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dtsdec: Port to 0.11

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This commit is contained in:
Sebastian Dröge 2012-01-10 16:59:18 +01:00
parent a01a4ea2d3
commit b7c53b95c1
3 changed files with 145 additions and 142 deletions
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2
configure.ac
View file

@ -305,7 +305,7 @@ GST_PLUGINS_NONPORTED=" adpcmdec adpcmenc aiff asfmux \
sdi segmentclip siren speed subenc stereo tta videofilters \
videomaxrate videomeasure videosignal vmnc \
decklink fbdev linsys shm vcd \
voaacenc apexsink bz2 cdaudio celt cog curl dc1394 dirac directfb dts resindvd \
voaacenc apexsink bz2 cdaudio celt cog curl dc1394 dirac directfb resindvd \
gsettings gsm jp2k ladspa modplug mpeg2enc mplex mimic \
musepack musicbrainz nas neon ofa openal opencv rsvg schro sdl smooth sndfile soundtouch spandsp timidity \
wildmidi xvid apple_media voamrwbenc lv2 teletextdec"

283
ext/dts/gstdtsdec.c
View file

@ -43,7 +43,7 @@
#include <stdlib.h>
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <gst/audio/audio.h>
#ifndef DTS_OLD
#include <dca.h>
@ -87,10 +87,13 @@ typedef struct dts_state_s dca_state_t;
#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
#define SAMPLE_WIDTH 16
#define SAMPLE_FORMAT GST_AUDIO_NE(S16)
#elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
#define SAMPLE_WIDTH 64
#define SAMPLE_FORMAT GST_AUDIO_NE(F64)
#else
#define SAMPLE_WIDTH 32
#define SAMPLE_FORMAT GST_AUDIO_NE(F32)
#endif
GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
@ -98,8 +101,8 @@ GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
enum
{
ARG_0,
ARG_DRC
PROP_0,
PROP_DRC
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
@ -108,27 +111,16 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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) " G_STRINGIFY (SAMPLE_WIDTH) ", " \
"depth = (int) 16"
#else
#define DTS_CAPS "audio/x-raw-float, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
"width = (int) " G_STRINGIFY (SAMPLE_WIDTH)
#endif
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (DTS_CAPS ", "
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " SAMPLE_FORMAT ", "
"layout = (string) interleaved, "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER);
G_DEFINE_TYPE (GstDtsDec, gst_dtsdec, GST_TYPE_AUDIO_DECODER);
static gboolean gst_dtsdec_start (GstAudioDecoder * dec);
static gboolean gst_dtsdec_stop (GstAudioDecoder * dec);
@ -140,45 +132,39 @@ static GstFlowReturn gst_dtsdec_handle_frame (GstAudioDecoder * dec,
static GstFlowReturn gst_dtsdec_pre_push (GstAudioDecoder * bdec,
GstBuffer ** buffer);
static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf);
static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static void gst_dtsdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dtsdec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void
gst_dtsdec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (element_class, "DTS audio decoder",
"Codec/Decoder/Audio",
"Decodes DTS audio streams",
"Jan Schmidt <thaytan@noraisin.net>, "
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder");
}
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;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (gstelement_class, "DTS audio decoder",
"Codec/Decoder/Audio",
"Decodes DTS audio streams",
"Jan Schmidt <thaytan@noraisin.net>, "
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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);
@ -194,7 +180,7 @@ gst_dtsdec_class_init (GstDtsDecClass * klass)
* softer and soft sounds louder, so you can more easily listen
* to the stream without disturbing other people.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DRC,
g_param_spec_boolean ("drc", "Dynamic Range Compression",
"Use Dynamic Range Compression", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@ -218,7 +204,7 @@ gst_dtsdec_class_init (GstDtsDecClass * klass)
}
static void
gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
gst_dtsdec_init (GstDtsDec * dtsdec)
{
dtsdec->request_channels = DCA_CHANNEL;
dtsdec->dynamic_range_compression = FALSE;
@ -284,12 +270,12 @@ gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
guint8 *data;
gint av, size;
gint length = 0, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_UNEXPECTED;
GstFlowReturn result = GST_FLOW_EOS;
dts = GST_DTSDEC (bdec);
size = av = gst_adapter_available (adapter);
data = (guint8 *) gst_adapter_peek (adapter, av);
data = (guint8 *) gst_adapter_map (adapter, av);
/* find and read header */
bit_rate = dts->bit_rate;
@ -313,6 +299,7 @@ gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
break;
}
}
gst_adapter_unmap (adapter);
*_offset = av - size;
*len = length;
@ -321,23 +308,16 @@ gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
}
static gint
gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition * pos)
{
gint chans = 0;
GstAudioChannelPosition *tpos = NULL;
if (pos) {
/* Allocate the maximum, for ease */
tpos = *pos = g_new (GstAudioChannelPosition, 7);
if (!tpos)
return 0;
}
switch (flags & DCA_CHANNEL_MASK) {
case DCA_MONO:
chans = 1;
if (tpos)
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
}
break;
/* case DCA_CHANNEL: */
case DCA_STEREO:
@ -345,64 +325,64 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
case DCA_STEREO_TOTAL:
case DCA_DOLBY:
chans = 2;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
break;
case DCA_3F:
chans = 3;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
break;
case DCA_2F1R:
chans = 3;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
break;
case DCA_3F1R:
chans = 4;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
break;
case DCA_2F2R:
chans = 4;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
case DCA_3F2R:
chans = 5;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
case DCA_4F2R:
chans = 6;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
default:
@ -410,8 +390,8 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
return 0;
}
if (flags & DCA_LFE) {
if (tpos) {
tpos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE;
if (pos) {
pos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE1;
}
chans += 1;
}
@ -422,24 +402,39 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
GstAudioChannelPosition *pos;
GstCaps *caps = gst_caps_from_string (DTS_CAPS);
gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
gint channels;
GstCaps *caps = NULL;
gboolean result = FALSE;
GstAudioChannelPosition from[6], to[6];
channels = gst_dtsdec_channels (dts->using_channels, from);
if (!channels)
goto done;
GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
GST_INFO_OBJECT (dts, "dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
gst_caps_set_simple (caps,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
memcpy (to, from, sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (to, channels);
gst_audio_get_channel_reorder_map (channels, from, to,
dts->channel_reorder_map);
if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dts), caps))
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, SAMPLE_FORMAT,
"layout", G_TYPE_STRING, "interleaved",
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, dts->sample_rate, NULL);
if (channels > 1) {
guint64 channel_mask = 0;
gst_audio_channel_positions_to_mask (to, channels, &channel_mask);
gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
NULL);
}
if (!gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (dts), caps))
goto done;
result = TRUE;
@ -457,7 +452,7 @@ gst_dtsdec_update_streaminfo (GstDtsDec * dts)
GstTagList *taglist;
if (dts->bit_rate > 3) {
taglist = gst_tag_list_new ();
taglist = gst_tag_list_new_empty ();
/* 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);
@ -477,8 +472,8 @@ 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);
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (dts),
gst_event_new_tag (dts->pending_tags));
dts->pending_tags = NULL;
}
@ -492,7 +487,8 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
gint channels, i, num_blocks;
gboolean need_renegotiation = FALSE;
guint8 *data;
gint size, chans;
gsize size;
gint chans;
gint length = 0, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_OK;
GstBuffer *outbuf;
@ -504,8 +500,7 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
return GST_FLOW_OK;
/* parsed stuff already, so this should work out fine */
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
g_assert (size >= 7);
bit_rate = dts->bit_rate;
@ -590,10 +585,12 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
flags |= DCA_ADJUST_LEVEL;
dts->level = 1;
if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
gst_buffer_unmap (buffer, data, size);
GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL),
("dts_frame error"), result);
goto exit;
}
gst_buffer_unmap (buffer, data, size);
channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
if (dts->using_channels != channels) {
@ -621,32 +618,34 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
/* 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;
outbuf =
gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
data = GST_BUFFER_DATA (outbuf);
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;
data = gst_buffer_map (outbuf, &size, NULL, GST_MAP_WRITE);
{
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 *) data)[n * chans + c] = dts->samples[c * 256 + n];
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);
}
data += 256 * chans * (SAMPLE_WIDTH / 8);
}
gst_buffer_unmap (outbuf, data, size);
result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
@ -684,18 +683,19 @@ gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
}
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
GstDtsDec *dts = GST_DTSDEC (parent);
gint first_access;
if (dts->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guint8 *data = GST_BUFFER_DATA (buf);
guint8 data[2];
gsize size;
gint offset, len;
GstBuffer *subbuf;
size = gst_buffer_extract (buf, 0, data, 2);
if (size < 2)
goto not_enough_data;
@ -711,10 +711,9 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
if (len <= 0 || offset + len > size)
goto bad_first_access_parameter;
subbuf = gst_buffer_create_sub (buf, offset, len);
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
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, subbuf);
ret = dts->base_chain (pad, parent, subbuf);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buf);
goto done;
@ -724,23 +723,23 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_create_sub (buf, offset, len);
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
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, subbuf);
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_create_sub (buf, offset, size - offset);
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
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, subbuf);
ret = dts->base_chain (pad, parent, subbuf);
gst_buffer_unref (buf);
}
} else {
ret = dts->base_chain (pad, buf);
ret = dts->base_chain (pad, parent, buf);
}
done:
@ -770,7 +769,7 @@ gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
case PROP_DRC:
dts->dynamic_range_compression = g_value_get_boolean (value);
break;
default:
@ -786,7 +785,7 @@ gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
case PROP_DRC:
g_value_set_boolean (value, dts->dynamic_range_compression);
break;
default:
@ -798,6 +797,8 @@ gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder");
#if HAVE_ORC
orc_init ();
#endif

2
ext/dts/gstdtsdec.h
View file

@ -56,6 +56,8 @@ struct _GstDtsDec {
gint request_channels;
gint using_channels;
gint channel_reorder_map[6];
/* decoding properties */
sample_t level;
sample_t bias;