gstreamer/ext/dts/gstdtsdec.c

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/* 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
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-dtsdec
*
* Digital Theatre System (DTS) audio decoder
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! dtsdec ! audioresample ! audioconvert ! alsasink
* ]| Play a DTS audio track from a dvd.
* |[
* gst-launch-1.0 filesrc location=abc.dts ! dtsdec ! audioresample ! audioconvert ! alsasink
* ]| Decode a standalone file and play it.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "_stdint.h"
#include <stdlib.h>
#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#ifndef DTS_OLD
#include <dca.h>
#else
#include <dts.h>
typedef struct dts_state_s dca_state_t;
#define DCA_MONO DTS_MONO
#define DCA_CHANNEL DTS_CHANNEL
#define DCA_STEREO DTS_STEREO
#define DCA_STEREO_SUMDIFF DTS_STEREO_SUMDIFF
#define DCA_STEREO_TOTAL DTS_STEREO_TOTAL
#define DCA_3F DTS_3F
#define DCA_2F1R DTS_2F1R
#define DCA_3F1R DTS_3F1R
#define DCA_2F2R DTS_2F2R
#define DCA_3F2R DTS_3F2R
#define DCA_4F2R DTS_4F2R
#define DCA_DOLBY DTS_DOLBY
#define DCA_CHANNEL_MAX DTS_CHANNEL_MAX
#define DCA_CHANNEL_BITS DTS_CHANNEL_BITS
#define DCA_CHANNEL_MASK DTS_CHANNEL_MASK
#define DCA_LFE DTS_LFE
#define DCA_ADJUST_LEVEL DTS_ADJUST_LEVEL
#define dca_init dts_init
#define dca_syncinfo dts_syncinfo
#define dca_frame dts_frame
#define dca_dynrng dts_dynrng
#define dca_blocks_num dts_blocks_num
#define dca_block dts_block
#define dca_samples dts_samples
#define dca_free dts_free
#endif
#include "gstdtsdec.h"
#if HAVE_ORC
#include <orc/orc.h>
#endif
#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
#define SAMPLE_WIDTH 16
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#define SAMPLE_FORMAT GST_AUDIO_NE(S16)
#define SAMPLE_TYPE GST_AUDIO_FORMAT_S16
#elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
#define SAMPLE_WIDTH 64
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#define SAMPLE_FORMAT GST_AUDIO_NE(F64)
#define SAMPLE_TYPE GST_AUDIO_FORMAT_F64
#else
#define SAMPLE_WIDTH 32
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#define SAMPLE_FORMAT GST_AUDIO_NE(F32)
#define SAMPLE_TYPE GST_AUDIO_FORMAT_F32
#endif
GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
#define GST_CAT_DEFAULT (dtsdec_debug)
enum
{
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PROP_0,
PROP_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")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " SAMPLE_FORMAT ", "
"layout = (string) interleaved, "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
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G_DEFINE_TYPE (GstDtsDec, gst_dtsdec, GST_TYPE_AUDIO_DECODER);
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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);
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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_class_init (GstDtsDecClass * klass)
{
GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioDecoderClass *gstbase_class;
guint cpuflags;
gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbase_class = (GstAudioDecoderClass *) klass;
gobject_class->set_property = gst_dtsdec_set_property;
gobject_class->get_property = gst_dtsdec_get_property;
gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
gst_element_class_set_static_metadata (gstelement_class, "DTS audio decoder",
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"Codec/Decoder/Audio",
"Decodes DTS audio streams",
"Jan Schmidt <thaytan@noraisin.net>, "
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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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);
/**
* GstDtsDec::drc
*
* Set to true to apply the recommended DTS dynamic range compression
* to the audio stream. Dynamic range compression makes loud sounds
* softer and soft sounds louder, so you can more easily listen
* to the stream without disturbing other people.
*/
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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));
klass->dts_cpuflags = 0;
#if HAVE_ORC
cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx"));
if (cpuflags & ORC_TARGET_MMX_MMX)
klass->dts_cpuflags |= MM_ACCEL_X86_MMX;
if (cpuflags & ORC_TARGET_MMX_3DNOW)
klass->dts_cpuflags |= MM_ACCEL_X86_3DNOW;
if (cpuflags & ORC_TARGET_MMX_MMXEXT)
klass->dts_cpuflags |= MM_ACCEL_X86_MMXEXT;
#else
cpuflags = 0;
klass->dts_cpuflags = 0;
#endif
GST_LOG ("CPU flags: dts=%08x, orc=%08x", klass->dts_cpuflags, cpuflags);
}
static void
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gst_dtsdec_init (GstDtsDec * dtsdec)
{
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dtsdec->request_channels = DCA_CHANNEL;
dtsdec->dynamic_range_compression = FALSE;
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dtsdec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dtsdec));
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/* 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));
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}
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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;
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/* call upon legacy upstream byte support (e.g. seeking) */
gst_audio_decoder_set_estimate_rate (dec, TRUE);
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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 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;
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GstFlowReturn result = GST_FLOW_EOS;
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dts = GST_DTSDEC (bdec);
size = av = gst_adapter_available (adapter);
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data = (guint8 *) gst_adapter_map (adapter, av);
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/* find and read header */
bit_rate = dts->bit_rate;
sample_rate = dts->sample_rate;
flags = 0;
while (size >= 7) {
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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;
}
}
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gst_adapter_unmap (adapter);
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*_offset = av - size;
*len = length;
return result;
}
static gint
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gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition * pos)
{
gint chans = 0;
switch (flags & DCA_CHANNEL_MASK) {
case DCA_MONO:
chans = 1;
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if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
}
break;
/* case DCA_CHANNEL: */
case DCA_STEREO:
case DCA_STEREO_SUMDIFF:
case DCA_STEREO_TOTAL:
case DCA_DOLBY:
chans = 2;
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if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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}
break;
case DCA_3F:
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chans = 3;
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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;
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}
break;
case DCA_2F1R:
chans = 3;
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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;
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}
break;
case DCA_3F1R:
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chans = 4;
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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;
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}
break;
case DCA_2F2R:
chans = 4;
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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;
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}
break;
case DCA_3F2R:
chans = 5;
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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;
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}
break;
case DCA_4F2R:
chans = 6;
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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;
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}
break;
default:
g_warning ("dtsdec: invalid flags 0x%x", flags);
return 0;
}
if (flags & DCA_LFE) {
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if (pos) {
pos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE1;
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}
chans += 1;
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}
return chans;
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
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gint channels;
gboolean result = FALSE;
GstAudioChannelPosition from[7], to[7];
GstAudioInfo info;
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channels = gst_dtsdec_channels (dts->using_channels, from);
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if (channels <= 0 || channels > 7)
goto done;
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GST_INFO_OBJECT (dts, "dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
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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);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
SAMPLE_TYPE, dts->sample_rate, channels, (channels > 1 ? to : NULL));
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dts), &info))
goto done;
result = TRUE;
done:
return result;
}
static void
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
{
GstTagList *taglist;
if (dts->bit_rate > 3) {
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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);
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dts), taglist,
GST_TAG_MERGE_REPLACE);
if (taglist)
gst_tag_list_unref (taglist);
}
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}
static GstFlowReturn
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gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
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GstDtsDec *dts;
gint channels, i, num_blocks;
gboolean need_renegotiation = FALSE;
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guint8 *data;
GstMapInfo map;
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gint chans;
#ifndef G_DISABLE_ASSERT
gsize size;
gint length;
#endif
gint flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_OK;
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GstBuffer *outbuf;
dts = GST_DTSDEC (bdec);
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
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/* parsed stuff already, so this should work out fine */
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
#ifndef G_DISABLE_ASSERT
size = map.size;
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g_assert (size >= 7);
#endif
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bit_rate = dts->bit_rate;
sample_rate = dts->sample_rate;
flags = 0;
#ifndef G_DISABLE_ASSERT
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length = dca_syncinfo (dts->state, data, &flags, &sample_rate, &bit_rate,
&frame_length);
g_assert (length == size);
#else
(void) dca_syncinfo (dts->state, data, &flags, &sample_rate, &bit_rate,
&frame_length);
#endif
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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;
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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;
}
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/* process */
flags |= DCA_ADJUST_LEVEL;
dts->level = 1;
if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
gst_buffer_unmap (buffer, &map);
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GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL),
("dts_frame error"), result);
goto exit;
}
gst_buffer_unmap (buffer, &map);
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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);
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if (!gst_dtsdec_renegotiate (dts))
goto failed_negotiation;
}
if (dts->dynamic_range_compression == FALSE) {
dca_dynrng (dts->state, NULL, NULL);
}
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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);
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outbuf =
gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
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gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
data = map.data;
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{
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];
}
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}
}
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ptr += 256 * chans * (SAMPLE_WIDTH / 8);
}
}
gst_buffer_unmap (outbuf, &map);
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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
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gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
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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 GstFlowReturn
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gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
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GstDtsDec *dts = GST_DTSDEC (parent);
gint first_access;
if (dts->dvdmode) {
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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;
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subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
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ret = dts->base_chain (pad, parent, subbuf);
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if (ret != GST_FLOW_OK) {
gst_buffer_unref (buf);
goto done;
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}
offset += len;
len = size - offset;
if (len > 0) {
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subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
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ret = dts->base_chain (pad, parent, subbuf);
}
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gst_buffer_unref (buf);
} else {
/* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
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subbuf =
gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset,
size - offset);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
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ret = dts->base_chain (pad, parent, subbuf);
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gst_buffer_unref (buf);
}
} else {
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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 void
gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
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case PROP_DRC:
dts->dynamic_range_compression = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
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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
plugin_init (GstPlugin * plugin)
{
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GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder");
#if HAVE_ORC
orc_init ();
#endif
if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
GST_TYPE_DTSDEC))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
dtsdec,
"Decodes DTS audio streams",
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);