static GstFlowReturn
gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
{
GstDtsDec *dts;
guint8 *data;
gint size;
gint length, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_OK;
dts = GST_DTSDEC (GST_PAD_PARENT (pad));
if (dts->cache) {
buf = gst_buffer_join (dts->cache, buf);
dts->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
length = 0;
while (size >= 7) {
length = dts_syncinfo (dts->state, data, &flags,
&sample_rate, &bit_rate, &frame_length);
if (length == 0) {
/* shift window to re-find sync */
data++;
size--;
} else if (length <= size) {
GST_DEBUG ("Sync: frame size %d", length);
result = gst_dtsdec_handle_frame (dts, data, length,
flags, sample_rate, bit_rate);
if (result != GST_FLOW_OK) {
size = 0;
break;
}
size -= length;
data += length;
} else {
GST_LOG ("Not enough data available (needed %d had %d)", length, size);
break;
}
}
/* keep cache */
if (length == 0) {
GST_LOG ("No sync found");
}
if (size > 0) {
dts->cache = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - size, size);
}
gst_buffer_unref (buf);
return result;
}
static GstStateChangeReturn
gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstDtsDec *dts = GST_DTSDEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:{
GstDtsDecClass *klass;
klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
dts->state = dts_init (klass->dts_cpuflags);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
dts->samples = dts_samples (dts->state);
dts->bit_rate = -1;
dts->sample_rate = -1;
dts->stream_channels = 0;
/* FIXME force stereo for now */
dts->request_channels = DTS_CHANNEL;
dts->using_channels = 0;
dts->level = 1;
dts->bias = 0;
dts->current_ts = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
dts->samples = NULL;
if (dts->cache) {
gst_buffer_unref (dts->cache);
dts->cache = NULL;
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
dts_free (dts->state);
dts->state = NULL;
break;
default:
break;
}
return ret;
}
static void
gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case PROP_DRC:
g_value_set_boolean (value, dts->dynamic_range_compression);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstDtsDec *dts = GST_DTSDEC (bdec);
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
dts->dvdmode = TRUE;
else
dts->dvdmode = FALSE;
return TRUE;
}
static gboolean
gst_dtsdec_stop (GstAudioDecoder * dec)
{
GstDtsDec *dts = GST_DTSDEC (dec);
GST_DEBUG_OBJECT (dec, "stop");
dts->samples = NULL;
if (dts->state) {
dca_free (dts->state);
dts->state = NULL;
}
return TRUE;
}
static gboolean
gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
dts->dvdmode = TRUE;
else
dts->dvdmode = FALSE;
gst_object_unref (dts);
return TRUE;
}
static GstFlowReturn
gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
gint * _offset, gint * len)
{
GstDtsDec *dts;
guint8 *data;
gint av, size;
gint length = 0, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_EOS;
dts = GST_DTSDEC (bdec);
size = av = gst_adapter_available (adapter);
data = (guint8 *) gst_adapter_map (adapter, av);
/* find and read header */
bit_rate = dts->bit_rate;
sample_rate = dts->sample_rate;
flags = 0;
while (size >= 7) {
length = dca_syncinfo (dts->state, data, &flags,
&sample_rate, &bit_rate, &frame_length);
if (length == 0) {
/* shift window to re-find sync */
data++;
size--;
} else if (length <= size) {
GST_LOG_OBJECT (dts, "Sync: frame size %d", length);
result = GST_FLOW_OK;
break;
} else {
GST_LOG_OBJECT (dts, "Not enough data available (needed %d had %d)",
length, size);
break;
}
}
gst_adapter_unmap (adapter);
*_offset = av - size;
*len = length;
return result;
}
static gboolean
gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
{
GstDtsDec *dtsdec = GST_DTSDEC (gst_pad_get_parent (pad));
gboolean ret = FALSE;
GST_LOG_OBJECT (dtsdec, "%s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
GstFormat format;
gint64 val;
gst_event_parse_new_segment (event, NULL, NULL, &format, &val, NULL,
NULL);
if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (val)) {
GST_WARNING ("No time in newsegment event %p", event);
} else {
dtsdec->current_ts = val;
}
if (dtsdec->cache) {
gst_buffer_unref (dtsdec->cache);
dtsdec->cache = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
}
case GST_EVENT_FLUSH_STOP:
if (dtsdec->cache) {
gst_buffer_unref (dtsdec->cache);
dtsdec->cache = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (dtsdec);
return ret;
}
static gboolean
gst_dtsdec_start (GstAudioDecoder * dec)
{
GstDtsDec *dts = GST_DTSDEC (dec);
GstDtsDecClass *klass;
GST_DEBUG_OBJECT (dec, "start");
klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
dts->state = dca_init (klass->dts_cpuflags);
dts->samples = dca_samples (dts->state);
dts->bit_rate = -1;
dts->sample_rate = -1;
dts->stream_channels = DCA_CHANNEL;
dts->using_channels = DCA_CHANNEL;
dts->level = 1;
dts->bias = 0;
dts->flag_update = TRUE;
/* call upon legacy upstream byte support (e.g. seeking) */
gst_audio_decoder_set_estimate_rate (dec, TRUE);
return TRUE;
}
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstDtsDec *dts = GST_DTSDEC (parent);
gint first_access;
if (dts->dvdmode) {
guint8 data[2];
gsize size;
gint offset, len;
GstBuffer *subbuf;
size = gst_buffer_get_size (buf);
if (size < 2)
goto not_enough_data;
gst_buffer_extract (buf, 0, data, 2);
first_access = (data[0] << 8) | data[1];
/* Skip the first_access header */
offset = 2;
if (first_access > 1) {
/* Length of data before first_access */
len = first_access - 1;
if (len <= 0 || offset + len > size)
goto bad_first_access_parameter;
subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
ret = dts->base_chain (pad, parent, subbuf);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buf);
goto done;
}
offset += len;
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = dts->base_chain (pad, parent, subbuf);
}
gst_buffer_unref (buf);
} else {
/* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
subbuf =
gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset,
size - offset);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = dts->base_chain (pad, parent, subbuf);
gst_buffer_unref (buf);
}
} else {
ret = dts->base_chain (pad, parent, buf);
}
done:
return ret;
/* ERRORS */
not_enough_data:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Insufficient data in buffer. Can't determine first_acess"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
bad_first_access_parameter:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Bad first_access parameter (%d) in buffer", first_access));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
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_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
gint first_access;
if (dts->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guint8 *data = GST_BUFFER_DATA (buf);
gint offset, len;
GstBuffer *subbuf;
if (size < 2)
goto not_enough_data;
first_access = (data[0] << 8) | data[1];
/* Skip the first_access header */
offset = 2;
if (first_access > 1) {
/* Length of data before first_access */
len = first_access - 1;
if (len <= 0 || offset + len > size)
goto bad_first_access_parameter;
subbuf = gst_buffer_create_sub (buf, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
res = gst_dtsdec_chain_raw (pad, subbuf);
if (res != GST_FLOW_OK)
goto done;
offset += len;
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_create_sub (buf, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
res = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
/* first_access = 0 or 1, so if there's a timestamp it applies
* to the first byte */
subbuf = gst_buffer_create_sub (buf, offset, size - offset);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
res = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
res = gst_dtsdec_chain_raw (pad, buf);
}
done:
return res;
/* ERRORS */
not_enough_data:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Insufficient data in buffer. Can't determine first_acess"));
return GST_FLOW_ERROR;
}
bad_first_access_parameter:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Bad first_access parameter (%d) in buffer", first_access));
return GST_FLOW_ERROR;
}
}