static GstFlowReturn
gst_omx_audio_enc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
{
GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR;
GstOMXAudioEnc *self;
GstOMXPort *port;
GstOMXBuffer *buf;
gsize size;
guint offset = 0;
GstClockTime timestamp, duration, timestamp_offset = 0;
OMX_ERRORTYPE err;
self = GST_OMX_AUDIO_ENC (encoder);
if (self->eos) {
GST_WARNING_OBJECT (self, "Got frame after EOS");
return GST_FLOW_EOS;
}
if (self->downstream_flow_ret != GST_FLOW_OK) {
return self->downstream_flow_ret;
}
if (inbuf == NULL)
return GST_FLOW_OK;
GST_DEBUG_OBJECT (self, "Handling frame");
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
port = self->enc_in_port;
size = gst_buffer_get_size (inbuf);
while (offset < 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_ENCODER_STREAM_UNLOCK (self);
acq_ret = gst_omx_port_acquire_buffer (port, &buf);
if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto component_error;
} else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
GST_AUDIO_ENCODER_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_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
/* Now get a new buffer and fill it */
GST_AUDIO_ENCODER_STREAM_LOCK (self);
continue;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
if (self->downstream_flow_ret != GST_FLOW_OK) {
gst_omx_port_release_buffer (port, buf);
return self->downstream_flow_ret;
}
if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) {
gst_omx_port_release_buffer (port, buf);
goto full_buffer;
}
GST_DEBUG_OBJECT (self, "Handling frame at offset %d", offset);
/* Copy the buffer content in chunks of size as requested
* by the port */
buf->omx_buf->nFilledLen =
MIN (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);
/* Interpolate timestamps if we're passing the buffer
* in multiple chunks */
if (offset != 0 && duration != GST_CLOCK_TIME_NONE) {
timestamp_offset = gst_util_uint64_scale (offset, duration, size);
}
if (timestamp != GST_CLOCK_TIME_NONE) {
buf->omx_buf->nTimeStamp =
gst_util_uint64_scale (timestamp + timestamp_offset,
OMX_TICKS_PER_SECOND, GST_SECOND);
self->last_upstream_ts = timestamp + timestamp_offset;
}
if (duration != GST_CLOCK_TIME_NONE) {
buf->omx_buf->nTickCount =
gst_util_uint64_scale (buf->omx_buf->nFilledLen, duration, size);
self->last_upstream_ts += duration;
}
offset += buf->omx_buf->nFilledLen;
self->started = TRUE;
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
}
GST_DEBUG_OBJECT (self, "Passed frame to component");
return self->downstream_flow_ret;
full_buffer:
{
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;
}
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->enc),
gst_omx_component_get_last_error (self->enc)));
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
return GST_FLOW_FLUSHING;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure input port"));
return GST_FLOW_ERROR;
}
release_error:
{
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 gboolean
gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
GstOMXAudioEnc *self;
GstOMXAudioEncClass *klass;
gboolean needs_disable = FALSE;
OMX_PARAM_PORTDEFINITIONTYPE port_def;
OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
gint i;
OMX_ERRORTYPE err;
self = GST_OMX_AUDIO_ENC (encoder);
klass = GST_OMX_AUDIO_ENC_GET_CLASS (encoder);
GST_DEBUG_OBJECT (self, "Setting new caps");
/* Set audio encoder base class properties */
gst_audio_encoder_set_frame_samples_min (encoder,
gst_util_uint64_scale_ceil (OMX_MIN_PCMPAYLOAD_MSEC,
GST_MSECOND * info->rate, GST_SECOND));
gst_audio_encoder_set_frame_samples_max (encoder, 0);
gst_omx_port_get_port_definition (self->enc_in_port, &port_def);
needs_disable =
gst_omx_component_get_state (self->enc,
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) {
GST_DEBUG_OBJECT (self, "Need to disable and drain encoder");
gst_omx_audio_enc_drain (self);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, TRUE);
/* Wait until the srcpad loop is finished,
* unlock GST_AUDIO_ENCODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
gst_pad_stop_task (GST_AUDIO_ENCODER_SRC_PAD (encoder));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if (gst_omx_port_set_enabled (self->enc_in_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_set_enabled (self->enc_out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (self->enc_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->enc_out_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_in_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
GST_DEBUG_OBJECT (self, "Encoder drained and disabled");
}
port_def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
GST_DEBUG_OBJECT (self, "Setting inport port definition");
if (gst_omx_port_update_port_definition (self->enc_in_port,
&port_def) != OMX_ErrorNone)
return FALSE;
GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = self->enc_in_port->index;
pcm_param.nChannels = info->channels;
pcm_param.eNumData =
((info->finfo->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) ?
OMX_NumericalDataSigned : OMX_NumericalDataUnsigned);
pcm_param.eEndian =
((info->finfo->endianness == G_LITTLE_ENDIAN) ?
OMX_EndianLittle : OMX_EndianBig);
pcm_param.bInterleaved = OMX_TRUE;
pcm_param.nBitPerSample = info->finfo->width;
pcm_param.nSamplingRate = info->rate;
pcm_param.ePCMMode = OMX_AUDIO_PCMModeLinear;
for (i = 0; i < pcm_param.nChannels; i++) {
OMX_AUDIO_CHANNELTYPE pos;
switch (info->position[i]) {
case GST_AUDIO_CHANNEL_POSITION_MONO:
case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
pos = OMX_AUDIO_ChannelCF;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
pos = OMX_AUDIO_ChannelLF;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
pos = OMX_AUDIO_ChannelRF;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT:
pos = OMX_AUDIO_ChannelLS;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT:
pos = OMX_AUDIO_ChannelRS;
break;
case GST_AUDIO_CHANNEL_POSITION_LFE1:
pos = OMX_AUDIO_ChannelLFE;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER:
pos = OMX_AUDIO_ChannelCS;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT:
pos = OMX_AUDIO_ChannelLR;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT:
pos = OMX_AUDIO_ChannelRR;
break;
default:
pos = OMX_AUDIO_ChannelNone;
break;
}
pcm_param.eChannelMapping[i] = pos;
}
GST_DEBUG_OBJECT (self, "Setting PCM parameters");
err =
gst_omx_component_set_parameter (self->enc, OMX_IndexParamAudioPcm,
&pcm_param);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to set PCM parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return FALSE;
}
if (klass->set_format) {
if (!klass->set_format (self, self->enc_in_port, info)) {
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->enc_out_port,
NULL) != OMX_ErrorNone)
return FALSE;
GST_DEBUG_OBJECT (self, "Enabling component");
if (needs_disable) {
if (gst_omx_port_set_enabled (self->enc_in_port, TRUE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_mark_reconfigured (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
} else {
/* Disable output port */
if (gst_omx_port_set_enabled (self->enc_out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_set_state (self->enc, OMX_StateIdle) != OMX_ErrorNone)
return FALSE;
/* Need to allocate buffers to reach Idle state */
if (gst_omx_port_allocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateIdle)
return FALSE;
if (gst_omx_component_set_state (self->enc,
OMX_StateExecuting) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateExecuting)
return FALSE;
}
/* Unset flushing to allow ports to accept data again */
gst_omx_port_set_flushing (self->enc_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, FALSE);
if (gst_omx_component_get_last_error (self->enc) != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)",
gst_omx_component_get_last_error_string (self->enc),
gst_omx_component_get_last_error (self->enc));
return FALSE;
}
/* Start the srcpad loop again */
GST_DEBUG_OBJECT (self, "Starting task again");
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_enc_loop, encoder, NULL);
return TRUE;
}
static void
gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
{
GstOMXAudioEncClass *klass;
GstOMXPort *port = self->enc_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
acq_return = gst_omx_port_acquire_buffer (port, &buf);
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_ENCODER_SRC_PAD (self))
|| acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
GstAudioInfo *info =
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self));
GstCaps *caps;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
/* Reallocate all buffers */
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
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;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
caps = klass->get_caps (self, self->enc_out_port, info);
if (!caps) {
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
GST_DEBUG_OBJECT (self, "Setting output caps: %" GST_PTR_FORMAT, caps);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
GST_AUDIO_ENCODER_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_ENCODER_STREAM_LOCK (self);
goto eos;
}
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
(guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp);
/* This prevents a deadlock between the srcpad stream
* lock and the videocodec stream lock, if ::reset()
* is called at the wrong time
*/
if (gst_omx_port_is_flushing (self->enc_out_port)) {
GST_DEBUG_OBJECT (self, "Flushing");
gst_omx_port_release_buffer (self->enc_out_port, buf);
goto flushing;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
&& buf->omx_buf->nFilledLen > 0) {
GstCaps *caps;
GstBuffer *codec_data;
GstMapInfo map = GST_MAP_INFO_INIT;
GST_DEBUG_OBJECT (self, "Handling codec data");
caps =
gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
(self)));
codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (codec_data, &map);
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
flow_ret = GST_FLOW_OK;
} else if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
guint n_samples;
GST_DEBUG_OBJECT (self, "Handling output data");
n_samples =
klass->get_num_samples (self, self->enc_out_port,
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
if (buf->omx_buf->nFilledLen > 0) {
GstMapInfo map = GST_MAP_INFO_INIT;
outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (outbuf, &map);
} else {
outbuf = gst_buffer_new ();
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
OMX_TICKS_PER_SECOND);
if (buf->omx_buf->nTickCount != 0)
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
OMX_TICKS_PER_SECOND);
flow_ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
outbuf, n_samples);
}
GST_DEBUG_OBJECT (self, "Handled output data");
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
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_ENCODER_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->enc),
gst_omx_component_get_last_error (self->enc)));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
return;
}
eos:
{
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_ENCODER_SRC_PAD (self));
} else {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_ENCODER_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_ENCODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
} else if (flow_ret == GST_FLOW_NOT_LINKED || 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_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
}
self->started = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
caps_failed:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (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_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
}
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);
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))
buf->omx_buf->nTimeStamp =
gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
else
buf->omx_buf->nTimeStamp = 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) {
buf->omx_buf->nTimeStamp =
gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
self->last_upstream_ts = timestamp;
} else {
buf->omx_buf->nTimeStamp = 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 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 {
if (gst_omx_port_set_enabled (self->dec_in_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_set_enabled (out_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_wait_buffers_released (out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->dec_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->dec_out_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_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_loop (GstOMXAudioDec * self)
{
GstOMXPort *port = self->dec_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
acq_return = gst_omx_port_acquire_buffer (port, &buf);
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) buf->omx_buf->nTimeStamp);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
gint nframes, spf;
GstMapInfo minfo;
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
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);
nframes = 1;
spf = klass->get_samples_per_frame (self, self->dec_out_port);
if (spf != -1) {
nframes = buf->omx_buf->nFilledLen / self->info.bpf;
if (nframes % spf != 0)
GST_WARNING_OBJECT (self, "Output buffer does not contain an integer "
"number of input frames (frames: %d, spf: %d)", nframes, spf);
nframes = (nframes + spf - 1) / spf;
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
OMX_TICKS_PER_SECOND);
if (buf->omx_buf->nTickCount != 0)
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
OMX_TICKS_PER_SECOND);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
GST_DEBUG_OBJECT (self, "Read frame from component");
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");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
return;
}
eos:
{
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");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
}
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;
}
}
