GstCaps* getGstAudioCaps(int channels, float sampleRate)
{
return gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(sampleRate),
"channels", G_TYPE_INT, channels,
"format", G_TYPE_STRING, gst_audio_format_to_string(GST_AUDIO_FORMAT_F32),
"layout", G_TYPE_STRING, "interleaved", nullptr);
}
GstCaps* getGstAudioCaps(int channels, float sampleRate)
{
#ifdef GST_API_VERSION_1
return gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(sampleRate),
"channels", G_TYPE_INT, channels,
"format", G_TYPE_STRING, gst_audio_format_to_string(GST_AUDIO_FORMAT_F32),
"layout", G_TYPE_STRING, "interleaved", NULL);
#else
return gst_caps_new_simple("audio/x-raw-float", "rate", G_TYPE_INT, static_cast<int>(sampleRate),
"channels", G_TYPE_INT, channels,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 32, NULL);
#endif
}
void AudioFileReader::plugDeinterleave(GstPad* pad)
{
// A decodebin pad was added, plug in a deinterleave element to
// separate each planar channel. Sub pipeline looks like
// ... decodebin2 ! audioconvert ! audioresample ! capsfilter ! deinterleave.
GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
GstElement* audioResample = gst_element_factory_make("audioresample", 0);
GstElement* capsFilter = gst_element_factory_make("capsfilter", 0);
m_deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
g_object_set(m_deInterleave.get(), "keep-positions", TRUE, NULL);
g_signal_connect(m_deInterleave.get(), "pad-added", G_CALLBACK(onGStreamerDeinterleavePadAddedCallback), this);
g_signal_connect(m_deInterleave.get(), "no-more-pads", G_CALLBACK(onGStreamerDeinterleaveReadyCallback), this);
GstCaps* caps = gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(m_sampleRate),
"channels", G_TYPE_INT, 2,
"format", G_TYPE_STRING, gst_audio_format_to_string(GST_AUDIO_FORMAT_F32),
"layout", G_TYPE_STRING, "interleaved", nullptr);
g_object_set(capsFilter, "caps", caps, NULL);
gst_caps_unref(caps);
gst_bin_add_many(GST_BIN(m_pipeline), audioConvert, audioResample, capsFilter, m_deInterleave.get(), NULL);
GstPad* sinkPad = gst_element_get_static_pad(audioConvert, "sink");
gst_pad_link_full(pad, sinkPad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref(GST_OBJECT(sinkPad));
gst_element_link_pads_full(audioConvert, "src", audioResample, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(audioResample, "src", capsFilter, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(capsFilter, "src", m_deInterleave.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_sync_state_with_parent(audioConvert);
gst_element_sync_state_with_parent(audioResample);
gst_element_sync_state_with_parent(capsFilter);
gst_element_sync_state_with_parent(m_deInterleave.get());
}
static HRESULT WINAPI Gstreamer_AudioConvert_SetMediaType(TransformFilter *tf, PIN_DIRECTION dir, const AM_MEDIA_TYPE *amt)
{
GstTfImpl *This = (GstTfImpl*)tf;
GstCaps *capsin, *capsout;
AM_MEDIA_TYPE *outpmt = &This->tf.pmt;
WAVEFORMATEX *inwfe;
WAVEFORMATEX *outwfe;
WAVEFORMATEXTENSIBLE *outwfx;
GstAudioFormat format;
HRESULT hr;
BOOL inisfloat = FALSE;
int indepth;
TRACE("%p 0x%x %p\n", This, dir, amt);
mark_wine_thread();
if (dir != PINDIR_INPUT)
return S_OK;
if (Gstreamer_AudioConvert_QueryConnect(&This->tf, amt) == S_FALSE || !amt->pbFormat)
return E_FAIL;
FreeMediaType(outpmt);
*outpmt = *amt;
outpmt->pUnk = NULL;
outpmt->cbFormat = sizeof(WAVEFORMATEXTENSIBLE);
outpmt->pbFormat = CoTaskMemAlloc(outpmt->cbFormat);
inwfe = (WAVEFORMATEX*)amt->pbFormat;
indepth = inwfe->wBitsPerSample;
if (inwfe->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
WAVEFORMATEXTENSIBLE *inwfx = (WAVEFORMATEXTENSIBLE*)inwfe;
inisfloat = IsEqualGUID(&inwfx->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT);
if (inwfx->Samples.wValidBitsPerSample)
indepth = inwfx->Samples.wValidBitsPerSample;
} else if (inwfe->wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
inisfloat = TRUE;
if (inisfloat)
format = inwfe->wBitsPerSample == 64 ? GST_AUDIO_FORMAT_F64LE : GST_AUDIO_FORMAT_F32LE;
else
format = gst_audio_format_build_integer(inwfe->wBitsPerSample != 8, G_LITTLE_ENDIAN,
inwfe->wBitsPerSample, indepth);
capsin = gst_caps_new_simple("audio/x-raw",
"format", G_TYPE_STRING, gst_audio_format_to_string(format),
"channels", G_TYPE_INT, inwfe->nChannels,
"rate", G_TYPE_INT, inwfe->nSamplesPerSec,
NULL);
outwfe = (WAVEFORMATEX*)outpmt->pbFormat;
outwfx = (WAVEFORMATEXTENSIBLE*)outwfe;
outwfe->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
outwfe->nChannels = 2;
outwfe->nSamplesPerSec = inwfe->nSamplesPerSec;
outwfe->wBitsPerSample = 16;
outwfe->nBlockAlign = outwfe->nChannels * outwfe->wBitsPerSample / 8;
outwfe->nAvgBytesPerSec = outwfe->nBlockAlign * outwfe->nSamplesPerSec;
outwfe->cbSize = sizeof(*outwfx) - sizeof(*outwfe);
outwfx->Samples.wValidBitsPerSample = outwfe->wBitsPerSample;
outwfx->dwChannelMask = SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT;
outwfx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
capsout = gst_caps_new_simple("audio/x-raw",
"format", G_TYPE_STRING, "S16LE",
"channels", G_TYPE_INT, outwfe->nChannels,
"rate", G_TYPE_INT, outwfe->nSamplesPerSec,
NULL);
hr = Gstreamer_transform_ConnectInput(This, amt, capsin, capsout);
gst_caps_unref(capsin);
gst_caps_unref(capsout);
This->cbBuffer = inwfe->nAvgBytesPerSec;
return hr;
}
/**
* gst_audio_info_to_caps:
* @info: a #GstAudioInfo
*
* Convert the values of @info into a #GstCaps.
*
* Returns: (transfer full): the new #GstCaps containing the
* info of @info.
*/
GstCaps *
gst_audio_info_to_caps (const GstAudioInfo * info)
{
GstCaps *caps;
const gchar *format;
const gchar *layout;
GstAudioFlags flags;
g_return_val_if_fail (info != NULL, NULL);
g_return_val_if_fail (info->finfo != NULL, NULL);
g_return_val_if_fail (info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
format = gst_audio_format_to_string (info->finfo->format);
g_return_val_if_fail (format != NULL, NULL);
if (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED)
layout = "interleaved";
else if (info->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED)
layout = "non-interleaved";
else
g_return_val_if_reached (NULL);
flags = info->flags;
if ((flags & GST_AUDIO_FLAG_UNPOSITIONED) && info->channels > 1
&& info->position[0] != GST_AUDIO_CHANNEL_POSITION_NONE) {
flags &= ~GST_AUDIO_FLAG_UNPOSITIONED;
g_warning ("Unpositioned audio channel position flag set but "
"channel positions present");
} else if (!(flags & GST_AUDIO_FLAG_UNPOSITIONED) && info->channels > 1
&& info->position[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
flags |= GST_AUDIO_FLAG_UNPOSITIONED;
g_warning ("Unpositioned audio channel position flag not set "
"but no channel positions present");
}
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, format,
"layout", G_TYPE_STRING, layout,
"rate", G_TYPE_INT, info->rate,
"channels", G_TYPE_INT, info->channels, NULL);
if (info->channels > 1
|| info->position[0] != GST_AUDIO_CHANNEL_POSITION_MONO) {
guint64 channel_mask = 0;
if ((flags & GST_AUDIO_FLAG_UNPOSITIONED)) {
channel_mask = 0;
} else {
if (!gst_audio_channel_positions_to_mask (info->position, info->channels,
TRUE, &channel_mask))
goto invalid_channel_positions;
}
if (info->channels == 1
&& info->position[0] == GST_AUDIO_CHANNEL_POSITION_MONO) {
/* Default mono special case */
} else {
gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
NULL);
}
}
return caps;
invalid_channel_positions:
{
GST_ERROR ("Invalid channel positions");
gst_caps_unref (caps);
return NULL;
}
}
GstCaps *QGstUtils::capsForAudioFormat(const QAudioFormat &format)
{
if (!format.isValid())
return 0;
#if GST_CHECK_VERSION(1,0,0)
const QAudioFormat::SampleType sampleType = format.sampleType();
const QAudioFormat::Endian byteOrder = format.byteOrder();
const int sampleSize = format.sampleSize();
for (int i = 0; i < lengthOf(qt_audioLookup); ++i) {
if (qt_audioLookup[i].sampleType != sampleType
|| qt_audioLookup[i].byteOrder != byteOrder
|| qt_audioLookup[i].sampleSize != sampleSize) {
continue;
}
return gst_caps_new_simple(
"audio/x-raw",
"format" , G_TYPE_STRING, gst_audio_format_to_string(qt_audioLookup[i].format),
"rate" , G_TYPE_INT , format.sampleRate(),
"channels", G_TYPE_INT , format.channelCount(),
NULL);
}
return 0;
#else
GstStructure *structure = 0;
if (format.isValid()) {
if (format.sampleType() == QAudioFormat::SignedInt || format.sampleType() == QAudioFormat::UnSignedInt) {
structure = gst_structure_new("audio/x-raw-int", NULL);
} else if (format.sampleType() == QAudioFormat::Float) {
structure = gst_structure_new("audio/x-raw-float", NULL);
}
}
GstCaps *caps = 0;
if (structure) {
gst_structure_set(structure, "rate", G_TYPE_INT, format.sampleRate(), NULL);
gst_structure_set(structure, "channels", G_TYPE_INT, format.channelCount(), NULL);
gst_structure_set(structure, "width", G_TYPE_INT, format.sampleSize(), NULL);
gst_structure_set(structure, "depth", G_TYPE_INT, format.sampleSize(), NULL);
if (format.byteOrder() == QAudioFormat::LittleEndian)
gst_structure_set(structure, "endianness", G_TYPE_INT, 1234, NULL);
else if (format.byteOrder() == QAudioFormat::BigEndian)
gst_structure_set(structure, "endianness", G_TYPE_INT, 4321, NULL);
if (format.sampleType() == QAudioFormat::SignedInt)
gst_structure_set(structure, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
else if (format.sampleType() == QAudioFormat::UnSignedInt)
gst_structure_set(structure, "signed", G_TYPE_BOOLEAN, FALSE, NULL);
caps = gst_caps_new_empty();
Q_ASSERT(caps);
gst_caps_append_structure(caps, structure);
}
return caps;
#endif
}
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;
}
}
static GstFlowReturn gst_imx_audio_uniaudio_dec_handle_frame(GstAudioDecoder *dec, GstBuffer *buffer)
{
GstMapInfo in_map;
GstBuffer *out_buffer;
gsize avail_out_size;
GstImxAudioUniaudioDec *imx_audio_uniaudio_dec = GST_IMX_AUDIO_UNIAUDIO_DEC(dec);
int32 dec_ret;
uint32 offset = 0;
uint8 *in_buf = NULL;
uint32 in_size = 0;
gboolean dec_loop = TRUE, flow_error = FALSE;
/* With some formats such as Vorbis, the first few buffers are actually redundant,
* since they contain codec data that was already specified in codec_data or
* streamheader caps earlier. If this is the case, skip these buffers. */
if (imx_audio_uniaudio_dec->skip_header_counter < imx_audio_uniaudio_dec->num_vorbis_headers)
{
GST_TRACE_OBJECT(dec, "skipping header buffer #%u", imx_audio_uniaudio_dec->skip_header_counter);
++imx_audio_uniaudio_dec->skip_header_counter;
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
if (buffer != NULL)
{
gst_buffer_map(buffer, &in_map, GST_MAP_READ);
in_buf = in_map.data;
in_size = in_map.size;
}
while (dec_loop)
{
GstBuffer *tmp_buf;
uint8 *out_buf = NULL;
uint32 out_size = 0;
if (buffer != NULL)
GST_TRACE_OBJECT(dec, "feeding %" G_GUINT32_FORMAT " bytes to the decoder", (guint32)in_size);
else
GST_TRACE_OBJECT(dec, "draining decoder");
dec_ret = imx_audio_uniaudio_dec->codec->decode_frame(
imx_audio_uniaudio_dec->handle,
in_buf, in_size,
&offset,
&out_buf, &out_size
);
GST_TRACE_OBJECT(dec, "decode_frame: return 0x%x offset %" G_GUINT32_FORMAT " out_size %" G_GUINT32_FORMAT, (unsigned int)dec_ret, (guint32)offset, (guint32)out_size);
if ((out_buf != NULL) && (out_size > 0))
{
tmp_buf = gst_audio_decoder_allocate_output_buffer(dec, out_size);
tmp_buf = gst_buffer_make_writable(tmp_buf);
gst_buffer_fill(tmp_buf, 0, out_buf, out_size);
gst_adapter_push(imx_audio_uniaudio_dec->out_adapter, tmp_buf);
}
if (out_buf != NULL)
{
gst_imx_audio_uniaudio_dec_free(out_buf);
}
if ((buffer != NULL) && (offset == in_map.size))
{
dec_loop = FALSE;
}
switch (dec_ret)
{
case ACODEC_SUCCESS:
break;
case ACODEC_END_OF_STREAM:
dec_loop = FALSE;
break;
case ACODEC_NOT_ENOUGH_DATA:
break;
case ACODEC_CAPIBILITY_CHANGE:
break;
default:
{
dec_loop = FALSE;
flow_error = TRUE;
GST_ELEMENT_ERROR(dec, STREAM, DECODE, ("could not decode"), ("error message: %s", imx_audio_uniaudio_dec->codec->get_last_error(imx_audio_uniaudio_dec->handle)));
}
}
}
if (buffer != NULL)
gst_buffer_unmap(buffer, &in_map);
if (flow_error)
return GST_FLOW_ERROR;
if (!(imx_audio_uniaudio_dec->has_audioinfo_set))
{
UniACodecParameter parameter;
GstAudioFormat pcm_fmt;
GstAudioInfo audio_info;
imx_audio_uniaudio_dec->codec->get_parameter(imx_audio_uniaudio_dec->handle, UNIA_OUTPUT_PCM_FORMAT, ¶meter);
if ((parameter.outputFormat.width == 0) || (parameter.outputFormat.depth == 0))
{
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "no output format available yet");
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "output sample width: %" G_GUINT32_FORMAT " depth: %" G_GUINT32_FORMAT, (guint32)(parameter.outputFormat.width), (guint32)(parameter.outputFormat.depth));
pcm_fmt = gst_audio_format_build_integer(TRUE, G_BYTE_ORDER, parameter.outputFormat.width, parameter.outputFormat.depth);
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "setting output format to: %s %d Hz %d channels", gst_audio_format_to_string(pcm_fmt), (gint)(parameter.outputFormat.samplerate), (gint)(parameter.outputFormat.channels));
gst_imx_audio_uniaudio_dec_clear_channel_positions(imx_audio_uniaudio_dec);
gst_imx_audio_uniaudio_dec_fill_channel_positions(imx_audio_uniaudio_dec, parameter.outputFormat.layout, parameter.outputFormat.channels);
imx_audio_uniaudio_dec->pcm_format = pcm_fmt;
imx_audio_uniaudio_dec->num_channels = parameter.outputFormat.channels;
gst_audio_info_set_format(
&audio_info,
pcm_fmt,
parameter.outputFormat.samplerate,
parameter.outputFormat.channels,
imx_audio_uniaudio_dec->reordered_channel_positions
);
gst_audio_decoder_set_output_format(dec, &audio_info);
imx_audio_uniaudio_dec->has_audioinfo_set = TRUE;
}
avail_out_size = gst_adapter_available(imx_audio_uniaudio_dec->out_adapter);
if (avail_out_size > 0)
{
out_buffer = gst_adapter_take_buffer(imx_audio_uniaudio_dec->out_adapter, avail_out_size);
if (imx_audio_uniaudio_dec->original_channel_positions != imx_audio_uniaudio_dec->reordered_channel_positions)
{
gst_audio_buffer_reorder_channels(
out_buffer,
imx_audio_uniaudio_dec->pcm_format,
imx_audio_uniaudio_dec->num_channels,
imx_audio_uniaudio_dec->original_channel_positions,
imx_audio_uniaudio_dec->reordered_channel_positions
);
}
return gst_audio_decoder_finish_frame(dec, out_buffer, 1);
}
else
{
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
}
