buffer (codec_data);
if (self->codec_data)
gst_buffer_ref (self->codec_data);
}
GST_DEBUG_OBJECT (self, "Enabling component");
if (needs_disable) {
if (gst_omx_port_set_enabled (self->dec_in_port, TRUE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
return FALSE;
if ((klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
if (gst_omx_port_set_enabled (self->dec_out_port, TRUE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->dec_out_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
}
if (gst_omx_port_wait_enabled (self->dec_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_mark_reconfigured (self->dec_in_port) != OMX_ErrorNone)
return FALSE;
} else {
if (!(klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
/* Disable output port */
if (gst_omx_port_set_enabled (self->dec_out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->dec_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_set_state (self->dec,
OMX_StateIdle) != OMX_ErrorNone)
return FALSE;
/* Need to allocate buffers to reach Idle state */
if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
return FALSE;
} else {
if (gst_omx_component_set_state (self->dec,
OMX_StateIdle) != OMX_ErrorNone)
return FALSE;
/* Need to allocate buffers to reach Idle state */
if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone)
return FALSE;
}
if (gst_omx_component_get_state (self->dec,
GST_CLOCK_TIME_NONE) != OMX_StateIdle)
return FALSE;
if (gst_omx_component_set_state (self->dec,
OMX_StateExecuting) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_get_state (self->dec,
GST_CLOCK_TIME_NONE) != OMX_StateExecuting)
return FALSE;
}
/* Unset flushing to allow ports to accept data again */
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
if (gst_omx_component_get_last_error (self->dec) != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)",
gst_omx_component_get_last_error_string (self->dec),
gst_omx_component_get_last_error (self->dec));
return FALSE;
}
self->downstream_flow_ret = GST_FLOW_OK;
return TRUE;
}
static void
gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
OMX_ERRORTYPE err = OMX_ErrorNone;
GST_DEBUG_OBJECT (self, "Flushing decoder");
if (gst_omx_component_get_state (self->dec, 0) == OMX_StateLoaded)
return;
/* 0) Pause the components */
if (gst_omx_component_get_state (self->dec, 0) == OMX_StateExecuting) {
gst_omx_component_set_state (self->dec, OMX_StatePause);
gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
}
/* 1) Flush the ports */
GST_DEBUG_OBJECT (self, "flushing ports");
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
/* 2) Wait until the srcpad loop is stopped,
* unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
GST_DEBUG_OBJECT (self, "Flushing -- task stopped");
GST_AUDIO_DECODER_STREAM_LOCK (self);
/* 3) Resume components */
gst_omx_component_set_state (self->dec, OMX_StateExecuting);
gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
/* 4) Unset flushing to allow ports to accept data again */
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
err = gst_omx_port_populate (self->dec_out_port);
if (err != OMX_ErrorNone) {
GST_WARNING_OBJECT (self, "Failed to populate output port: %s (0x%08x)",
gst_omx_error_to_string (err), err);
}
/* Reset our state */
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
self->last_upstream_ts = 0;
self->downstream_flow_ret = GST_FLOW_OK;
self->started = FALSE;
GST_DEBUG_OBJECT (self, "Flush finished");
}
static GstFlowReturn
gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR;
GstOMXAudioDec *self;
GstOMXPort *port;
GstOMXBuffer *buf;
GstBuffer *codec_data = NULL;
guint offset = 0;
GstClockTime timestamp, duration;
OMX_ERRORTYPE err;
GstMapInfo minfo;
self = GST_OMX_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Handling frame");
if (self->downstream_flow_ret != GST_FLOW_OK) {
return self->downstream_flow_ret;
}
if (!self->started) {
GST_DEBUG_OBJECT (self, "Starting task");
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL);
}
if (inbuf == NULL)
return gst_omx_audio_dec_drain (self);
/* Make sure to keep a reference to the input here,
* it can be unreffed from the other thread if
* finish_frame() is called */
gst_buffer_ref (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
port = self->dec_in_port;
gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
while (offset < minfo.size) {
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
acq_ret = gst_omx_port_acquire_buffer (port, &buf, GST_OMX_WAIT);
if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto component_error;
} else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto flushing;
} else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
/* Reallocate all buffers */
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
/* Now get a new buffer and fill it */
GST_AUDIO_DECODER_STREAM_LOCK (self);
continue;
}
GST_AUDIO_DECODER_STREAM_LOCK (self);
g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) {
gst_omx_port_release_buffer (port, buf);
goto full_buffer;
}
if (self->downstream_flow_ret != GST_FLOW_OK) {
gst_omx_port_release_buffer (port, buf);
goto flow_error;
}
if (self->codec_data) {
GST_DEBUG_OBJECT (self, "Passing codec data to the component");
codec_data = self->codec_data;
if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <
gst_buffer_get_size (codec_data)) {
gst_omx_port_release_buffer (port, buf);
goto too_large_codec_data;
}
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_CODECCONFIG;
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
buf->omx_buf->nFilledLen = gst_buffer_get_size (codec_data);
gst_buffer_extract (codec_data, 0,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp,
gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND,
GST_SECOND));
else
GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp, G_GUINT64_CONSTANT (0));
buf->omx_buf->nTickCount = 0;
self->started = TRUE;
err = gst_omx_port_release_buffer (port, buf);
gst_buffer_replace (&self->codec_data, NULL);
if (err != OMX_ErrorNone)
goto release_error;
/* Acquire new buffer for the actual frame */
continue;
}
/* Now handle the frame */
GST_DEBUG_OBJECT (self, "Passing frame offset %d to the component", offset);
/* Copy the buffer content in chunks of size as requested
* by the port */
buf->omx_buf->nFilledLen =
MIN (minfo.size - offset,
buf->omx_buf->nAllocLen - buf->omx_buf->nOffset);
gst_buffer_extract (inbuf, offset,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
if (timestamp != GST_CLOCK_TIME_NONE) {
GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp,
gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND));
self->last_upstream_ts = timestamp;
} else {
GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp, G_GUINT64_CONSTANT (0));
}
if (duration != GST_CLOCK_TIME_NONE && offset == 0) {
buf->omx_buf->nTickCount =
gst_util_uint64_scale (duration, OMX_TICKS_PER_SECOND, GST_SECOND);
self->last_upstream_ts += duration;
} else {
buf->omx_buf->nTickCount = 0;
}
if (offset == 0)
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_SYNCFRAME;
/* TODO: Set flags
* - OMX_BUFFERFLAG_DECODEONLY for buffers that are outside
* the segment
*/
offset += buf->omx_buf->nFilledLen;
if (offset == minfo.size)
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
self->started = TRUE;
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
}
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_DEBUG_OBJECT (self, "Passed frame to component");
return self->downstream_flow_ret;
full_buffer:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Got OpenMAX buffer with no free space (%p, %u/%u)", buf,
(guint) buf->omx_buf->nOffset, (guint) buf->omx_buf->nAllocLen));
return GST_FLOW_ERROR;
}
flow_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
}
too_large_codec_data:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
("codec_data larger than supported by OpenMAX port "
"(%" G_GSIZE_FORMAT " > %u)", gst_buffer_get_size (codec_data),
(guint) self->dec_in_port->port_def.nBufferSize));
return GST_FLOW_ERROR;
}
component_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->dec),
gst_omx_component_get_last_error (self->dec)));
return GST_FLOW_ERROR;
}
flushing:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
return GST_FLOW_FLUSHING;
}
reconfigure_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure input port"));
return GST_FLOW_ERROR;
}
release_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase input buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_omx_audio_dec_drain (GstOMXAudioDec * self)
{
GstOMXAudioDecClass *klass;
GstOMXBuffer *buf;
GstOMXAcquireBufferReturn acq_ret;
OMX_ERRORTYPE err;
GST_DEBUG_OBJECT (self, "Draining component");
klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
if (!self->started) {
GST_DEBUG_OBJECT (self, "Component not started yet");
return GST_FLOW_OK;
}
self->started = FALSE;
if ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
return GST_FLOW_OK;
}
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Send an EOS buffer to the component and let the base
* class drop the EOS event. We will send it later when
* the EOS buffer arrives on the output port. */
acq_ret = gst_omx_port_acquire_buffer (self->dec_in_port, &buf, GST_OMX_WAIT);
if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
GST_ERROR_OBJECT (self, "Failed to acquire buffer for draining: %d",
acq_ret);
return GST_FLOW_ERROR;
}
g_mutex_lock (&self->drain_lock);
self->draining = TRUE;
buf->omx_buf->nFilledLen = 0;
GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp,
gst_util_uint64_scale (self->last_upstream_ts, OMX_TICKS_PER_SECOND,
GST_SECOND));
buf->omx_buf->nTickCount = 0;
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_EOS;
err = gst_omx_port_release_buffer (self->dec_in_port, buf);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to drain component: %s (0x%08x)",
gst_omx_error_to_string (err), err);
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (self, "Waiting until component is drained");
if (G_UNLIKELY (self->dec->hacks & GST_OMX_HACK_DRAIN_MAY_NOT_RETURN)) {
gint64 wait_until = g_get_monotonic_time () + G_TIME_SPAN_SECOND / 2;
if (!g_cond_wait_until (&self->drain_cond, &self->drain_lock, wait_until))
GST_WARNING_OBJECT (self, "Drain timed out");
else
GST_DEBUG_OBJECT (self, "Drained component");
} else {
g_cond_wait (&self->drain_cond, &self->drain_lock);
GST_DEBUG_OBJECT (self, "Drained component");
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
self->started = FALSE;
return GST_FLOW_OK;
}
Reference in a new issue View git blame Copy permalink