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-rw-r--r--ext/dts/gstdtsdec.c283
-rw-r--r--ext/dts/gstdtsdec.h2
2 files changed, 144 insertions, 141 deletions
diff --git a/ext/dts/gstdtsdec.c b/ext/dts/gstdtsdec.c
index f71219478..74ef14c01 100644
--- a/ext/dts/gstdtsdec.c
+++ b/ext/dts/gstdtsdec.c
@@ -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,
+ 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);
+
+ 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, (gint) dts->sample_rate, NULL);
- gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
- g_free (pos);
+ "rate", G_TYPE_INT, dts->sample_rate, NULL);
+
+ if (channels > 1) {
+ guint64 channel_mask = 0;
- if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dts), caps))
+ 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;
-
- for (n = 0; n < 256; n++) {
- for (c = 0; c < chans; c++) {
- ((sample_t *) data)[n * chans + c] = dts->samples[c * 256 + n];
+ 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 *) 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
diff --git a/ext/dts/gstdtsdec.h b/ext/dts/gstdtsdec.h
index be6005a0a..16b7e91ea 100644
--- a/ext/dts/gstdtsdec.h
+++ b/ext/dts/gstdtsdec.h
@@ -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;
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