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788 lines
22 KiB
C
788 lines
22 KiB
C
/* GStreamer DTS decoder plugin based on libdtsdec
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* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
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* Copyright (C) 2009 Jan Schmidt <thaytan@noraisin.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-dtsdec
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*
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* Digital Theatre System (DTS) audio decoder
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch dvdreadsrc title=1 ! mpegpsdemux ! dtsdec ! audioresample ! audioconvert ! alsasink
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* ]| Play a DTS audio track from a dvd.
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* |[
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* gst-launch filesrc location=abc.dts ! dtsdec ! audioresample ! audioconvert ! alsasink
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* ]| Decode a standalone file and play it.
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "_stdint.h"
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#include <stdlib.h>
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#ifndef DTS_OLD
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#include <dca.h>
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#else
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#include <dts.h>
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typedef struct dts_state_s dca_state_t;
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#define DCA_MONO DTS_MONO
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#define DCA_CHANNEL DTS_CHANNEL
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#define DCA_STEREO DTS_STEREO
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#define DCA_STEREO_SUMDIFF DTS_STEREO_SUMDIFF
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#define DCA_STEREO_TOTAL DTS_STEREO_TOTAL
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#define DCA_3F DTS_3F
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#define DCA_2F1R DTS_2F1R
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#define DCA_3F1R DTS_3F1R
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#define DCA_2F2R DTS_2F2R
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#define DCA_3F2R DTS_3F2R
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#define DCA_4F2R DTS_4F2R
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#define DCA_DOLBY DTS_DOLBY
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#define DCA_CHANNEL_MAX DTS_CHANNEL_MAX
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#define DCA_CHANNEL_BITS DTS_CHANNEL_BITS
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#define DCA_CHANNEL_MASK DTS_CHANNEL_MASK
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#define DCA_LFE DTS_LFE
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#define DCA_ADJUST_LEVEL DTS_ADJUST_LEVEL
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#define dca_init dts_init
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#define dca_syncinfo dts_syncinfo
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#define dca_frame dts_frame
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#define dca_dynrng dts_dynrng
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#define dca_blocks_num dts_blocks_num
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#define dca_block dts_block
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#define dca_samples dts_samples
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#define dca_free dts_free
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#endif
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#include "gstdtsdec.h"
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#if HAVE_ORC
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#include <orc/orc.h>
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#endif
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#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
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#define SAMPLE_WIDTH 16
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#define SAMPLE_FORMAT GST_AUDIO_NE(S16)
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#define SAMPLE_TYPE GST_AUDIO_FORMAT_S16
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#elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
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#define SAMPLE_WIDTH 64
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#define SAMPLE_FORMAT GST_AUDIO_NE(F64)
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#define SAMPLE_TYPE GST_AUDIO_FORMAT_F64
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#else
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#define SAMPLE_WIDTH 32
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#define SAMPLE_FORMAT GST_AUDIO_NE(F32)
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#define SAMPLE_TYPE GST_AUDIO_FORMAT_F32
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#endif
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GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
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#define GST_CAT_DEFAULT (dtsdec_debug)
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enum
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{
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PROP_0,
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PROP_DRC
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};
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-dts; audio/x-private1-dts")
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);
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " SAMPLE_FORMAT ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
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);
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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);
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static gboolean gst_dtsdec_stop (GstAudioDecoder * dec);
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static gboolean gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
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static gboolean gst_dtsdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length);
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static GstFlowReturn gst_dtsdec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buf);
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static void gst_dtsdec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_dtsdec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void
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gst_dtsdec_class_init (GstDtsDecClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioDecoderClass *gstbase_class;
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guint cpuflags;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbase_class = (GstAudioDecoderClass *) klass;
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gobject_class->set_property = gst_dtsdec_set_property;
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gobject_class->get_property = gst_dtsdec_get_property;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_set_static_metadata (gstelement_class, "DTS audio decoder",
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"Codec/Decoder/Audio",
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"Decodes DTS audio streams",
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"Jan Schmidt <thaytan@noraisin.net>, "
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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gstbase_class->start = GST_DEBUG_FUNCPTR (gst_dtsdec_start);
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gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_dtsdec_stop);
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gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_dtsdec_set_format);
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gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_dtsdec_parse);
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gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_dtsdec_handle_frame);
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/**
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* GstDtsDec::drc
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*
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* Set to true to apply the recommended DTS dynamic range compression
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* to the audio stream. Dynamic range compression makes loud sounds
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* softer and soft sounds louder, so you can more easily listen
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* to the stream without disturbing other people.
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DRC,
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g_param_spec_boolean ("drc", "Dynamic Range Compression",
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"Use Dynamic Range Compression", FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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klass->dts_cpuflags = 0;
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#if HAVE_ORC
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cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx"));
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if (cpuflags & ORC_TARGET_MMX_MMX)
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klass->dts_cpuflags |= MM_ACCEL_X86_MMX;
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if (cpuflags & ORC_TARGET_MMX_3DNOW)
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klass->dts_cpuflags |= MM_ACCEL_X86_3DNOW;
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if (cpuflags & ORC_TARGET_MMX_MMXEXT)
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klass->dts_cpuflags |= MM_ACCEL_X86_MMXEXT;
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#else
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cpuflags = 0;
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klass->dts_cpuflags = 0;
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#endif
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GST_LOG ("CPU flags: dts=%08x, orc=%08x", klass->dts_cpuflags, cpuflags);
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}
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static void
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gst_dtsdec_init (GstDtsDec * dtsdec)
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{
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dtsdec->request_channels = DCA_CHANNEL;
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dtsdec->dynamic_range_compression = FALSE;
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/* retrieve and intercept base class chain.
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* Quite HACKish, but that's dvd specs for you,
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* since one buffer needs to be split into 2 frames */
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dtsdec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (dtsdec));
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gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (dtsdec),
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GST_DEBUG_FUNCPTR (gst_dtsdec_chain));
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}
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static gboolean
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gst_dtsdec_start (GstAudioDecoder * dec)
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{
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GstDtsDec *dts = GST_DTSDEC (dec);
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GstDtsDecClass *klass;
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GST_DEBUG_OBJECT (dec, "start");
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klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
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dts->state = dca_init (klass->dts_cpuflags);
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dts->samples = dca_samples (dts->state);
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dts->bit_rate = -1;
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dts->sample_rate = -1;
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dts->stream_channels = DCA_CHANNEL;
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dts->using_channels = DCA_CHANNEL;
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dts->level = 1;
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dts->bias = 0;
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dts->flag_update = TRUE;
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/* call upon legacy upstream byte support (e.g. seeking) */
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gst_audio_decoder_set_estimate_rate (dec, TRUE);
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return TRUE;
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}
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static gboolean
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gst_dtsdec_stop (GstAudioDecoder * dec)
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{
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GstDtsDec *dts = GST_DTSDEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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dts->samples = NULL;
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if (dts->state) {
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dca_free (dts->state);
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dts->state = NULL;
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
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gint * _offset, gint * len)
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{
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GstDtsDec *dts;
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guint8 *data;
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gint av, size;
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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);
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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 */
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bit_rate = dts->bit_rate;
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sample_rate = dts->sample_rate;
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flags = 0;
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while (size >= 7) {
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length = dca_syncinfo (dts->state, data, &flags,
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&sample_rate, &bit_rate, &frame_length);
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if (length == 0) {
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/* shift window to re-find sync */
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data++;
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size--;
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} else if (length <= size) {
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GST_LOG_OBJECT (dts, "Sync: frame size %d", length);
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result = GST_FLOW_OK;
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break;
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} else {
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GST_LOG_OBJECT (dts, "Not enough data available (needed %d had %d)",
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length, size);
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break;
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}
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}
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gst_adapter_unmap (adapter);
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*_offset = av - size;
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*len = length;
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return result;
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}
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static gint
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gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition * pos)
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{
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gint chans = 0;
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switch (flags & DCA_CHANNEL_MASK) {
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case DCA_MONO:
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chans = 1;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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}
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break;
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/* case DCA_CHANNEL: */
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case DCA_STEREO:
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case DCA_STEREO_SUMDIFF:
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case DCA_STEREO_TOTAL:
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case DCA_DOLBY:
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chans = 2;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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}
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break;
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case DCA_3F:
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chans = 3;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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}
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break;
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case DCA_2F1R:
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chans = 3;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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}
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break;
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case DCA_3F1R:
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chans = 4;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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}
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break;
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case DCA_2F2R:
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chans = 4;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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}
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break;
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case DCA_3F2R:
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chans = 5;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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}
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break;
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case DCA_4F2R:
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chans = 6;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
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pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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pos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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}
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break;
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default:
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g_warning ("dtsdec: invalid flags 0x%x", flags);
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return 0;
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}
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if (flags & DCA_LFE) {
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if (pos) {
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pos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE1;
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}
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chans += 1;
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}
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return chans;
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}
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static gboolean
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gst_dtsdec_renegotiate (GstDtsDec * dts)
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{
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gint channels;
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gboolean result = FALSE;
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GstAudioChannelPosition from[6], to[6];
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GstAudioInfo info;
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channels = gst_dtsdec_channels (dts->using_channels, from);
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if (!channels)
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goto done;
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GST_INFO_OBJECT (dts, "dtsdec renegotiate, channels=%d, rate=%d",
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channels, dts->sample_rate);
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memcpy (to, from, sizeof (GstAudioChannelPosition) * channels);
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gst_audio_channel_positions_to_valid_order (to, channels);
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gst_audio_get_channel_reorder_map (channels, from, to,
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dts->channel_reorder_map);
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info,
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SAMPLE_TYPE, dts->sample_rate, channels, (channels > 1 ? to : NULL));
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if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dts), &info))
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goto done;
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result = TRUE;
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done:
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return result;
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}
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static void
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gst_dtsdec_update_streaminfo (GstDtsDec * dts)
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{
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GstTagList *taglist;
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if (dts->bit_rate > 3) {
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taglist = gst_tag_list_new_empty ();
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/* 1 => open bitrate, 2 => variable bitrate, 3 => lossless */
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gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
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(guint) dts->bit_rate, NULL);
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gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dts), taglist,
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GST_TAG_MERGE_REPLACE);
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}
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}
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static GstFlowReturn
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gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
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{
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GstDtsDec *dts;
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gint channels, i, num_blocks;
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gboolean need_renegotiation = FALSE;
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guint8 *data;
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gsize size;
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GstMapInfo map;
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gint chans;
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gint length = 0, flags, sample_rate, bit_rate, frame_length;
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GstFlowReturn result = GST_FLOW_OK;
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GstBuffer *outbuf;
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dts = GST_DTSDEC (bdec);
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|
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/* no fancy draining */
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if (G_UNLIKELY (!buffer))
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return GST_FLOW_OK;
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|
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/* parsed stuff already, so this should work out fine */
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
data = map.data;
|
|
size = map.size;
|
|
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) {
|
|
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;
|
|
|
|
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;
|
|
}
|
|
|
|
/* process */
|
|
flags |= DCA_ADJUST_LEVEL;
|
|
dts->level = 1;
|
|
if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
|
|
gst_buffer_unmap (buffer, &map);
|
|
GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL),
|
|
("dts_frame error"), result);
|
|
goto exit;
|
|
}
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
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);
|
|
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);
|
|
outbuf =
|
|
gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
|
|
|
|
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
|
|
data = map.data;
|
|
size = map.size;
|
|
{
|
|
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);
|
|
}
|
|
}
|
|
gst_buffer_unmap (outbuf, &map);
|
|
|
|
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_set_format (GstAudioDecoder * bdec, GstCaps * caps)
|
|
{
|
|
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
|
|
gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstDtsDec *dts = GST_DTSDEC (parent);
|
|
gint first_access;
|
|
|
|
if (dts->dvdmode) {
|
|
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;
|
|
|
|
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, parent, subbuf);
|
|
if (ret != GST_FLOW_OK) {
|
|
gst_buffer_unref (buf);
|
|
goto done;
|
|
}
|
|
|
|
offset += len;
|
|
len = size - offset;
|
|
|
|
if (len > 0) {
|
|
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, 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_copy_region (buf, GST_BUFFER_COPY_ALL, offset,
|
|
size - offset);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
|
|
ret = dts->base_chain (pad, parent, subbuf);
|
|
gst_buffer_unref (buf);
|
|
}
|
|
} else {
|
|
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) {
|
|
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) {
|
|
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)
|
|
{
|
|
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);
|