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_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
{
gboolean need_renegotiation = FALSE;
gint channels, num_blocks;
GstBuffer *out;
gint i, s, c, num_c;
sample_t *samples;
GstFlowReturn result = GST_FLOW_OK;
/* go over stream properties, update caps/streaminfo if needed */
if (dts->sample_rate != sample_rate) {
need_renegotiation = TRUE;
dts->sample_rate = sample_rate;
}
dts->stream_channels = flags;
if (bit_rate != dts->bit_rate) {
dts->bit_rate = bit_rate;
gst_dtsdec_update_streaminfo (dts);
}
if (dts->request_channels == DTS_CHANNEL) {
GstCaps *caps;
caps = gst_pad_get_allowed_caps (dts->srcpad);
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] = {
DTS_MONO,
DTS_STEREO,
DTS_STEREO | DTS_LFE,
DTS_2F2R,
DTS_2F2R | DTS_LFE,
DTS_3F2R | DTS_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)
dts->request_channels = dts_channels[channels - 1];
else
dts->request_channels = dts_channels[5];
gst_caps_unref (copy);
} else if (flags) {
dts->request_channels = dts->stream_channels;
} else {
dts->request_channels = DTS_3F2R | DTS_LFE;
}
if (caps)
gst_caps_unref (caps);
}
/* process */
flags = dts->request_channels | DTS_ADJUST_LEVEL;
dts->level = 1;
if (dts_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
GST_WARNING ("dts_frame error");
return GST_FLOW_OK;
}
channels = flags & (DTS_CHANNEL_MASK | DTS_LFE);
if (dts->using_channels != channels) {
need_renegotiation = TRUE;
dts->using_channels = channels;
}
if (need_renegotiation == TRUE) {
GST_DEBUG ("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)) {
GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
}
if (dts->dynamic_range_compression == FALSE) {
dts_dynrng (dts->state, NULL, NULL);
}
/* handle decoded data, one block is 256 samples */
num_blocks = dts_blocks_num (dts->state);
for (i = 0; i < num_blocks; i++) {
if (dts_block (dts->state)) {
GST_WARNING ("dts_block error %d", i);
continue;
}
samples = dts_samples (dts->state);
num_c = gst_dtsdec_channels (dts->using_channels, NULL);
result = gst_pad_alloc_buffer_and_set_caps (dts->srcpad, 0,
(SAMPLE_WIDTH / 8) * 256 * num_c, GST_PAD_CAPS (dts->srcpad), &out);
if (result != GST_FLOW_OK)
break;
GST_BUFFER_TIMESTAMP (out) = dts->current_ts;
GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
dts->current_ts += GST_BUFFER_DURATION (out);
/* libdts returns buffers in 256-sample-blocks per channel,
* we want interleaved. And we need to copy anyway... */
data = GST_BUFFER_DATA (out);
for (s = 0; s < 256; s++) {
for (c = 0; c < num_c; c++) {
*(sample_t *) data = samples[s + c * 256];
data += (SAMPLE_WIDTH / 8);
}
}
/* push on */
result = gst_pad_push (dts->srcpad, out);
if (result != GST_FLOW_OK)
break;
}
return result;
}
