static GstFlowReturn
gst_atdec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer)
{
AudioTimeStamp timestamp = { 0 };
AudioStreamPacketDescription packet;
AudioQueueBufferRef input_buffer, output_buffer;
GstBuffer *out;
GstMapInfo info;
GstAudioInfo *audio_info;
int size, out_frames;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstATDec *atdec = GST_ATDEC (decoder);
if (buffer == NULL)
return GST_FLOW_OK;
audio_info = gst_audio_decoder_get_audio_info (decoder);
/* copy the input buffer into an AudioQueueBuffer */
size = gst_buffer_get_size (buffer);
AudioQueueAllocateBuffer (atdec->queue, size, &input_buffer);
gst_buffer_extract (buffer, 0, input_buffer->mAudioData, size);
input_buffer->mAudioDataByteSize = size;
/* assume framed input */
packet.mStartOffset = 0;
packet.mVariableFramesInPacket = 1;
packet.mDataByteSize = size;
/* enqueue the buffer. It will get free'd once the gst_atdec_buffer_emptied
* callback is called
*/
AudioQueueEnqueueBuffer (atdec->queue, input_buffer, 1, &packet);
/* figure out how many frames we need to pull out of the queue */
out_frames = GST_CLOCK_TIME_TO_FRAMES (GST_BUFFER_DURATION (buffer),
audio_info->rate);
size = out_frames * audio_info->bpf;
AudioQueueAllocateBuffer (atdec->queue, size, &output_buffer);
/* pull the frames */
AudioQueueOfflineRender (atdec->queue, ×tamp, output_buffer, out_frames);
if (output_buffer->mAudioDataByteSize) {
out =
gst_audio_decoder_allocate_output_buffer (decoder,
output_buffer->mAudioDataByteSize);
gst_buffer_map (out, &info, GST_MAP_WRITE);
memcpy (info.data, output_buffer->mAudioData,
output_buffer->mAudioDataByteSize);
gst_buffer_unmap (out, &info);
flow_ret = gst_audio_decoder_finish_frame (decoder, out, 1);
}
AudioQueueFreeBuffer (atdec->queue, output_buffer);
return flow_ret;
}
static GstFlowReturn
gst_mulawdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstMapInfo inmap, outmap;
gint16 *linear_data;
guint8 *mulaw_data;
gsize mulaw_size, linear_size;
GstBuffer *outbuf;
if (!buffer) {
return GST_FLOW_OK;
}
if (!gst_buffer_map (buffer, &inmap, GST_MAP_READ)) {
GST_ERROR ("failed to map input buffer");
goto error_failed_map_input_buffer;
}
mulaw_data = inmap.data;
mulaw_size = inmap.size;
linear_size = mulaw_size * 2;
outbuf = gst_audio_decoder_allocate_output_buffer (dec, linear_size);
if (!gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE)) {
GST_ERROR ("failed to map input buffer");
goto error_failed_map_output_buffer;
}
linear_data = (gint16 *) outmap.data;
mulaw_decode (mulaw_data, linear_data, mulaw_size);
gst_buffer_unmap (outbuf, &outmap);
gst_buffer_unmap (buffer, &inmap);
return gst_audio_decoder_finish_frame (dec, outbuf, -1);
error_failed_map_output_buffer:
gst_buffer_unref (outbuf);
error_failed_map_input_buffer:
return GST_FLOW_ERROR;
}
/* called when ffmpeg wants us to allocate a buffer to write the decoded frame
* into. We try to give it memory from our pool */
static int
gst_ffmpegauddec_get_buffer (AVCodecContext * context, AVFrame * frame)
{
GstFFMpegAudDec *ffmpegdec;
GstAudioInfo *info;
BufferInfo *buffer_info;
ffmpegdec = (GstFFMpegAudDec *) context->opaque;
if (G_UNLIKELY (!gst_ffmpegauddec_negotiate (ffmpegdec, FALSE)))
goto negotiate_failed;
/* Always use the default allocator for planar audio formats because
* we will have to copy and deinterleave later anyway */
if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt))
goto fallback;
info = gst_audio_decoder_get_audio_info (GST_AUDIO_DECODER (ffmpegdec));
buffer_info = g_slice_new (BufferInfo);
buffer_info->buffer =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (ffmpegdec),
frame->nb_samples * info->bpf);
gst_buffer_map (buffer_info->buffer, &buffer_info->map, GST_MAP_WRITE);
frame->opaque = buffer_info;
frame->data[0] = buffer_info->map.data;
frame->extended_data = frame->data;
frame->linesize[0] = buffer_info->map.size;
frame->type = FF_BUFFER_TYPE_USER;
return 0;
/* fallbacks */
negotiate_failed:
{
GST_DEBUG_OBJECT (ffmpegdec, "negotiate failed");
goto fallback;
}
fallback:
{
return avcodec_default_get_buffer (context, frame);
}
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gsize size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
GstMapInfo map, omap;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
gst_opus_dec_negotiate (dec, NULL);
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_GST_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& gst_buffer_get_size (dec->last_buffer) >
0) ? dec->last_buffer : buffer;
if (buf && gst_buffer_get_size (buf) > 0) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
packet_size = samples * dec->n_channels * 2;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
if (dec->use_inband_fec) {
if (dec->last_buffer) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
}
gst_buffer_unmap (outbuf, &omap);
if (data != NULL)
gst_buffer_unmap (buf, &map);
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
gst_buffer_unref (outbuf);
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
/* Skip any samples that need skipping */
if (dec->pre_skip > 0) {
guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
dec->pre_skip -= scaled_skip;
GST_INFO_OBJECT (dec,
"Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
scaled_skip, dec->pre_skip);
}
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
gint16 *samples;
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
samples = (gint16 *) omap.data;
rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
gst_buffer_unmap (outbuf, &omap);
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
done:
return res;
creation_failed:
GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
return GST_FLOW_ERROR;
buffer_failed:
GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size);
return GST_FLOW_ERROR;
}
static GstFlowReturn
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
GstClockTime timestamp, GstClockTime duration)
{
#ifdef USE_TREMOLO
vorbis_sample_t *pcm;
#else
vorbis_sample_t **pcm;
#endif
guint sample_count;
GstBuffer *out = NULL;
GstFlowReturn result;
GstMapInfo map;
gsize size;
if (G_UNLIKELY (!vd->initialized)) {
result = vorbis_dec_handle_header_caps (vd);
if (result != GST_FLOW_OK)
goto not_initialized;
}
/* normal data packet */
/* FIXME, we can skip decoding if the packet is outside of the
* segment, this is however not very trivial as we need a previous
* packet to decode the current one so we must be careful not to
* throw away too much. For now we decode everything and clip right
* before pushing data. */
#ifdef USE_TREMOLO
if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vd, packet, 1)))
goto could_not_read;
#else
if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet)))
goto could_not_read;
if (G_UNLIKELY (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0))
goto not_accepted;
#endif
/* assume all goes well here */
result = GST_FLOW_OK;
/* count samples ready for reading */
#ifdef USE_TREMOLO
if ((sample_count = vorbis_dsp_pcmout (&vd->vd, NULL, 0)) == 0)
#else
if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0)
goto done;
#endif
size = sample_count * vd->info.bpf;
GST_LOG_OBJECT (vd, "%d samples ready for reading, size %" G_GSIZE_FORMAT,
sample_count, size);
/* alloc buffer for it */
out = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (vd), size);
gst_buffer_map (out, &map, GST_MAP_WRITE);
/* get samples ready for reading now, should be sample_count */
#ifdef USE_TREMOLO
if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, map.data, sample_count) !=
sample_count))
#else
if (G_UNLIKELY (vorbis_synthesis_pcmout (&vd->vd, &pcm) != sample_count))
#endif
goto wrong_samples;
#ifdef USE_TREMOLO
if (vd->info.channels < 9)
gst_audio_reorder_channels (map.data, map.size, GST_VORBIS_AUDIO_FORMAT,
vd->info.channels, gst_vorbis_channel_positions[vd->info.channels - 1],
gst_vorbis_default_channel_positions[vd->info.channels - 1]);
#else
/* copy samples in buffer */
vd->copy_samples ((vorbis_sample_t *) map.data, pcm,
sample_count, vd->info.channels);
#endif
GST_LOG_OBJECT (vd, "have output size of %" G_GSIZE_FORMAT, size);
gst_buffer_unmap (out, &map);
done:
/* whether or not data produced, consume one frame and advance time */
result = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), out, 1);
#ifdef USE_TREMOLO
vorbis_dsp_read (&vd->vd, sample_count);
#else
vorbis_synthesis_read (&vd->vd, sample_count);
#endif
return result;
/* ERRORS */
not_initialized:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("no header sent yet"));
return GST_FLOW_NOT_NEGOTIATED;
}
could_not_read:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't read data packet"));
return GST_FLOW_ERROR;
}
not_accepted:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("vorbis decoder did not accept data packet"));
return GST_FLOW_ERROR;
}
wrong_samples:
{
gst_buffer_unref (out);
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("vorbis decoder reported wrong number of samples"));
return GST_FLOW_ERROR;
}
}
static void
gst_amc_audio_dec_loop (GstAmcAudioDec * self)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
gboolean is_eos;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
gint idx;
GError *err = NULL;
GST_AUDIO_DECODER_STREAM_LOCK (self);
retry:
/*if (self->input_caps_changed) {
idx = INFO_OUTPUT_FORMAT_CHANGED;
} else { */
GST_DEBUG_OBJECT (self, "Waiting for available output buffer");
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx =
gst_amc_codec_dequeue_output_buffer (self->codec, &buffer_info, 100000,
&err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
/*} */
if (idx < 0) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
switch (idx) {
case INFO_OUTPUT_BUFFERS_CHANGED:
/* Handled internally */
g_assert_not_reached ();
break;
case INFO_OUTPUT_FORMAT_CHANGED:{
GstAmcFormat *format;
gchar *format_string;
GST_DEBUG_OBJECT (self, "Output format has changed");
format = gst_amc_codec_get_output_format (self->codec, &err);
if (!format)
goto format_error;
format_string = gst_amc_format_to_string (format, &err);
if (err) {
gst_amc_format_free (format);
goto format_error;
}
GST_DEBUG_OBJECT (self, "Got new output format: %s", format_string);
g_free (format_string);
if (!gst_amc_audio_dec_set_src_caps (self, format)) {
gst_amc_format_free (format);
goto format_error;
}
gst_amc_format_free (format);
goto retry;
}
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out");
goto retry;
case G_MININT:
GST_ERROR_OBJECT (self, "Failure dequeueing output buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
goto retry;
}
GST_DEBUG_OBJECT (self,
"Got output buffer at index %d: offset %d size %d time %" G_GINT64_FORMAT
" flags 0x%08x", idx, buffer_info.offset, buffer_info.size,
buffer_info.presentation_time_us, buffer_info.flags);
is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);
buf = gst_amc_codec_get_output_buffer (self->codec, idx, &err);
if (!buf)
goto failed_to_get_output_buffer;
if (buffer_info.size > 0) {
GstBuffer *outbuf;
GstMapInfo minfo;
/* This sometimes happens at EOS or if the input is not properly framed,
* let's handle it gracefully by allocating a new buffer for the current
* caps and filling it
*/
if (buffer_info.size % self->info.bpf != 0)
goto invalid_buffer_size;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buffer_info.size);
if (!outbuf)
goto failed_allocate;
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->data + buffer_info.offset);
n_samples = buffer_info.size / 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 {
orc_memcpy (minfo.data, buf->data + buffer_info.offset, buffer_info.size);
}
gst_buffer_unmap (outbuf, &minfo);
if (self->spf != -1) {
gst_adapter_push (self->output_adapter, outbuf);
} else {
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
}
}
gst_amc_buffer_free (buf);
buf = NULL;
if (self->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.
* Otherwise we take a multiple of codec frames and push
* them downstream
*/
avail /= self->info.bpf;
if (!is_eos) {
nframes = avail / self->spf;
avail = nframes * self->spf;
} else {
nframes = (avail + self->spf - 1) / self->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);
}
}
if (!gst_amc_codec_release_output_buffer (self->codec, idx, &err)) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto failed_release;
}
if (is_eos || flow_ret == GST_FLOW_EOS) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
} else if (flow_ret == GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
} else {
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
dequeue_error:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
format_error:
{
if (err)
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
else
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to handle format"));
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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_release:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
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;
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));
} 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));
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_to_get_output_buffer:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
invalid_buffer_size:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Invalid buffer size %u (bfp %d)", buffer_info.size, self->info.bpf));
gst_amc_codec_release_output_buffer (self->codec, idx, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_allocate:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to allocate output buffer"));
gst_amc_codec_release_output_buffer (self->codec, idx, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gsize size;
guint8 *data;
GstBuffer *outbuf, *bufd;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
GstMapInfo map, omap;
GstAudioClippingMeta *cmeta = NULL;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
if (!gst_opus_dec_negotiate (dec, NULL))
return GST_FLOW_NOT_NEGOTIATED;
}
if (dec->n_channels == 2 && dec->n_streams == 1
&& dec->n_stereo_streams == 0) {
/* if we are automatically decoding 2 channels, but only have
a single encoded one, direct both channels to it */
dec->channel_mapping[1] = 0;
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_GST_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& gst_buffer_get_size (dec->last_buffer) >
0) ? dec->last_buffer : buffer;
/* That's the buffer we get duration from */
bufd = dec->use_inband_fec ? dec->last_buffer : buffer;
if (buf && gst_buffer_get_size (buf) > 0) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
if (gst_buffer_get_size (bufd) == 0) {
GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;
GstClockTime aligned_missing_duration;
GstClockTime missing_duration = GST_BUFFER_DURATION (bufd);
if (!GST_CLOCK_TIME_IS_VALID (missing_duration) || missing_duration == 0) {
if (GST_CLOCK_TIME_IS_VALID (dec->last_known_buffer_duration)) {
missing_duration = dec->last_known_buffer_duration;
GST_WARNING_OBJECT (dec,
"Missing duration, using last duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (missing_duration));
} else {
GST_WARNING_OBJECT (dec,
"Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms");
missing_duration = 20 * GST_MSECOND;
}
}
GST_DEBUG_OBJECT (dec,
"missing buffer, doing PLC duration %" GST_TIME_FORMAT
" plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (dec->leftover_plc_duration));
/* add the leftover PLC duration to that of the buffer */
missing_duration += dec->leftover_plc_duration;
/* align the combined buffer and leftover PLC duration to multiples
* of 2.5ms, rounding to nearest, and store excess duration for later */
aligned_missing_duration =
((missing_duration +
opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment;
dec->leftover_plc_duration = missing_duration - aligned_missing_duration;
/* Opus' PLC cannot operate with less than 2.5ms; skip PLC
* and accumulate the missing duration in the leftover_plc_duration
* for the next PLC attempt */
if (aligned_missing_duration < opus_plc_alignment) {
GST_DEBUG_OBJECT (dec,
"current duration %" GST_TIME_FORMAT
" of missing data not enough for PLC (minimum needed: %"
GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (opus_plc_alignment));
goto done;
}
/* convert the duration (in nanoseconds) to sample count */
samples =
gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,
GST_SECOND);
GST_DEBUG_OBJECT (dec,
"calculated PLC frame length: %" GST_TIME_FORMAT
" num frame samples: %d new leftover: %" GST_TIME_FORMAT,
GST_TIME_ARGS (aligned_missing_duration), samples,
GST_TIME_ARGS (dec->leftover_plc_duration));
} else {
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
}
packet_size = samples * dec->n_channels * 2;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
if (size > 0)
dec->last_known_buffer_duration = packet_duration_opus (data, size);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
do {
if (dec->use_inband_fec) {
if (gst_buffer_get_size (dec->last_buffer) > 0) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
}
if (n == OPUS_BUFFER_TOO_SMALL) {
/* if too small, add 2.5 milliseconds and try again, up to the
* Opus max size of 120 milliseconds */
if (samples >= 120 * dec->sample_rate / 1000)
break;
samples += 25 * dec->sample_rate / 10000;
packet_size = samples * dec->n_channels * 2;
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
}
} while (n == OPUS_BUFFER_TOO_SMALL);
gst_buffer_unmap (outbuf, &omap);
if (data != NULL)
gst_buffer_unmap (buf, &map);
if (n < 0) {
GstFlowReturn ret = GST_FLOW_ERROR;
gst_buffer_unref (outbuf);
GST_AUDIO_DECODER_ERROR (dec, 1, STREAM, DECODE, (NULL),
("Decoding error (%d): %s", n, opus_strerror (n)), ret);
return ret;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
GST_BUFFER_DURATION (outbuf) = samples * GST_SECOND / dec->sample_rate;
samples = n;
cmeta = gst_buffer_get_audio_clipping_meta (buf);
g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
/* Skip any samples that need skipping */
if (cmeta && cmeta->start) {
guint pre_skip = cmeta->start;
guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
GST_INFO_OBJECT (dec,
"Skipping %u samples at the beginning (%u at 48000 Hz)",
skip, scaled_skip);
}
if (cmeta && cmeta->end) {
guint post_skip = cmeta->end;
guint scaled_post_skip = post_skip * dec->sample_rate / 48000;
guint skip = scaled_post_skip > n ? n : scaled_post_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
guint outsize = gst_buffer_get_size (outbuf);
guint skip_bytes = skip * 2 * dec->n_channels;
if (outsize > skip_bytes)
outsize -= skip_bytes;
else
outsize = 0;
gst_buffer_resize (outbuf, 0, outsize);
GST_INFO_OBJECT (dec,
"Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip);
}
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
gint16 *samples;
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
samples = (gint16 *) omap.data;
rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
gst_buffer_unmap (outbuf, &omap);
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
done:
return res;
creation_failed:
GST_ELEMENT_ERROR (dec, LIBRARY, INIT, ("Failed to create Opus decoder"),
("Failed to create Opus decoder (%d): %s", err, opus_strerror (err)));
return GST_FLOW_ERROR;
buffer_failed:
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to create %u byte buffer", packet_size));
return GST_FLOW_ERROR;
}
static void
gst_amc_audio_dec_loop (GstAmcAudioDec * self)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
gboolean is_eos;
GstAmcBufferInfo buffer_info;
gint idx;
GST_AUDIO_DECODER_STREAM_LOCK (self);
retry:
/*if (self->input_caps_changed) {
idx = INFO_OUTPUT_FORMAT_CHANGED;
} else { */
GST_DEBUG_OBJECT (self, "Waiting for available output buffer");
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx = gst_amc_codec_dequeue_output_buffer (self->codec, &buffer_info, 100000);
GST_AUDIO_DECODER_STREAM_LOCK (self);
/*} */
if (idx < 0) {
if (self->flushing)
goto flushing;
switch (idx) {
case INFO_OUTPUT_BUFFERS_CHANGED:{
GST_DEBUG_OBJECT (self, "Output buffers have changed");
if (self->output_buffers)
gst_amc_codec_free_buffers (self->output_buffers,
self->n_output_buffers);
self->output_buffers =
gst_amc_codec_get_output_buffers (self->codec,
&self->n_output_buffers);
if (!self->output_buffers)
goto get_output_buffers_error;
break;
}
case INFO_OUTPUT_FORMAT_CHANGED:{
GstAmcFormat *format;
gchar *format_string;
GST_DEBUG_OBJECT (self, "Output format has changed");
format = gst_amc_codec_get_output_format (self->codec);
if (!format)
goto format_error;
format_string = gst_amc_format_to_string (format);
GST_DEBUG_OBJECT (self, "Got new output format: %s", format_string);
g_free (format_string);
if (!gst_amc_audio_dec_set_src_caps (self, format)) {
gst_amc_format_free (format);
goto format_error;
}
gst_amc_format_free (format);
if (self->output_buffers)
gst_amc_codec_free_buffers (self->output_buffers,
self->n_output_buffers);
self->output_buffers =
gst_amc_codec_get_output_buffers (self->codec,
&self->n_output_buffers);
if (!self->output_buffers)
goto get_output_buffers_error;
goto retry;
break;
}
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out");
goto retry;
break;
case G_MININT:
GST_ERROR_OBJECT (self, "Failure dequeueing output buffer");
goto dequeue_error;
break;
default:
g_assert_not_reached ();
break;
}
goto retry;
}
GST_DEBUG_OBJECT (self,
"Got output buffer at index %d: size %d time %" G_GINT64_FORMAT
" flags 0x%08x", idx, buffer_info.size, buffer_info.presentation_time_us,
buffer_info.flags);
is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);
self->n_buffers++;
if (buffer_info.size > 0) {
GstAmcAudioDecClass *klass = GST_AMC_AUDIO_DEC_GET_CLASS (self);
GstBuffer *outbuf;
GstAmcBuffer *buf;
GstMapInfo minfo;
/* This sometimes happens at EOS or if the input is not properly framed,
* let's handle it gracefully by allocating a new buffer for the current
* caps and filling it
*/
if (idx >= self->n_output_buffers)
goto invalid_buffer_index;
if (strcmp (klass->codec_info->name, "OMX.google.mp3.decoder") == 0) {
/* Google's MP3 decoder outputs garbage in the first output buffer
* so we just drop it here */
if (self->n_buffers == 1) {
GST_DEBUG_OBJECT (self,
"Skipping first buffer of Google MP3 decoder output");
goto done;
}
}
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buffer_info.size);
if (!outbuf)
goto failed_allocate;
gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE);
buf = &self->output_buffers[idx];
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->data + buffer_info.offset);
n_samples = buffer_info.size / 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 {
orc_memcpy (minfo.data, buf->data + buffer_info.offset, buffer_info.size);
}
gst_buffer_unmap (outbuf, &minfo);
/* FIXME: We should get one decoded input frame here for
* every buffer. If this is not the case somewhere, we will
* error out at some point and will need to add workarounds
*/
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
}
done:
if (!gst_amc_codec_release_output_buffer (self->codec, idx))
goto failed_release;
if (is_eos || flow_ret == GST_FLOW_EOS) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
} else if (flow_ret == GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
} else {
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
dequeue_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to dequeue output 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_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
get_output_buffers_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to get output buffers"));
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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
format_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to handle format"));
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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
failed_release:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to release output buffer index %d", idx));
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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
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;
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));
} 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_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
invalid_buffer_index:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Invalid input buffer index %d of %d", idx, self->n_input_buffers));
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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
failed_allocate:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to allocate output 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_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
}
static GstFlowReturn
gst_siren_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstSirenDec *dec;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size, in_size;
gint decode_ret;
GstMapInfo inmap, outmap;
dec = GST_SIREN_DEC (bdec);
size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (dec, "Received buffer of size %u", size);
g_return_val_if_fail (size % 40 == 0, GST_FLOW_ERROR);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
/* process 40 input bytes into 640 output bytes */
num_frames = size / 40;
/* this is the input/output size */
in_size = num_frames * 40;
out_size = num_frames * 640;
GST_LOG_OBJECT (dec, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
out_buf = gst_audio_decoder_allocate_output_buffer (bdec, out_size);
if (out_buf == NULL)
goto alloc_failed;
/* get the input data for all the frames */
gst_buffer_map (buf, &inmap, GST_MAP_READ);
gst_buffer_map (out_buf, &outmap, GST_MAP_WRITE);
in_data = inmap.data;
out_data = outmap.data;
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (dec, "Decoding frame %u/%u", i, num_frames);
/* decode 40 input bytes to 640 output bytes */
decode_ret = Siren7_DecodeFrame (dec->decoder, in_data, out_data);
if (decode_ret != 0)
goto decode_error;
/* move to next frame */
out_data += 640;
in_data += 40;
}
gst_buffer_unmap (buf, &inmap);
gst_buffer_unmap (out_buf, &outmap);
GST_LOG_OBJECT (dec, "Finished decoding");
/* might really be multiple frames,
* but was treated as one for all purposes here */
ret = gst_audio_decoder_finish_frame (bdec, out_buf, 1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (dec, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
decode_error:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("Error decoding frame: %d", decode_ret), ret);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
gst_buffer_unref (out_buf);
goto done;
}
}
static gint
gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
AVCodec * in_plugin, guint8 * data, guint size,
GstBuffer ** outbuf, GstFlowReturn * ret)
{
gint len = -1;
gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
AVPacket packet;
AVFrame frame;
GST_DEBUG_OBJECT (ffmpegdec, "size: %d", size);
gst_avpacket_init (&packet, data, size);
memset (&frame, 0, sizeof (frame));
avcodec_get_frame_defaults (&frame);
len = avcodec_decode_audio4 (ffmpegdec->context, &frame, &have_data, &packet);
GST_DEBUG_OBJECT (ffmpegdec,
"Decode audio: len=%d, have_data=%d", len, have_data);
if (len >= 0 && have_data > 0) {
BufferInfo *buffer_info = frame.opaque;
gint nsamples, channels, byte_per_sample;
gsize output_size;
if (!gst_ffmpegauddec_negotiate (ffmpegdec, FALSE)) {
*outbuf = NULL;
*ret = GST_FLOW_NOT_NEGOTIATED;
len = -1;
goto beach;
}
channels = ffmpegdec->info.channels;
nsamples = frame.nb_samples;
byte_per_sample = ffmpegdec->info.finfo->width / 8;
/* frame.linesize[0] might contain padding, allocate only what's needed */
output_size = nsamples * byte_per_sample * channels;
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
if (buffer_info) {
*outbuf = buffer_info->buffer;
gst_buffer_unmap (buffer_info->buffer, &buffer_info->map);
g_slice_free (BufferInfo, buffer_info);
frame.opaque = NULL;
} else if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt)
&& channels > 1) {
gint i, j;
GstMapInfo minfo;
/* note: linesize[0] might contain padding, allocate only what's needed */
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_map (*outbuf, &minfo, GST_MAP_WRITE);
switch (ffmpegdec->info.finfo->width) {
case 8:{
guint8 *odata = minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint8 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
case 16:{
guint16 *odata = (guint16 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint16 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
case 32:{
guint32 *odata = (guint32 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint32 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
case 64:{
guint64 *odata = (guint64 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint64 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
gst_buffer_unmap (*outbuf, &minfo);
} else {
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_fill (*outbuf, 0, frame.data[0], output_size);
}
GST_DEBUG_OBJECT (ffmpegdec, "Buffer created. Size: %d", have_data);
/* Reorder channels to the GStreamer channel order */
if (ffmpegdec->needs_reorder) {
*outbuf = gst_buffer_make_writable (*outbuf);
gst_audio_buffer_reorder_channels (*outbuf, ffmpegdec->info.finfo->format,
ffmpegdec->info.channels, ffmpegdec->ffmpeg_layout,
ffmpegdec->info.position);
}
} else {
*outbuf = NULL;
}
beach:
GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
*ret, *outbuf, len);
return len;
}
static gint
gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
AVCodec * in_plugin, guint8 * data, guint size, gint * have_data,
GstBuffer ** outbuf, GstFlowReturn * ret)
{
gint len = -1;
AVPacket packet;
GST_DEBUG_OBJECT (ffmpegdec, "size: %d", size);
gst_avpacket_init (&packet, data, size);
len =
avcodec_decode_audio4 (ffmpegdec->context, ffmpegdec->frame, have_data,
&packet);
GST_DEBUG_OBJECT (ffmpegdec,
"Decode audio: len=%d, have_data=%d", len, *have_data);
if (len >= 0 && *have_data) {
gint nsamples, channels, byte_per_sample;
gsize output_size;
if (!gst_ffmpegauddec_negotiate (ffmpegdec, ffmpegdec->context,
ffmpegdec->frame, FALSE)) {
*outbuf = NULL;
*ret = GST_FLOW_NOT_NEGOTIATED;
len = -1;
goto beach;
}
channels = ffmpegdec->info.channels;
nsamples = ffmpegdec->frame->nb_samples;
byte_per_sample = ffmpegdec->info.finfo->width / 8;
/* ffmpegdec->frame->linesize[0] might contain padding, allocate only what's needed */
output_size = nsamples * byte_per_sample * channels;
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt)
&& channels > 1) {
gint i, j;
GstMapInfo minfo;
/* note: linesize[0] might contain padding, allocate only what's needed */
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_map (*outbuf, &minfo, GST_MAP_WRITE);
switch (ffmpegdec->info.finfo->width) {
case 8:{
guint8 *odata = minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint8 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
case 16:{
guint16 *odata = (guint16 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint16 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
case 32:{
guint32 *odata = (guint32 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint32 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
case 64:{
guint64 *odata = (guint64 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint64 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
gst_buffer_unmap (*outbuf, &minfo);
} else {
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_fill (*outbuf, 0, ffmpegdec->frame->data[0], output_size);
}
GST_DEBUG_OBJECT (ffmpegdec, "Buffer created. Size: %" G_GSIZE_FORMAT,
output_size);
/* Reorder channels to the GStreamer channel order */
if (ffmpegdec->needs_reorder) {
*outbuf = gst_buffer_make_writable (*outbuf);
gst_audio_buffer_reorder_channels (*outbuf, ffmpegdec->info.finfo->format,
ffmpegdec->info.channels, ffmpegdec->ffmpeg_layout,
ffmpegdec->info.position);
}
/* Mark corrupted frames as corrupted */
if (ffmpegdec->frame->flags & AV_FRAME_FLAG_CORRUPT)
GST_BUFFER_FLAG_SET (*outbuf, GST_BUFFER_FLAG_CORRUPTED);
} else {
*outbuf = NULL;
}
beach:
av_frame_unref (ffmpegdec->frame);
GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
*ret, *outbuf, len);
return len;
}
static GstFlowReturn
gst_sbc_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf)
{
GstSbcDec *dec = GST_SBC_DEC (audio_dec);
GstBuffer *outbuf = NULL;
GstMapInfo out_map;
GstMapInfo in_map;
gsize output_size;
guint num_frames, i;
/* no fancy draining */
if (G_UNLIKELY (buf == NULL))
return GST_FLOW_OK;
if (G_UNLIKELY (dec->frame_len == 0))
return GST_FLOW_NOT_NEGOTIATED;
gst_buffer_map (buf, &in_map, GST_MAP_READ);
if (G_UNLIKELY (in_map.size == 0))
goto done;
/* we assume all frames are of the same size, this is implied by the
* input caps applying to the whole input buffer, and the parser should
* also have made sure of that */
if (G_UNLIKELY (in_map.size % dec->frame_len != 0))
goto mixed_frames;
num_frames = in_map.size / dec->frame_len;
output_size = num_frames * dec->samples_per_frame * sizeof (gint16);
outbuf = gst_audio_decoder_allocate_output_buffer (audio_dec, output_size);
if (outbuf == NULL)
goto no_buffer;
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
for (i = 0; i < num_frames; ++i) {
gssize ret;
gsize written;
ret = sbc_decode (&dec->sbc, in_map.data + (i * dec->frame_len),
dec->frame_len, out_map.data + (i * dec->samples_per_frame * 2),
dec->samples_per_frame * 2, &written);
if (ret <= 0 || written != (dec->samples_per_frame * 2)) {
GST_WARNING_OBJECT (dec, "decoding error, ret = %" G_GSSIZE_FORMAT ", "
"written = %" G_GSSIZE_FORMAT, ret, written);
break;
}
}
gst_buffer_unmap (outbuf, &out_map);
if (i > 0)
gst_buffer_set_size (outbuf, i * dec->samples_per_frame * 2);
else
gst_buffer_replace (&outbuf, NULL);
done:
gst_buffer_unmap (buf, &in_map);
return gst_audio_decoder_finish_frame (audio_dec, outbuf, 1);
/* ERRORS */
mixed_frames:
{
GST_WARNING_OBJECT (dec, "inconsistent input data/frames, skipping");
goto done;
}
no_buffer:
{
GST_ERROR_OBJECT (dec, "could not allocate output buffer");
goto done;
}
}
static void
gst_droidadec_data_available (void *data, DroidMediaCodecData * encoded)
{
GstFlowReturn flow_ret;
GstDroidADec *dec = (GstDroidADec *) data;
GstAudioDecoder *decoder = GST_AUDIO_DECODER (dec);
GstBuffer *out;
GstMapInfo info;
GST_DEBUG_OBJECT (dec, "data available of size %d", encoded->data.size);
GST_AUDIO_DECODER_STREAM_LOCK (decoder);
if (G_UNLIKELY (dec->downstream_flow_ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "not handling data in error state: %s",
gst_flow_get_name (dec->downstream_flow_ret));
flow_ret = dec->downstream_flow_ret;
gst_audio_decoder_finish_frame (decoder, NULL, 1);
goto out;
}
if (G_UNLIKELY (gst_audio_decoder_get_audio_info (GST_AUDIO_DECODER
(dec))->finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
DroidMediaCodecMetaData md;
DroidMediaRect crop; /* TODO: get rid of that */
GstAudioInfo info;
memset (&md, 0x0, sizeof (md));
droid_media_codec_get_output_info (dec->codec, &md, &crop);
GST_INFO_OBJECT (dec, "output rate=%d, output channels=%d", md.sample_rate,
md.channels);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, md.sample_rate,
md.channels, NULL);
if (!gst_audio_decoder_set_output_format (decoder, &info)) {
flow_ret = GST_FLOW_ERROR;
goto out;
}
dec->info = gst_audio_decoder_get_audio_info (GST_AUDIO_DECODER (dec));
}
out = gst_audio_decoder_allocate_output_buffer (decoder, encoded->data.size);
gst_buffer_map (out, &info, GST_MAP_READWRITE);
orc_memcpy (info.data, encoded->data.data, encoded->data.size);
gst_buffer_unmap (out, &info);
// GST_WARNING_OBJECT (dec, "bpf %d, bps %d", dec->info->bpf, GST_AUDIO_INFO_BPS(dec->info));
if (dec->spf == -1 || (encoded->data.size == dec->spf * dec->info->bpf
&& gst_adapter_available (dec->adapter) == 0)) {
/* fast path. no need for anything */
goto push;
}
gst_adapter_push (dec->adapter, out);
if (gst_adapter_available (dec->adapter) >= dec->spf * dec->info->bpf) {
out = gst_adapter_take_buffer (dec->adapter, dec->spf * dec->info->bpf);
} else {
flow_ret = GST_FLOW_OK;
goto out;
}
push:
GST_DEBUG_OBJECT (dec, "pushing %d bytes out", gst_buffer_get_size (out));
flow_ret = gst_audio_decoder_finish_frame (decoder, out, 1);
if (flow_ret == GST_FLOW_OK || flow_ret == GST_FLOW_FLUSHING) {
goto out;
} else if (flow_ret == GST_FLOW_EOS) {
GST_INFO_OBJECT (dec, "eos");
} else if (flow_ret < GST_FLOW_OK) {
GST_ELEMENT_ERROR (dec, STREAM, FAILED,
("Internal data stream error."), ("stream stopped, reason %s",
gst_flow_get_name (flow_ret)));
}
out:
dec->downstream_flow_ret = flow_ret;
GST_AUDIO_DECODER_STREAM_UNLOCK (decoder);
}
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);
}
}
