static void
gst_amc_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstAmcAudioDec *self;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Resetting decoder");
if (!self->started) {
GST_DEBUG_OBJECT (self, "Codec not started yet");
return;
}
self->flushing = TRUE;
gst_amc_codec_flush (self->codec);
/* Wait until the srcpad loop is finished,
* unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
GST_PAD_STREAM_LOCK (GST_AUDIO_DECODER_SRC_PAD (self));
GST_PAD_STREAM_UNLOCK (GST_AUDIO_DECODER_SRC_PAD (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
self->flushing = FALSE;
/* Start the srcpad loop again */
self->last_upstream_ts = 0;
self->eos = FALSE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
GST_DEBUG_OBJECT (self, "Reset decoder");
}
static gboolean
gst_omx_audio_dec_stop (GstAudioDecoder * decoder)
{
GstOMXAudioDec *self;
self = GST_OMX_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Stopping decoder");
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
if (gst_omx_component_get_state (self->dec, 0) > OMX_StateIdle)
gst_omx_component_set_state (self->dec, OMX_StateIdle);
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
gst_omx_component_get_state (self->dec, 5 * GST_SECOND);
gst_buffer_replace (&self->codec_data, NULL);
GST_DEBUG_OBJECT (self, "Stopped decoder");
return TRUE;
}
static void
gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
OMX_ERRORTYPE err = OMX_ErrorNone;
GST_DEBUG_OBJECT (self, "Flushing decoder");
if (gst_omx_component_get_state (self->dec, 0) == OMX_StateLoaded)
return;
/* 0) Pause the components */
if (gst_omx_component_get_state (self->dec, 0) == OMX_StateExecuting) {
gst_omx_component_set_state (self->dec, OMX_StatePause);
gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
}
/* 1) Wait until the srcpad loop is stopped,
* unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
GST_DEBUG_OBJECT (self, "Flushing -- task stopped");
GST_AUDIO_DECODER_STREAM_LOCK (self);
/* 2) Flush the ports */
GST_DEBUG_OBJECT (self, "flushing ports");
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
/* 3) Resume components */
gst_omx_component_set_state (self->dec, OMX_StateExecuting);
gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
/* 4) Unset flushing to allow ports to accept data again */
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
err = gst_omx_port_populate (self->dec_out_port);
if (err != OMX_ErrorNone) {
GST_WARNING_OBJECT (self, "Failed to populate output port: %s (0x%08x)",
gst_omx_error_to_string (err), err);
}
/* Reset our state */
self->last_upstream_ts = 0;
self->downstream_flow_ret = GST_FLOW_OK;
self->started = FALSE;
GST_DEBUG_OBJECT (self, "Flush finished");
}
static void
gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
{
GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
GstStructure *s;
GstAudioInfo info;
if (caps) {
caps = gst_caps_truncate (caps);
caps = gst_caps_make_writable (caps);
s = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", 48000);
gst_structure_get_int (s, "rate", &dec->sample_rate);
gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
gst_structure_get_int (s, "channels", &dec->n_channels);
gst_caps_unref (caps);
} else {
dec->sample_rate = 48000;
}
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
dec->sample_rate);
/* pass valid order to audio info */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
}
/* set up source format */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
/* but we still need the opus order for later reordering */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
} else {
dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
}
dec->info = info;
}
static GstCaps *
gst_opus_dec_negotiate (GstOpusDec * dec)
{
GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
GstStructure *s;
caps = gst_caps_make_writable (caps);
gst_caps_truncate (caps);
s = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", 48000);
gst_structure_get_int (s, "rate", &dec->sample_rate);
gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
gst_structure_get_int (s, "channels", &dec->n_channels);
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
dec->sample_rate);
return caps;
}
static gboolean
gst_amc_audio_dec_stop (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Stopping decoder");
self->flushing = TRUE;
if (self->started) {
gst_amc_codec_flush (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_amc_codec_stop (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
self->started = FALSE;
}
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
memset (self->positions, 0, sizeof (self->positions));
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
g_list_foreach (self->codec_datas, (GFunc) g_free, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->drained = TRUE;
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
GST_DEBUG_OBJECT (self, "Stopped decoder");
return TRUE;
}
static gboolean
gst_amc_audio_dec_stop (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Stopping decoder");
self->flushing = TRUE;
if (self->started) {
gst_amc_codec_flush (self->codec);
gst_amc_codec_stop (self->codec);
self->started = FALSE;
if (self->input_buffers)
gst_amc_codec_free_buffers (self->input_buffers, self->n_input_buffers);
self->input_buffers = NULL;
if (self->output_buffers)
gst_amc_codec_free_buffers (self->output_buffers, self->n_output_buffers);
self->output_buffers = NULL;
}
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
memset (self->positions, 0, sizeof (self->positions));
g_list_foreach (self->codec_datas, (GFunc) g_free, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->eos = FALSE;
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
GST_DEBUG_OBJECT (self, "Stopped decoder");
return TRUE;
}
static GstFlowReturn
gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
gint i, fpp;
SpeexBits *bits;
GstMapInfo map;
if (!dec->frame_duration)
goto not_negotiated;
if (G_LIKELY (gst_buffer_get_size (buf))) {
/* send data to the bitstream */
gst_buffer_map (buf, &map, GST_MAP_READ);
speex_bits_read_from (&dec->bits, (gchar *) map.data, map.size);
gst_buffer_unmap (buf, &map);
fpp = dec->header->frames_per_packet;
bits = &dec->bits;
GST_DEBUG_OBJECT (dec, "received buffer of size %" G_GSIZE_FORMAT
", fpp %d, %d bits", map.size, fpp, speex_bits_remaining (bits));
} else {
/* FIXME ? actually consider how much concealment is needed */
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
fpp = dec->header->frames_per_packet;
bits = NULL;
}
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
for (i = 0; i < fpp; i++) {
GstBuffer *outbuf;
gboolean corrupted = FALSE;
gint ret;
GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp,
bits ? speex_bits_remaining (bits) : -1);
#if 0
res =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
#endif
/* FIXME, we can use a bufferpool because we have fixed size buffers. We
* could also use an allocator */
outbuf =
gst_buffer_new_allocate (NULL,
dec->frame_size * dec->header->nb_channels * 2, NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
ret = speex_decode_int (dec->state, bits, (spx_int16_t *) map.data);
if (ret == -1) {
/* uh? end of stream */
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
corrupted = TRUE;
} else if (ret == -2) {
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
corrupted = TRUE;
}
if (bits && speex_bits_remaining (bits) < 0) {
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
corrupted = TRUE;
}
if (dec->header->nb_channels == 2)
speex_decode_stereo_int ((spx_int16_t *) map.data, dec->frame_size,
dec->stereo);
gst_buffer_unmap (outbuf, &map);
if (!corrupted) {
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
} else {
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
}
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
break;
}
}
return res;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
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 gboolean
gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
{
GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
GstStructure *s;
GstAudioInfo info;
if (caps) {
gint rate = dec->sample_rate, channels = dec->n_channels;
GstCaps *constraint, *inter;
constraint = gst_caps_from_string ("audio/x-raw");
if (dec->n_channels <= 2) { /* including 0 */
gst_caps_set_simple (constraint, "channels", GST_TYPE_INT_RANGE, 1, 2,
NULL);
} else {
gst_caps_set_simple (constraint, "channels", G_TYPE_INT, dec->n_channels,
NULL);
}
inter = gst_caps_intersect (caps, constraint);
gst_caps_unref (constraint);
if (gst_caps_is_empty (inter)) {
GST_DEBUG_OBJECT (dec, "Empty intersection, failed to negotiate");
gst_caps_unref (inter);
gst_caps_unref (caps);
return FALSE;
}
inter = gst_caps_truncate (inter);
s = gst_caps_get_structure (inter, 0);
rate = dec->sample_rate > 0 ? dec->sample_rate : 48000;
gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate);
gst_structure_get_int (s, "rate", &rate);
channels = dec->n_channels > 0 ? dec->n_channels : 2;
gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
gst_structure_get_int (s, "channels", &channels);
gst_caps_unref (inter);
dec->sample_rate = rate;
dec->n_channels = channels;
gst_caps_unref (caps);
}
if (dec->n_channels == 0) {
GST_DEBUG_OBJECT (dec, "Using a default of 2 channels");
dec->n_channels = 2;
pos = NULL;
}
if (dec->sample_rate == 0) {
GST_DEBUG_OBJECT (dec, "Using a default of 48kHz sample rate");
dec->sample_rate = 48000;
}
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
dec->sample_rate);
/* pass valid order to audio info */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
}
/* set up source format */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
/* but we still need the opus order for later reordering */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
} else {
dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
}
dec->info = info;
return TRUE;
}
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_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
GstDtsDec *dts;
gint channels, i, num_blocks;
gboolean need_renegotiation = FALSE;
guint8 *data;
gsize size;
GstMapInfo map;
gint chans;
gint length = 0, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_OK;
GstBuffer *outbuf;
dts = GST_DTSDEC (bdec);
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
/* parsed stuff already, so this should work out fine */
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
g_assert (size >= 7);
bit_rate = dts->bit_rate;
sample_rate = dts->sample_rate;
flags = 0;
length = dca_syncinfo (dts->state, data, &flags, &sample_rate, &bit_rate,
&frame_length);
g_assert (length == size);
if (flags != dts->prev_flags) {
dts->prev_flags = flags;
dts->flag_update = TRUE;
}
/* go over stream properties, renegotiate or update streaminfo if needed */
if (dts->sample_rate != sample_rate) {
need_renegotiation = TRUE;
dts->sample_rate = sample_rate;
}
if (flags) {
dts->stream_channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
}
if (bit_rate != dts->bit_rate) {
dts->bit_rate = bit_rate;
gst_dtsdec_update_streaminfo (dts);
}
/* If we haven't had an explicit number of channels chosen through properties
* at this point, choose what to downmix to now, based on what the peer will
* accept - this allows a52dec to do downmixing in preference to a
* downstream element such as audioconvert.
* FIXME: Add the property back in for forcing output channels.
*/
if (dts->request_channels != DCA_CHANNEL) {
flags = dts->request_channels;
} else if (dts->flag_update) {
GstCaps *caps;
dts->flag_update = FALSE;
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dts));
if (caps && gst_caps_get_size (caps) > 0) {
GstCaps *copy = gst_caps_copy_nth (caps, 0);
GstStructure *structure = gst_caps_get_structure (copy, 0);
gint channels;
const int dts_channels[6] = {
DCA_MONO,
DCA_STEREO,
DCA_STEREO | DCA_LFE,
DCA_2F2R,
DCA_2F2R | DCA_LFE,
DCA_3F2R | DCA_LFE,
};
/* Prefer the original number of channels, but fixate to something
* preferred (first in the caps) downstream if possible.
*/
gst_structure_fixate_field_nearest_int (structure, "channels",
flags ? gst_dtsdec_channels (flags, NULL) : 6);
gst_structure_get_int (structure, "channels", &channels);
if (channels <= 6)
flags = dts_channels[channels - 1];
else
flags = dts_channels[5];
gst_caps_unref (copy);
} else if (flags) {
flags = dts->stream_channels;
} else {
flags = DCA_3F2R | DCA_LFE;
}
if (caps)
gst_caps_unref (caps);
} else {
flags = dts->using_channels;
}
/* process */
flags |= DCA_ADJUST_LEVEL;
dts->level = 1;
if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
gst_buffer_unmap (buffer, &map);
GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL),
("dts_frame error"), result);
goto exit;
}
gst_buffer_unmap (buffer, &map);
channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
if (dts->using_channels != channels) {
need_renegotiation = TRUE;
dts->using_channels = channels;
}
/* negotiate if required */
if (need_renegotiation) {
GST_DEBUG_OBJECT (dts,
"dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
dts->sample_rate, dts->stream_channels, dts->using_channels);
if (!gst_dtsdec_renegotiate (dts))
goto failed_negotiation;
}
if (dts->dynamic_range_compression == FALSE) {
dca_dynrng (dts->state, NULL, NULL);
}
flags &= (DCA_CHANNEL_MASK | DCA_LFE);
chans = gst_dtsdec_channels (flags, NULL);
if (!chans)
goto invalid_flags;
/* handle decoded data, one block is 256 samples */
num_blocks = dca_blocks_num (dts->state);
outbuf =
gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
data = map.data;
size = map.size;
{
guint8 *ptr = data;
for (i = 0; i < num_blocks; i++) {
if (dca_block (dts->state)) {
/* also marks discont */
GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL),
("error decoding block %d", i), result);
if (result != GST_FLOW_OK)
goto exit;
} else {
gint n, c;
gint *reorder_map = dts->channel_reorder_map;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) ptr)[n * chans + reorder_map[c]] =
dts->samples[c * 256 + n];
}
}
}
ptr += 256 * chans * (SAMPLE_WIDTH / 8);
}
}
gst_buffer_unmap (outbuf, &map);
result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
exit:
return result;
/* ERRORS */
failed_negotiation:
{
GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
invalid_flags:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Invalid channel flags: %d", flags));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data = GST_BUFFER_DATA (buf);
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
if (!gst_opus_header_is_id_header (buf)) {
GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
return GST_FLOW_ERROR;
}
if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
return GST_FLOW_ERROR;
}
dec->n_channels = data[9];
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
dec->r128_gain = GST_READ_UINT16_LE (data + 16);
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
dec->channel_mapping_family = data[18];
if (dec->channel_mapping_family == 0) {
/* implicit mapping */
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
dec->n_streams = dec->n_stereo_streams = 1;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
} else {
dec->n_streams = data[19];
dec->n_stereo_streams = data[20];
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
pos = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, dec->n_channels);
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
}
caps = gst_opus_dec_negotiate (dec);
if (pos) {
GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (dec->n_channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gint size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GstCaps *caps;
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
caps = gst_opus_dec_negotiate (dec);
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
/* 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_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_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 %u",
GST_BUFFER_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_SIZE (dec->last_buffer) > 0) ? dec->last_buffer : buffer;
if (buf && GST_BUFFER_SIZE (buf) > 0) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG_OBJECT (dec, "Using buffer of size %u", 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;
res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, packet_size,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
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);
}
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_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_SIZE (outbuf) -= skip * 2 * dec->n_channels;
GST_BUFFER_DATA (outbuf) += skip * 2 * dec->n_channels;
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_SIZE (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
}
/* 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) {
unsigned int i, nsamples = GST_BUFFER_SIZE (outbuf) / 2;
double volume = dec->r128_gain_volume;
gint16 *samples = (gint16 *) GST_BUFFER_DATA (outbuf);
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
}
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;
}
static GstFlowReturn
gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
gint i, fpp;
guint size;
guint8 *data;
SpeexBits *bits;
if (!dec->frame_duration)
goto not_negotiated;
if (G_LIKELY (GST_BUFFER_SIZE (buf))) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* send data to the bitstream */
speex_bits_read_from (&dec->bits, (char *) data, size);
fpp = dec->header->frames_per_packet;
bits = &dec->bits;
GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d, %d bits",
size, fpp, speex_bits_remaining (bits));
} else {
/* FIXME ? actually consider how much concealment is needed */
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
fpp = dec->header->frames_per_packet;
bits = NULL;
}
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
for (i = 0; i < fpp; i++) {
GstBuffer *outbuf;
gint16 *out_data;
gint ret;
GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp,
bits ? speex_bits_remaining (bits) : -1);
res =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
ret = speex_decode_int (dec->state, bits, out_data);
if (ret == -1) {
/* uh? end of stream */
if (fpp == 0 && speex_bits_remaining (bits) < 8) {
/* if we did not know how many frames to expect, then we get this
at the end if there are leftover bits to pad to the next byte */
GST_DEBUG_OBJECT (dec, "Discarding leftover bits");
} else {
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
}
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
} else if (ret == -2) {
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
}
if (bits && speex_bits_remaining (bits) < 0) {
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
}
if (dec->header->nb_channels == 2)
speex_decode_stereo_int (out_data, dec->frame_size, dec->stereo);
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));
break;
}
}
return res;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static gboolean
gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstAmcAudioDec *self;
GstStructure *s;
GstAmcFormat *format;
const gchar *mime;
gboolean is_format_change = FALSE;
gboolean needs_disable = FALSE;
gchar *format_string;
gint rate, channels;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
/* Check if the caps change is a real format change or if only irrelevant
* parts of the caps have changed or nothing at all.
*/
is_format_change |= (!self->input_caps
|| !gst_caps_is_equal (self->input_caps, caps));
needs_disable = self->started;
/* If the component is not started and a real format change happens
* we have to restart the component. If no real format change
* happened we can just exit here.
*/
if (needs_disable && !is_format_change) {
/* Framerate or something minor changed */
self->input_caps_changed = TRUE;
GST_DEBUG_OBJECT (self,
"Already running and caps did not change the format");
return TRUE;
}
if (needs_disable && is_format_change) {
gst_amc_audio_dec_drain (self);
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_amc_audio_dec_stop (GST_AUDIO_DECODER (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_audio_dec_close (GST_AUDIO_DECODER (self));
if (!gst_amc_audio_dec_open (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to open codec again");
return FALSE;
}
if (!gst_amc_audio_dec_start (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to start codec again");
}
}
/* srcpad task is not running at this point */
mime = caps_to_mime (caps);
if (!mime) {
GST_ERROR_OBJECT (self, "Failed to convert caps to mime");
return FALSE;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Failed to get rate/channels");
return FALSE;
}
format = gst_amc_format_new_audio (mime, rate, channels, &err);
if (!format) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
/* FIXME: These buffers needs to be valid until the codec is stopped again */
g_list_foreach (self->codec_datas, (GFunc) gst_buffer_unref, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
if (gst_structure_has_field (s, "codec_data")) {
const GValue *h = gst_structure_get_value (s, "codec_data");
GstBuffer *codec_data = gst_value_get_buffer (h);
GstMapInfo minfo;
guint8 *data;
gst_buffer_map (codec_data, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (codec_data, &minfo);
} else if (gst_structure_has_field (s, "streamheader")) {
const GValue *sh = gst_structure_get_value (s, "streamheader");
gint nsheaders = gst_value_array_get_size (sh);
GstBuffer *buf;
const GValue *h;
gint i, j;
gchar *fname;
GstMapInfo minfo;
guint8 *data;
for (i = 0, j = 0; i < nsheaders; i++) {
h = gst_value_array_get_value (sh, i);
buf = gst_value_get_buffer (h);
if (strcmp (mime, "audio/vorbis") == 0) {
guint8 header_type;
gst_buffer_extract (buf, 0, &header_type, 1);
/* Only use the identification and setup packets */
if (header_type != 0x01 && header_type != 0x05)
continue;
}
fname = g_strdup_printf ("csd-%d", j);
gst_buffer_map (buf, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, fname, data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (buf, &minfo);
g_free (fname);
j++;
}
}
format_string = gst_amc_format_to_string (format, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
GST_DEBUG_OBJECT (self, "Configuring codec with format: %s",
GST_STR_NULL (format_string));
g_free (format_string);
if (!gst_amc_codec_configure (self->codec, format, 0, &err)) {
GST_ERROR_OBJECT (self, "Failed to configure codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
gst_amc_format_free (format);
if (!gst_amc_codec_start (self->codec, &err)) {
GST_ERROR_OBJECT (self, "Failed to start codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->spf = -1;
/* TODO: Implement for other codecs too */
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion = -1;
gst_structure_get_int (s, "mpegversion", &mpegversion);
if (mpegversion == 1) {
gint layer = -1, mpegaudioversion = -1;
gst_structure_get_int (s, "layer", &layer);
gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion);
if (layer == 1)
self->spf = 384;
else if (layer == 2)
self->spf = 1152;
else if (layer == 3 && mpegaudioversion != -1)
self->spf = (mpegaudioversion == 1 ? 1152 : 576);
}
}
self->started = TRUE;
self->input_caps_changed = TRUE;
/* Start the srcpad loop again */
self->flushing = FALSE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
return TRUE;
}
static GstFlowReturn
gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
gint width, depth, i, j, max;
gint32 *dec_data = NULL;
guint8 *out_data;
GstMapInfo map, omap;
dec = GST_WAVPACK_DEC (bdec);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
gst_buffer_map (buf, &map, GST_MAP_READ);
/* check input, we only accept framed input with complete chunks */
if (map.size < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, map.data))
goto invalid_header;
if (map.size < wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed;
dec->wv_id.buffer = map.data;
dec->wv_id.length = map.size;
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
* was already one (i.e. caps were set on the srcpad) check whether
* the new one has the same caps */
if (!dec->context) {
gchar error_msg[80];
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
/* expect this to work */
if (!dec->context) {
GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
goto context_failed;
}
}
g_assert (dec->context != NULL);
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
(dec->channel_mask != WavpackGetChannelMask (dec->context));
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
format_changed) {
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
channel_mask = WavpackGetChannelMask (dec->context);
if (channel_mask == 0)
channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
dec->channel_mask = channel_mask;
gst_wavpack_dec_negotiate (dec);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
/* alloc output buffer */
dec_data = g_malloc (4 * wph.block_samples * dec->channels);
/* decode */
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
outbuf = gst_buffer_new_and_alloc (unpacked_size);
/* legacy; pass along offset, whatever that might entail */
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = omap.data;
width = dec->width;
depth = dec->depth;
max = dec->channels * wph.block_samples;
if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
}
} else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
if (width != depth) {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ =
(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
}
} else {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buf, &map);
buf = NULL;
g_free (dec_data);
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out:
if (buf)
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
ret = GST_FLOW_ERROR;
goto out;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
ret = GST_FLOW_ERROR;
goto out;
}
context_failed:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
("error creating Wavpack context"), ret);
goto out;
}
decode_error:
{
const gchar *reason = "unknown";
if (dec->context) {
reason = WavpackGetErrorMessage (dec->context);
} else {
reason = "couldn't create decoder context";
}
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret);
g_free (dec_data);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
goto out;
}
}
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, newresult;
gint 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->vi.channels * vd->width;
GST_LOG_OBJECT (vd, "%d samples ready for reading, size %d", sample_count,
size);
/* alloc buffer for it */
result =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (vd),
GST_BUFFER_OFFSET_NONE, size,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (vd)), &out);
if (G_UNLIKELY (result != GST_FLOW_OK))
goto done;
/* get samples ready for reading now, should be sample_count */
#ifdef USE_TREMOLO
pcm = GST_BUFFER_DATA (out);
if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, pcm, sample_count) !=
sample_count))
#else
if (G_UNLIKELY (vorbis_synthesis_pcmout (&vd->vd, &pcm) != sample_count))
#endif
goto wrong_samples;
#ifndef USE_TREMOLO
/* copy samples in buffer */
vd->copy_samples ((vorbis_sample_t *) GST_BUFFER_DATA (out), pcm,
sample_count, vd->vi.channels, vd->width);
#endif
GST_LOG_OBJECT (vd, "setting output size to %d", size);
GST_BUFFER_SIZE (out) = size;
done:
/* whether or not data produced, consume one frame and advance time */
newresult = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), out, 1);
if (G_UNLIKELY (newresult != GST_FLOW_OK))
result = newresult;
#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 GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
gint width = GST_VORBIS_DEC_DEFAULT_SAMPLE_WIDTH;
switch (vd->vi.channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
pos = gst_vorbis_channel_positions[vd->vi.channels - 1];
break;
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, vd->vi.channels);
GST_ELEMENT_WARNING (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < vd->vi.channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
/* negotiate width with downstream */
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (vd));
if (caps) {
if (!gst_caps_is_empty (caps)) {
GstStructure *s;
s = gst_caps_get_structure (caps, 0);
/* template ensures 16 or 32 */
gst_structure_get_int (s, "width", &width);
GST_INFO_OBJECT (vd, "using %s with %d channels and %d bit audio depth",
gst_structure_get_name (s), vd->vi.channels, width);
}
gst_caps_unref (caps);
}
vd->width = width >> 3;
/* select a copy_samples function, this way we can have specialized versions
* for mono/stereo and avoid the depth switch in tremor case */
vd->copy_samples = get_copy_sample_func (vd->vi.channels, vd->width);
caps =
gst_caps_copy (gst_pad_get_pad_template_caps
(GST_AUDIO_DECODER_SRC_PAD (vd)));
gst_caps_set_simple (caps, "rate", G_TYPE_INT, vd->vi.rate, "channels",
G_TYPE_INT, vd->vi.channels, "width", G_TYPE_INT, width, NULL);
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (vd->vi.channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (vd), caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR;
GstOMXAudioDec *self;
GstOMXPort *port;
GstOMXBuffer *buf;
GstBuffer *codec_data = NULL;
guint offset = 0;
GstClockTime timestamp, duration;
OMX_ERRORTYPE err;
GstMapInfo minfo;
self = GST_OMX_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Handling frame");
if (self->downstream_flow_ret != GST_FLOW_OK) {
return self->downstream_flow_ret;
}
if (!self->started) {
GST_DEBUG_OBJECT (self, "Starting task");
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL);
}
if (inbuf == NULL)
return gst_omx_audio_dec_drain (self);
/* Make sure to keep a reference to the input here,
* it can be unreffed from the other thread if
* finish_frame() is called */
gst_buffer_ref (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
port = self->dec_in_port;
gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
while (offset < minfo.size) {
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
acq_ret = gst_omx_port_acquire_buffer (port, &buf);
if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto component_error;
} else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto flushing;
} else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
/* Reallocate all buffers */
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto reconfigure_error;
}
/* Now get a new buffer and fill it */
GST_AUDIO_DECODER_STREAM_LOCK (self);
continue;
}
GST_AUDIO_DECODER_STREAM_LOCK (self);
g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) {
gst_omx_port_release_buffer (port, buf);
goto full_buffer;
}
if (self->downstream_flow_ret != GST_FLOW_OK) {
gst_omx_port_release_buffer (port, buf);
goto flow_error;
}
if (self->codec_data) {
GST_DEBUG_OBJECT (self, "Passing codec data to the component");
codec_data = self->codec_data;
if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <
gst_buffer_get_size (codec_data)) {
gst_omx_port_release_buffer (port, buf);
goto too_large_codec_data;
}
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_CODECCONFIG;
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
buf->omx_buf->nFilledLen = gst_buffer_get_size (codec_data);
gst_buffer_extract (codec_data, 0,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
buf->omx_buf->nTimeStamp =
gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
else
buf->omx_buf->nTimeStamp = 0;
buf->omx_buf->nTickCount = 0;
self->started = TRUE;
err = gst_omx_port_release_buffer (port, buf);
gst_buffer_replace (&self->codec_data, NULL);
if (err != OMX_ErrorNone)
goto release_error;
/* Acquire new buffer for the actual frame */
continue;
}
/* Now handle the frame */
GST_DEBUG_OBJECT (self, "Passing frame offset %d to the component", offset);
/* Copy the buffer content in chunks of size as requested
* by the port */
buf->omx_buf->nFilledLen =
MIN (minfo.size - offset,
buf->omx_buf->nAllocLen - buf->omx_buf->nOffset);
gst_buffer_extract (inbuf, offset,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
if (timestamp != GST_CLOCK_TIME_NONE) {
buf->omx_buf->nTimeStamp =
gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND);
self->last_upstream_ts = timestamp;
} else {
buf->omx_buf->nTimeStamp = 0;
}
if (duration != GST_CLOCK_TIME_NONE && offset == 0) {
buf->omx_buf->nTickCount =
gst_util_uint64_scale (duration, OMX_TICKS_PER_SECOND, GST_SECOND);
self->last_upstream_ts += duration;
} else {
buf->omx_buf->nTickCount = 0;
}
if (offset == 0)
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_SYNCFRAME;
/* TODO: Set flags
* - OMX_BUFFERFLAG_DECODEONLY for buffers that are outside
* the segment
*/
offset += buf->omx_buf->nFilledLen;
if (offset == minfo.size)
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
self->started = TRUE;
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
}
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_DEBUG_OBJECT (self, "Passed frame to component");
return self->downstream_flow_ret;
full_buffer:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Got OpenMAX buffer with no free space (%p, %u/%u)", buf,
(guint) buf->omx_buf->nOffset, (guint) buf->omx_buf->nAllocLen));
return GST_FLOW_ERROR;
}
flow_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
}
too_large_codec_data:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
("codec_data larger than supported by OpenMAX port "
"(%" G_GSIZE_FORMAT " > %u)", gst_buffer_get_size (codec_data),
(guint) self->dec_in_port->port_def.nBufferSize));
return GST_FLOW_ERROR;
}
component_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->dec),
gst_omx_component_get_last_error (self->dec)));
return GST_FLOW_ERROR;
}
flushing:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
return GST_FLOW_FLUSHING;
}
reconfigure_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure input port"));
return GST_FLOW_ERROR;
}
release_error:
{
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase input buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
return GST_FLOW_ERROR;
}
}
static gboolean
gst_atdec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
OSStatus status;
AudioStreamBasicDescription input_format = { 0 };
AudioStreamBasicDescription output_format = { 0 };
GstAudioInfo output_info = { 0 };
AudioChannelLayout output_layout = { 0 };
GstCaps *output_caps;
GstATDec *atdec = GST_ATDEC (decoder);
GST_DEBUG_OBJECT (atdec, "set_format");
if (atdec->queue)
gst_atdec_destroy_queue (atdec, TRUE);
/* configure input_format from caps */
gst_caps_to_at_format (caps, &input_format);
/* negotiate output caps */
output_caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
if (!output_caps)
output_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
output_caps = gst_caps_fixate (output_caps);
gst_caps_set_simple (output_caps,
"rate", G_TYPE_INT, (int) input_format.mSampleRate,
"channels", G_TYPE_INT, input_format.mChannelsPerFrame, NULL);
/* configure output_format from caps */
gst_caps_to_at_format (output_caps, &output_format);
/* set the format we want to negotiate downstream */
gst_audio_info_from_caps (&output_info, output_caps);
gst_audio_info_set_format (&output_info,
output_format.mFormatFlags & kLinearPCMFormatFlagIsSignedInteger ?
GST_AUDIO_FORMAT_S16LE : GST_AUDIO_FORMAT_F32LE,
output_format.mSampleRate, output_format.mChannelsPerFrame, NULL);
gst_audio_decoder_set_output_format (decoder, &output_info);
gst_caps_unref (output_caps);
status = AudioQueueNewOutput (&input_format, gst_atdec_buffer_emptied,
atdec, NULL, NULL, 0, &atdec->queue);
if (status)
goto create_queue_error;
/* FIXME: figure out how to map this properly */
if (output_format.mChannelsPerFrame == 1)
output_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Mono;
else
output_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
status = AudioQueueSetOfflineRenderFormat (atdec->queue,
&output_format, &output_layout);
if (status)
goto set_format_error;
status = AudioQueueStart (atdec->queue, NULL);
if (status)
goto start_error;
return TRUE;
create_queue_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueNewOutput returned error: %d", status));
return FALSE;
set_format_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueSetOfflineRenderFormat returned error: %d", status));
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
start_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueStart returned error: %d", status));
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
}
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 FLAC__StreamDecoderWriteStatus
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
const FLAC__int32 * const buffer[])
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
guint depth = frame->header.bits_per_sample;
guint width, gdepth;
guint sample_rate = frame->header.sample_rate;
guint channels = frame->header.channels;
guint samples = frame->header.blocksize;
guint j, i;
GstMapInfo map;
gboolean caps_changed;
GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);
if (depth == 0) {
if (flacdec->depth < 4 || flacdec->depth > 32) {
GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
flacdec->depth);
ret = GST_FLOW_ERROR;
goto done;
}
depth = flacdec->depth;
}
switch (depth) {
case 8:
gdepth = width = 8;
break;
case 12:
case 16:
gdepth = width = 16;
break;
case 20:
case 24:
gdepth = 24;
width = 32;
break;
case 32:
gdepth = width = 32;
break;
default:
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
ret = GST_FLOW_ERROR;
goto done;
}
if (sample_rate == 0) {
if (flacdec->info.rate != 0) {
sample_rate = flacdec->info.rate;
} else {
GST_ERROR_OBJECT (flacdec, "unknown sample rate");
ret = GST_FLOW_ERROR;
goto done;
}
}
caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info))
|| (width != GST_AUDIO_INFO_WIDTH (&flacdec->info))
|| (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info))
|| (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info));
if (caps_changed
|| !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate,
channels);
gst_audio_info_set_format (&flacdec->info,
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
sample_rate, channels, NULL);
memcpy (flacdec->info.position,
channel_positions[flacdec->info.channels - 1],
sizeof (GstAudioChannelPosition) * flacdec->info.channels);
gst_audio_channel_positions_to_valid_order (flacdec->info.position,
flacdec->info.channels);
/* Note: we create the inverse reordering map here */
gst_audio_get_channel_reorder_map (flacdec->info.channels,
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
flacdec->channel_reorder_map);
flacdec->depth = depth;
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec),
&flacdec->info);
}
outbuf =
gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
if (width == 8) {
gint8 *outbuffer = (gint8 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint8) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint8) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint16) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint16) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 32) {
gint32 *outbuffer = (gint32 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint32) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint32) buffer[reorder_map[j]][i];
}
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &map);
GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);
if (flacdec->error_count)
flacdec->error_count--;
ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
}
done:
/* we act on the flow return value later in the handle_frame function, as we
* don't want to mess up the internal decoder state by returning ABORT when
* the error is in fact non-fatal (like a pad in flushing mode) and we want
* to continue later. So just pretend everything's dandy and act later. */
flacdec->last_flow = ret;
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
static GstFlowReturn
gst_speex_dec_parse_header (GstSpeexDec * dec, GstBuffer * buf)
{
GstCaps *caps;
/* get the header */
dec->header = speex_packet_to_header ((char *) GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
if (!dec->header)
goto no_header;
if (dec->header->mode >= SPEEX_NB_MODES || dec->header->mode < 0)
goto mode_too_old;
dec->mode = speex_lib_get_mode (dec->header->mode);
/* initialize the decoder */
dec->state = speex_decoder_init (dec->mode);
if (!dec->state)
goto init_failed;
speex_decoder_ctl (dec->state, SPEEX_SET_ENH, &dec->enh);
speex_decoder_ctl (dec->state, SPEEX_GET_FRAME_SIZE, &dec->frame_size);
if (dec->header->nb_channels != 1) {
dec->stereo = speex_stereo_state_init ();
dec->callback.callback_id = SPEEX_INBAND_STEREO;
dec->callback.func = speex_std_stereo_request_handler;
dec->callback.data = dec->stereo;
speex_decoder_ctl (dec->state, SPEEX_SET_HANDLER, &dec->callback);
}
speex_decoder_ctl (dec->state, SPEEX_SET_SAMPLING_RATE, &dec->header->rate);
dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
GST_SECOND, dec->header->rate);
speex_bits_init (&dec->bits);
/* set caps */
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->header->rate,
"channels", G_TYPE_INT, dec->header->nb_channels,
"signed", G_TYPE_BOOLEAN, TRUE,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
goto nego_failed;
gst_caps_unref (caps);
return GST_FLOW_OK;
/* ERRORS */
no_header:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't read header"));
return GST_FLOW_ERROR;
}
mode_too_old:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL),
("Mode number %d does not (yet/any longer) exist in this version",
dec->header->mode));
return GST_FLOW_ERROR;
}
init_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't initialize decoder"));
return GST_FLOW_ERROR;
}
nego_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't negotiate format"));
gst_caps_unref (caps);
return GST_FLOW_NOT_NEGOTIATED;
}
}
static gboolean
gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
{
/* Using the parsed information upstream, and the list of allowed caps
* downstream, this code tries to find a suitable audio info. It is important
* to keep in mind that the rate and number of channels should never deviate
* from the one the bitstream has, otherwise mpg123 has to mix channels and/or
* resample (and as its docs say, its internal resampler is very crude). The
* sample format, however, can be chosen freely, because the MPEG specs do not
* mandate any special format. Therefore, rate and number of channels are taken
* from upstream (which parsed the MPEG frames, so the input_caps contain
* exactly the rate and number of channels the bitstream actually has), while
* the sample format is chosen by trying out all caps that are allowed by
* downstream. This way, the output is adjusted to what the downstream prefers.
*
* Also, the new output audio info is not set immediately. Instead, it is
* considered the "next audioinfo". The code waits for mpg123 to notice the new
* format (= when mpg123_decode_frame() returns MPG123_AUDIO_DEC_NEW_FORMAT),
* and then sets the next audioinfo. Otherwise, the next audioinfo is set too
* soon, which may cause problems with mp3s containing several format headers.
* One example would be an mp3 with the first 30 seconds using 44.1 kHz, then
* the next 30 seconds using 32 kHz. Rare, but possible.
*
* STEPS:
*
* 1. get rate and channels from input_caps
* 2. get allowed caps from src pad
* 3. for each structure in allowed caps:
* 3.1. take format
* 3.2. if the combination of format with rate and channels is unsupported by
* mpg123, go to (3), or exit with error if there are no more structures
* to try
* 3.3. create next audioinfo out of rate,channels,format, and exit
*/
int rate, channels;
GstMpg123AudioDec *mpg123_decoder;
GstCaps *allowed_srccaps;
guint structure_nr;
gboolean match_found = FALSE;
mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
g_assert (mpg123_decoder->handle != NULL);
mpg123_decoder->has_next_audioinfo = FALSE;
/* Get rate and channels from input_caps */
{
GstStructure *structure;
gboolean err = FALSE;
/* Only the first structure is used (multiple
* input caps structures don't make sense */
structure = gst_caps_get_structure (input_caps, 0);
if (!gst_structure_get_int (structure, "rate", &rate)) {
err = TRUE;
GST_ERROR_OBJECT (dec, "Input caps do not have a rate value");
}
if (!gst_structure_get_int (structure, "channels", &channels)) {
err = TRUE;
GST_ERROR_OBJECT (dec, "Input caps do not have a channel value");
}
if (err)
return FALSE;
}
/* Get the caps that are allowed by downstream */
{
GstCaps *allowed_srccaps_unnorm =
gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
allowed_srccaps = gst_caps_normalize (allowed_srccaps_unnorm);
}
/* Go through all allowed caps, pick the first one that matches */
for (structure_nr = 0; structure_nr < gst_caps_get_size (allowed_srccaps);
++structure_nr) {
GstStructure *structure;
gchar const *format_str;
GstAudioFormat format;
int encoding;
structure = gst_caps_get_structure (allowed_srccaps, structure_nr);
format_str = gst_structure_get_string (structure, "format");
if (format_str == NULL) {
GST_DEBUG_OBJECT (dec, "Could not get format from src caps");
continue;
}
format = gst_audio_format_from_string (format_str);
if (format == GST_AUDIO_FORMAT_UNKNOWN) {
GST_DEBUG_OBJECT (dec, "Unknown format %s", format_str);
continue;
}
switch (format) {
case GST_AUDIO_FORMAT_S16:
encoding = MPG123_ENC_SIGNED_16;
break;
case GST_AUDIO_FORMAT_S24:
encoding = MPG123_ENC_SIGNED_24;
break;
case GST_AUDIO_FORMAT_S32:
encoding = MPG123_ENC_SIGNED_32;
break;
case GST_AUDIO_FORMAT_U16:
encoding = MPG123_ENC_UNSIGNED_16;
break;
case GST_AUDIO_FORMAT_U24:
encoding = MPG123_ENC_UNSIGNED_24;
break;
case GST_AUDIO_FORMAT_U32:
encoding = MPG123_ENC_UNSIGNED_32;
break;
case GST_AUDIO_FORMAT_F32:
encoding = MPG123_ENC_FLOAT_32;
break;
default:
GST_DEBUG_OBJECT (dec,
"Format %s in srccaps is not supported", format_str);
continue;
}
{
int err;
/* Cleanup old formats & set new one */
mpg123_format_none (mpg123_decoder->handle);
err = mpg123_format (mpg123_decoder->handle, rate, channels, encoding);
if (err != MPG123_OK) {
GST_DEBUG_OBJECT (dec,
"mpg123 cannot use caps %" GST_PTR_FORMAT
" because mpg123_format() failed: %s", structure,
mpg123_strerror (mpg123_decoder->handle));
continue;
}
}
gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format, rate,
channels, NULL);
GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
format_str, rate, channels);
mpg123_decoder->has_next_audioinfo = TRUE;
match_found = TRUE;
break;
}
gst_caps_unref (allowed_srccaps);
return match_found;
}
static GstFlowReturn
gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
GstA52Dec *a52dec;
gint channels, i;
gboolean need_reneg = FALSE;
gint chans;
gint length = 0, flags, sample_rate, bit_rate;
GstMapInfo map;
GstFlowReturn result = GST_FLOW_OK;
GstBuffer *outbuf;
const gint num_blocks = 6;
a52dec = GST_A52DEC (bdec);
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
/* parsed stuff already, so this should work out fine */
gst_buffer_map (buffer, &map, GST_MAP_READ);
g_assert (map.size >= 7);
/* re-obtain some sync header info,
* should be same as during _parse and could also be cached there,
* but anyway ... */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
length = a52_syncinfo (map.data, &flags, &sample_rate, &bit_rate);
g_assert (length == map.size);
/* update stream information, renegotiate or re-streaminfo if needed */
need_reneg = FALSE;
if (a52dec->sample_rate != sample_rate) {
GST_DEBUG_OBJECT (a52dec, "sample rate changed");
need_reneg = TRUE;
a52dec->sample_rate = sample_rate;
}
if (flags) {
if (a52dec->stream_channels != (flags & (A52_CHANNEL_MASK | A52_LFE))) {
GST_DEBUG_OBJECT (a52dec, "stream channel flags changed, marking update");
a52dec->flag_update = TRUE;
}
a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
}
if (bit_rate != a52dec->bit_rate) {
a52dec->bit_rate = bit_rate;
gst_a52dec_update_streaminfo (a52dec);
}
/* If we haven't had an explicit number of channels chosen through properties
* at this point, choose what to downmix to now, based on what the peer will
* accept - this allows a52dec to do downmixing in preference to a
* downstream element such as audioconvert.
*/
if (a52dec->request_channels != A52_CHANNEL) {
flags = a52dec->request_channels;
} else if (a52dec->flag_update) {
GstCaps *caps;
a52dec->flag_update = FALSE;
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
if (caps && gst_caps_get_size (caps) > 0) {
GstCaps *copy = gst_caps_copy_nth (caps, 0);
GstStructure *structure = gst_caps_get_structure (copy, 0);
gint orig_channels = flags ? gst_a52dec_channels (flags, NULL) : 6;
gint fixed_channels = 0;
const int a52_channels[6] = {
A52_MONO,
A52_STEREO,
A52_STEREO | A52_LFE,
A52_2F2R,
A52_2F2R | A52_LFE,
A52_3F2R | A52_LFE,
};
/* Prefer the original number of channels, but fixate to something
* preferred (first in the caps) downstream if possible.
*/
gst_structure_fixate_field_nearest_int (structure, "channels",
orig_channels);
if (gst_structure_get_int (structure, "channels", &fixed_channels)
&& fixed_channels <= 6) {
if (fixed_channels < orig_channels)
flags = a52_channels[fixed_channels - 1];
} else {
flags = a52_channels[5];
}
gst_caps_unref (copy);
} else if (flags)
flags = a52dec->stream_channels;
else
flags = A52_3F2R | A52_LFE;
if (caps)
gst_caps_unref (caps);
} else {
flags = a52dec->using_channels;
}
/* process */
flags |= A52_ADJUST_LEVEL;
a52dec->level = 1;
if (a52_frame (a52dec->state, map.data, &flags, &a52dec->level, a52dec->bias)) {
gst_buffer_unmap (buffer, &map);
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
("a52_frame error"), result);
goto exit;
}
gst_buffer_unmap (buffer, &map);
channels = flags & (A52_CHANNEL_MASK | A52_LFE);
if (a52dec->using_channels != channels) {
need_reneg = TRUE;
a52dec->using_channels = channels;
}
/* negotiate if required */
if (need_reneg) {
GST_DEBUG_OBJECT (a52dec,
"a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
if (!gst_a52dec_reneg (a52dec))
goto failed_negotiation;
}
if (a52dec->dynamic_range_compression == FALSE) {
a52_dynrng (a52dec->state, NULL, NULL);
}
flags &= (A52_CHANNEL_MASK | A52_LFE);
chans = gst_a52dec_channels (flags, NULL);
if (!chans)
goto invalid_flags;
/* handle decoded data;
* each frame has 6 blocks, one block is 256 samples, ea */
outbuf =
gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
{
guint8 *ptr = map.data;
for (i = 0; i < num_blocks; i++) {
if (a52_block (a52dec->state)) {
/* also marks discont */
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
("error decoding block %d", i), result);
if (result != GST_FLOW_OK) {
gst_buffer_unmap (outbuf, &map);
goto exit;
}
} else {
gint n, c;
gint *reorder_map = a52dec->channel_reorder_map;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) ptr)[n * chans + reorder_map[c]] =
a52dec->samples[c * 256 + n];
}
}
}
ptr += 256 * chans * (SAMPLE_WIDTH / 8);
}
}
gst_buffer_unmap (outbuf, &map);
result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
exit:
return result;
/* ERRORS */
failed_negotiation:
{
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
invalid_flags:
{
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
("Invalid channel flags: %d", flags));
return GST_FLOW_ERROR;
}
}
