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dtsdec: port to audiodecoder

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This commit is contained in:
Mark Nauwelaerts 2011-11-23 23:29:10 +01:00
parent da43e59aab
commit 8b5fbcaedd
3 changed files with 236 additions and 407 deletions
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5
ext/dts/Makefile.am
View file

@ -1,8 +1,9 @@
plugin_LTLIBRARIES = libgstdtsdec.la
libgstdtsdec_la_SOURCES = gstdtsdec.c
libgstdtsdec_la_CFLAGS = $(GST_CFLAGS) $(ORC_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
libgstdtsdec_la_LIBADD = $(DTS_LIBS) $(ORC_LIBS) $(GST_PLUGINS_BASE_LIBS) \
libgstdtsdec_la_CFLAGS = -DGST_USE_UNSTABLE_API \
$(GST_CFLAGS) $(ORC_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
libgstdtsdec_la_LIBADD = $(DTS_LIBS) $(ORC_LIBS) $(GST_PLUGINS_BASE_LIBS) \
-lgstaudio-@GST_MAJORMINOR@
libgstdtsdec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstdtsdec_la_LIBTOOLFLAGS = --tag=disable-static

618
ext/dts/gstdtsdec.c
View file

@ -127,14 +127,20 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstElement, GST_TYPE_ELEMENT);
GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER);
static gboolean gst_dtsdec_start (GstAudioDecoder * dec);
static gboolean gst_dtsdec_stop (GstAudioDecoder * dec);
static gboolean gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
static gboolean gst_dtsdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length);
static GstFlowReturn gst_dtsdec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static GstFlowReturn gst_dtsdec_pre_push (GstAudioDecoder * bdec,
GstBuffer ** buffer);
static gboolean gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf);
static GstFlowReturn gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf);
static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element,
GstStateChange transition);
static void gst_dtsdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
@ -164,16 +170,21 @@ static void
gst_dtsdec_class_init (GstDtsDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioDecoderClass *gstbase_class;
guint cpuflags;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbase_class = (GstAudioDecoderClass *) klass;
gobject_class->set_property = gst_dtsdec_set_property;
gobject_class->get_property = gst_dtsdec_get_property;
gstelement_class->change_state = gst_dtsdec_change_state;
gstbase_class->start = GST_DEBUG_FUNCPTR (gst_dtsdec_start);
gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_dtsdec_stop);
gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_dtsdec_set_format);
gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_dtsdec_parse);
gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_dtsdec_handle_frame);
gstbase_class->pre_push = GST_DEBUG_FUNCPTR (gst_dtsdec_pre_push);
/**
* GstDtsDec::drc
@ -209,23 +220,104 @@ gst_dtsdec_class_init (GstDtsDecClass * klass)
static void
gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
{
/* create the sink and src pads */
dtsdec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_sink_setcaps));
gst_pad_set_chain_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_chain));
gst_pad_set_event_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_sink_event));
gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad);
dtsdec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->srcpad);
dtsdec->request_channels = DCA_CHANNEL;
dtsdec->dynamic_range_compression = FALSE;
gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
/* retrieve and intercept base class chain.
* Quite HACKish, but that's dvd specs for you,
* since one buffer needs to be split into 2 frames */
dtsdec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (dtsdec));
gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (dtsdec),
GST_DEBUG_FUNCPTR (gst_dtsdec_chain));
}
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_byte_time (dec, TRUE);
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;
}
if (dts->pending_tags) {
gst_tag_list_free (dts->pending_tags);
dts->pending_tags = NULL;
}
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_UNEXPECTED;
dts = GST_DTSDEC (bdec);
size = av = gst_adapter_available (adapter);
data = (guint8 *) gst_adapter_peek (adapter, av);
/* find and read header */
bit_rate = dts->bit_rate;
sample_rate = dts->sample_rate;
flags = 0;
while (av >= 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;
}
}
*_offset = av - size;
*len = length;
return result;
}
static gint
@ -327,105 +419,6 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
return chans;
}
static void
clear_queued (GstDtsDec * dec)
{
g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->queued);
dec->queued = NULL;
}
static GstFlowReturn
flush_queued (GstDtsDec * dec)
{
GstFlowReturn ret = GST_FLOW_OK;
while (dec->queued) {
GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* iterate ouput queue an push downstream */
ret = gst_pad_push (dec->srcpad, buf);
dec->queued = g_list_delete_link (dec->queued, dec->queued);
}
return ret;
}
static GstFlowReturn
gst_dtsdec_drain (GstDtsDec * dec)
{
GstFlowReturn ret = GST_FLOW_OK;
if (dec->segment.rate < 0.0) {
/* if we have some queued frames for reverse playback, flush
* them now */
ret = flush_queued (dec);
}
return ret;
}
static GstFlowReturn
gst_dtsdec_push (GstDtsDec * dtsdec,
GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
{
GstBuffer *buf;
int chans, n, c;
GstFlowReturn result;
flags &= (DCA_CHANNEL_MASK | DCA_LFE);
chans = gst_dtsdec_channels (flags, NULL);
if (!chans) {
GST_ELEMENT_ERROR (GST_ELEMENT (dtsdec), STREAM, DECODE, (NULL),
("Invalid channel flags: %d", flags));
return GST_FLOW_ERROR;
}
result =
gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
if (result != GST_FLOW_OK)
return result;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
samples[c * 256 + n];
}
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / dtsdec->sample_rate;
result = GST_FLOW_OK;
if ((buf = gst_audio_buffer_clip (buf, &dtsdec->segment,
dtsdec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
/* set discont when needed */
if (dtsdec->discont) {
GST_LOG_OBJECT (dtsdec, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
dtsdec->discont = FALSE;
}
if (dtsdec->segment.rate > 0.0) {
GST_DEBUG_OBJECT (dtsdec,
"Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
result = gst_pad_push (srcpad, buf);
} else {
/* reverse playback, queue frame till later when we get a discont. */
GST_DEBUG_OBJECT (dtsdec, "queued frame");
dtsdec->queued = g_list_prepend (dtsdec->queued, buf);
}
}
return result;
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
@ -446,7 +439,7 @@ gst_dtsdec_renegotiate (GstDtsDec * dts)
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
if (!gst_pad_set_caps (dts->srcpad, caps))
if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dts), caps))
goto done;
result = TRUE;
@ -458,100 +451,70 @@ done:
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;
gboolean update;
gint64 start, end, pos;
gdouble rate;
gst_event_parse_new_segment (event, &update, &rate, &format, &start, &end,
&pos);
/* drain queued buffers before activating the segment so that we can clip
* against the old segment first */
gst_dtsdec_drain (dtsdec);
if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) {
GST_WARNING ("No time in newsegment event %p (format is %s)",
event, gst_format_get_name (format));
gst_event_unref (event);
dtsdec->sent_segment = FALSE;
/* set some dummy values, FIXME: do proper conversion */
dtsdec->time = start = pos = 0;
format = GST_FORMAT_TIME;
end = -1;
} else {
dtsdec->time = start;
dtsdec->sent_segment = TRUE;
ret = gst_pad_push_event (dtsdec->srcpad, event);
}
gst_segment_set_newsegment (&dtsdec->segment, update, rate, format, start,
end, pos);
break;
}
case GST_EVENT_TAG:
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
case GST_EVENT_EOS:
gst_dtsdec_drain (dtsdec);
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
case GST_EVENT_FLUSH_START:
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
if (dtsdec->cache) {
gst_buffer_unref (dtsdec->cache);
dtsdec->cache = NULL;
}
clear_queued (dtsdec);
gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
default:
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
}
gst_object_unref (dtsdec);
return ret;
}
static void
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
{
GstTagList *taglist;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_AUDIO_CODEC, "DTS DCA", NULL);
if (dts->bit_rate > 3) {
taglist = gst_tag_list_new ();
/* 1 => open bitrate, 2 => variable bitrate, 3 => lossless */
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
(guint) dts->bit_rate, NULL);
}
gst_element_found_tags_for_pad (GST_ELEMENT (dts), dts->srcpad, taglist);
if (dts->pending_tags) {
gst_tag_list_free (dts->pending_tags);
dts->pending_tags = NULL;
}
dts->pending_tags = taglist;
}
}
static GstFlowReturn
gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
gst_dtsdec_pre_push (GstAudioDecoder * bdec, GstBuffer ** buffer)
{
GstDtsDec *dts = GST_DTSDEC (bdec);
if (G_UNLIKELY (dts->pending_tags)) {
gst_element_found_tags_for_pad (GST_ELEMENT (dts),
GST_AUDIO_DECODER_SRC_PAD (dts), dts->pending_tags);
dts->pending_tags = NULL;
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
GstDtsDec *dts;
gint channels, i, num_blocks;
gboolean need_renegotiation = FALSE;
guint8 *data;
gint size, chans;
gint length = 0, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_UNEXPECTED;
GstBuffer *outbuf;
dts = GST_DTSDEC (bdec);
/* parsed stuff already, so this should work out fine */
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
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) {
@ -581,7 +544,7 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
dts->flag_update = FALSE;
caps = gst_pad_get_allowed_caps (dts->srcpad);
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);
@ -618,14 +581,16 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
} 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_WARNING_OBJECT (dts, "dts_frame error");
dts->discont = TRUE;
return GST_FLOW_OK;
GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL),
("dts_frame error"), result);
goto exit;
}
channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
if (dts->using_channels != channels) {
need_renegotiation = TRUE;
@ -636,42 +601,71 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
if (need_renegotiation) {
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 (!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);
result =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dts), 0,
256 * chans * (SAMPLE_WIDTH / 8) * num_blocks,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dts)), &outbuf);
if (result != GST_FLOW_OK)
goto exit;
data = GST_BUFFER_DATA (outbuf);
for (i = 0; i < num_blocks; i++) {
if (dca_block (dts->state)) {
/* Ignore errors, but mark a discont */
GST_WARNING_OBJECT (dts, "dts_block error %d", i);
dts->discont = TRUE;
/* 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 {
GstFlowReturn ret;
gint n, c;
/* push on */
ret = gst_dtsdec_push (dts, dts->srcpad, dts->using_channels,
dts->samples, dts->time);
if (ret != GST_FLOW_OK)
return ret;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) data)[n * chans + c] = dts->samples[c * 256 + n];
}
}
}
dts->time += GST_SECOND * 256 / dts->sample_rate;
data += 256 * chans * (SAMPLE_WIDTH / 8);
}
return GST_FLOW_OK;
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 gboolean
gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
GstDtsDec *dts = GST_DTSDEC (bdec);
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
@ -681,8 +675,6 @@ gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
else
dts->dvdmode = FALSE;
gst_object_unref (dts);
return TRUE;
}
@ -693,17 +685,6 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
gint first_access;
if (GST_BUFFER_IS_DISCONT (buf)) {
GST_LOG_OBJECT (dts, "received DISCONT");
gst_dtsdec_drain (dts);
/* clear cache on discont and mark a discont in the element */
if (dts->cache) {
gst_buffer_unref (dts->cache);
dts->cache = NULL;
}
dts->discont = TRUE;
}
if (dts->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guint8 *data = GST_BUFFER_DATA (buf);
@ -726,33 +707,38 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
goto bad_first_access_parameter;
subbuf = gst_buffer_create_sub (buf, offset, len);
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
ret = gst_dtsdec_chain_raw (pad, subbuf);
if (ret != GST_FLOW_OK)
ret = dts->base_chain (pad, subbuf);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buf);
goto done;
}
offset += len;
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_create_sub (buf, offset, len);
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = gst_dtsdec_chain_raw (pad, subbuf);
ret = dts->base_chain (pad, 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_create_sub (buf, offset, size - offset);
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = gst_dtsdec_chain_raw (pad, subbuf);
ret = dts->base_chain (pad, subbuf);
gst_buffer_unref (buf);
}
} else {
gst_buffer_ref (buf);
ret = gst_dtsdec_chain_raw (pad, buf);
ret = dts->base_chain (pad, buf);
}
done:
gst_buffer_unref (buf);
return ret;
/* ERRORS */
@ -772,154 +758,6 @@ bad_first_access_parameter:
}
}
static GstFlowReturn
gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
{
GstDtsDec *dts;
guint8 *data;
gint size;
gint length = 0, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_OK;
dts = GST_DTSDEC (GST_PAD_PARENT (pad));
if (!dts->sent_segment) {
GstSegment segment;
/* Create a basic segment. Usually, we'll get a new-segment sent by
* another element that will know more information (a demuxer). If we're
* just looking at a raw AC3 stream, we won't - so we need to send one
* here, but we don't know much info, so just send a minimal TIME
* new-segment event
*/
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_push_event (dts->srcpad, gst_event_new_new_segment (FALSE,
segment.rate, segment.format, segment.start,
segment.duration, segment.start));
dts->sent_segment = TRUE;
}
/* merge with cache, if any. Also make sure timestamps match */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
dts->time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (dts,
"Received buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
if (dts->cache) {
buf = gst_buffer_join (dts->cache, buf);
dts->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* 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_DEBUG ("Sync: frame size %d", length);
if (flags != dts->prev_flags)
dts->flag_update = TRUE;
dts->prev_flags = flags;
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 = dca_init (klass->dts_cpuflags);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
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->time = 0;
dts->sent_segment = FALSE;
dts->flag_update = TRUE;
gst_segment_init (&dts->segment, GST_FORMAT_UNDEFINED);
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;
}
clear_queued (dts);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
dca_free (dts->state);
dts->state = NULL;
break;
default:
break;
}
return ret;
}
static void
gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)

20
ext/dts/gstdtsdec.h
View file

@ -22,6 +22,7 @@
#define __GST_DTSDEC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiodecoder.h>
G_BEGIN_DECLS
@ -40,16 +41,11 @@ typedef struct _GstDtsDec GstDtsDec;
typedef struct _GstDtsDecClass GstDtsDecClass;
struct _GstDtsDec {
GstElement element;
GstAudioDecoder element;
/* pads */
GstPad *sinkpad;
GstPad *srcpad;
GstSegment segment;
GstPadChainFunction base_chain;
gboolean dvdmode;
gboolean sent_segment;
gboolean discont;
gboolean flag_update;
gboolean prev_flags;
@ -71,17 +67,11 @@ struct _GstDtsDec {
dts_state_t *state;
#endif
/* Data left over from the previous buffer */
GstBuffer *cache;
GstClockTime time;
/* reverse playback */
GList *queued;
GstTagList *pending_tags;
};
struct _GstDtsDecClass {
GstElementClass parent_class;
GstAudioDecoderClass parent_class;
guint32 dts_cpuflags;
};