/* GStreamer DTS decoder plugin based on libdtsdec
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * Copyright (C) 2009 Jan Schmidt <thaytan@noraisin.net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#include <liboil/liboilcpu.h>
#include <liboil/liboilfunction.h>
-GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
-#define GST_CAT_DEFAULT (dtsdec_debug)
-
static const GstElementDetails gst_dtsdec_details =
GST_ELEMENT_DETAILS ("DTS audio decoder",
"Codec/Decoder/Audio",
"Decodes DTS audio streams",
+ "Jan Schmidt <thaytan@noraisin.net>\n"
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
+#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
+#define SAMPLE_WIDTH 16
+#elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
+#define SAMPLE_WIDTH 64
+#else
+#define SAMPLE_WIDTH 32
+#endif
+
+GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
+#define GST_CAT_DEFAULT (dtsdec_debug)
+
enum
{
ARG_0,
ARG_DRC
};
+
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-dts;" "audio/x-private1-dts")
+ GST_STATIC_CAPS ("audio/x-dts; audio/x-private1-dts")
);
#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
#define DTS_CAPS "audio/x-raw-int, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
"signed = (boolean) true, " \
- "width = (int) 16, " \
+ "width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", " \
"depth = (int) 16"
-#define SAMPLE_WIDTH 16
-#elif defined(LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
-#define DTS_CAPS "audio/x-raw-float, " \
- "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
- "width = (int) 64"
-#define SAMPLE_WIDTH 64
#else
#define DTS_CAPS "audio/x-raw-float, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
- "width = (int) 32"
-#define SAMPLE_WIDTH 32
+ "width = (int) " G_STRINGIFY (SAMPLE_WIDTH)
#endif
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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);
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details (element_class, &gst_dtsdec_details);
- GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
+ GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder");
}
static void
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_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad);
dtsdec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_pad_use_fixed_caps (dtsdec->srcpad);
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);
}
static gint
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)
{
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);
+ 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->current_ts = val;
+ dtsdec->time = start;
+ dtsdec->sent_segment = TRUE;
+ ret = gst_pad_push_event (dtsdec->srcpad, event);
}
- if (dtsdec->cache) {
- gst_buffer_unref (dtsdec->cache);
- dtsdec->cache = NULL;
- }
- ret = gst_pad_event_default (pad, 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;
}
- ret = gst_pad_event_default (pad, event);
+ clear_queued (dtsdec);
+ gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
+ ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
default:
- ret = gst_pad_event_default (pad, event);
+ ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
}
return ret;
}
-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 void
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
{
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
+ GST_TAG_AUDIO_CODEC, "DTS DCA",
GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (dts), dts->srcpad, taglist);
gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
{
+ gint channels, i, num_blocks;
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 */
+ /* go over stream properties, renegotiate or update streaminfo if needed */
if (dts->sample_rate != sample_rate) {
need_renegotiation = TRUE;
dts->sample_rate = sample_rate;
}
- dts->stream_channels = flags;
+ 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 (dts->request_channels == DCA_CHANNEL) {
+ /* 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 (dts->srcpad);
if (caps && gst_caps_get_size (caps) > 0) {
GstCaps *copy = gst_caps_copy_nth (caps, 0);
flags ? gst_dtsdec_channels (flags, NULL) : 6);
gst_structure_get_int (structure, "channels", &channels);
if (channels <= 6)
- dts->request_channels = dts_channels[channels - 1];
+ flags = dts_channels[channels - 1];
else
- dts->request_channels = dts_channels[5];
+ flags = dts_channels[5];
gst_caps_unref (copy);
} else if (flags) {
- dts->request_channels = dts->stream_channels;
+ flags = dts->stream_channels;
} else {
- dts->request_channels = DCA_3F2R | DCA_LFE;
+ flags = DCA_3F2R | DCA_LFE;
}
if (caps)
gst_caps_unref (caps);
+ } else {
+ flags = dts->using_channels;
}
-
/* process */
- flags = dts->request_channels | DCA_ADJUST_LEVEL;
+ flags |= DCA_ADJUST_LEVEL;
dts->level = 1;
-
if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
- GST_WARNING ("dts_frame error");
+ GST_WARNING_OBJECT (dts, "dts_frame error");
+ dts->discont = TRUE;
return GST_FLOW_OK;
}
-
channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
-
if (dts->using_channels != channels) {
need_renegotiation = TRUE;
dts->using_channels = channels;
}
- if (need_renegotiation == TRUE) {
+ /* negotiate if required */
+ 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)) {
num_blocks = dca_blocks_num (dts->state);
for (i = 0; i < num_blocks; i++) {
if (dca_block (dts->state)) {
- GST_WARNING ("dts_block error %d", i);
- continue;
- }
-
- samples = dca_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;
+ /* Ignore errors, but mark a discont */
+ GST_WARNING_OBJECT (dts, "dts_block error %d", i);
+ dts->discont = TRUE;
+ } else {
+ GstFlowReturn ret;
- 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 */
+ ret = gst_dtsdec_push (dts, dts->srcpad, dts->using_channels,
+ dts->samples, dts->time);
+ if (ret != GST_FLOW_OK)
+ return ret;
}
-
- /* push on */
- result = gst_pad_push (dts->srcpad, out);
-
- if (result != GST_FLOW_OK)
- break;
+ dts->time += GST_SECOND * 256 / dts->sample_rate;
}
- return result;
+ return GST_FLOW_OK;
}
-static GstFlowReturn
-gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
+static gboolean
+gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
- 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;
- }
+ GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
+ GstStructure *structure;
- data = GST_BUFFER_DATA (buf);
- size = GST_BUFFER_SIZE (buf);
- length = 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);
- 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;
- }
- }
+ structure = gst_caps_get_structure (caps, 0);
- /* 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);
- }
+ if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
+ dts->dvdmode = TRUE;
+ else
+ dts->dvdmode = FALSE;
- gst_buffer_unref (buf);
+ gst_object_unref (dts);
- return result;
+ return TRUE;
}
-
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
{
- GstFlowReturn res = GST_FLOW_OK;
+ GstFlowReturn ret = GST_FLOW_OK;
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);
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)
+ ret = gst_dtsdec_chain_raw (pad, subbuf);
+ if (ret != GST_FLOW_OK)
goto done;
offset += len;
subbuf = gst_buffer_create_sub (buf, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
- res = gst_dtsdec_chain_raw (pad, subbuf);
+ ret = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
- /* first_access = 0 or 1, so if there's a timestamp it applies
- * to the first byte */
+ /* 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);
+ ret = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
- res = gst_dtsdec_chain_raw (pad, buf);
+ ret = gst_dtsdec_chain_raw (pad, buf);
}
done:
- return res;
+ return ret;
/* ERRORS */
not_enough_data:
("Bad first_access parameter (%d) in buffer", first_access));
return GST_FLOW_ERROR;
}
+}
+
+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
dts->samples = dca_samples (dts->state);
dts->bit_rate = -1;
dts->sample_rate = -1;
- dts->stream_channels = 0;
- /* FIXME force stereo for now */
- dts->request_channels = DCA_CHANNEL;
- dts->using_channels = 0;
+ dts->stream_channels = DCA_CHANNEL;
+ dts->using_channels = DCA_CHANNEL;
dts->level = 1;
dts->bias = 0;
- dts->current_ts = 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;
gst_buffer_unref (dts->cache);
dts->cache = NULL;
}
+ clear_queued (dts);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
dca_free (dts->state);
projects / platform / upstream / gst-plugins-bad.git / commitdiff