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libdca returns the bitrate as 1/2/3 for open/variable/lossless files respectively. This makes sure we don't emit these values.
973 lines
27 KiB
C
973 lines
27 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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/multichannel.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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#elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
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#define SAMPLE_WIDTH 64
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#else
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#define SAMPLE_WIDTH 32
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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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ARG_0,
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ARG_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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#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
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#define DTS_CAPS "audio/x-raw-int, " \
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
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"signed = (boolean) true, " \
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"width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", " \
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"depth = (int) 16"
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#else
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#define DTS_CAPS "audio/x-raw-float, " \
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
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"width = (int) " G_STRINGIFY (SAMPLE_WIDTH)
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#endif
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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 (DTS_CAPS ", "
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"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
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);
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GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstElement, GST_TYPE_ELEMENT);
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static gboolean gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps);
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static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf);
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static GstFlowReturn gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf);
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static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element,
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GstStateChange transition);
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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_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_set_details_simple (element_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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GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder");
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}
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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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guint cpuflags;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) 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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gstelement_class->change_state = gst_dtsdec_change_state;
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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), ARG_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, liboil=%08x", klass->dts_cpuflags, cpuflags);
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}
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static void
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gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
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{
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/* create the sink and src pads */
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dtsdec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
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gst_pad_set_setcaps_function (dtsdec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_dtsdec_sink_setcaps));
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gst_pad_set_chain_function (dtsdec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_dtsdec_chain));
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gst_pad_set_event_function (dtsdec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_dtsdec_sink_event));
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gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad);
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dtsdec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
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gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->srcpad);
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dtsdec->request_channels = DCA_CHANNEL;
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dtsdec->dynamic_range_compression = FALSE;
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gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
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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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GstAudioChannelPosition *tpos = NULL;
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if (pos) {
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/* Allocate the maximum, for ease */
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tpos = *pos = g_new (GstAudioChannelPosition, 7);
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if (!tpos)
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return 0;
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}
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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 (tpos)
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
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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 (tpos) {
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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tpos[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 (tpos) {
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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tpos[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 (tpos) {
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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tpos[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 (tpos) {
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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tpos[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 (tpos) {
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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tpos[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 (tpos) {
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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tpos[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 (tpos) {
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tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
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tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
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tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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tpos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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tpos[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 (tpos) {
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tpos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE;
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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 void
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clear_queued (GstDtsDec * dec)
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{
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g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (dec->queued);
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dec->queued = NULL;
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}
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static GstFlowReturn
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flush_queued (GstDtsDec * dec)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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while (dec->queued) {
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GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
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GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
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GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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/* iterate ouput queue an push downstream */
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ret = gst_pad_push (dec->srcpad, buf);
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dec->queued = g_list_delete_link (dec->queued, dec->queued);
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}
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return ret;
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}
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static GstFlowReturn
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gst_dtsdec_drain (GstDtsDec * dec)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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if (dec->segment.rate < 0.0) {
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/* if we have some queued frames for reverse playback, flush
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* them now */
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ret = flush_queued (dec);
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}
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return ret;
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}
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static GstFlowReturn
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gst_dtsdec_push (GstDtsDec * dtsdec,
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GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
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{
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GstBuffer *buf;
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int chans, n, c;
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GstFlowReturn result;
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flags &= (DCA_CHANNEL_MASK | DCA_LFE);
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chans = gst_dtsdec_channels (flags, NULL);
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if (!chans) {
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GST_ELEMENT_ERROR (GST_ELEMENT (dtsdec), STREAM, DECODE, (NULL),
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("Invalid channel flags: %d", flags));
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return GST_FLOW_ERROR;
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}
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result =
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gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
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256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
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if (result != GST_FLOW_OK)
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return result;
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for (n = 0; n < 256; n++) {
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for (c = 0; c < chans; c++) {
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((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
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samples[c * 256 + n];
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}
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}
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GST_BUFFER_TIMESTAMP (buf) = timestamp;
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GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / dtsdec->sample_rate;
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result = GST_FLOW_OK;
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if ((buf = gst_audio_buffer_clip (buf, &dtsdec->segment,
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dtsdec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
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/* set discont when needed */
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if (dtsdec->discont) {
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GST_LOG_OBJECT (dtsdec, "marking DISCONT");
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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dtsdec->discont = FALSE;
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}
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if (dtsdec->segment.rate > 0.0) {
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GST_DEBUG_OBJECT (dtsdec,
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"Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
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GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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result = gst_pad_push (srcpad, buf);
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} else {
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/* reverse playback, queue frame till later when we get a discont. */
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GST_DEBUG_OBJECT (dtsdec, "queued frame");
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dtsdec->queued = g_list_prepend (dtsdec->queued, buf);
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}
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}
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return result;
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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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GstAudioChannelPosition *pos;
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GstCaps *caps = gst_caps_from_string (DTS_CAPS);
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gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
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gboolean result = FALSE;
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if (!channels)
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goto done;
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GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
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channels, dts->sample_rate);
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gst_caps_set_simple (caps,
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"channels", G_TYPE_INT, channels,
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"rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
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gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
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g_free (pos);
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if (!gst_pad_set_caps (dts->srcpad, caps))
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goto done;
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|
|
result = TRUE;
|
|
|
|
done:
|
|
if (caps) {
|
|
gst_caps_unref (caps);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstDtsDec *dtsdec = GST_DTSDEC (gst_pad_get_parent (pad));
|
|
gboolean ret = FALSE;
|
|
|
|
GST_LOG_OBJECT (dtsdec, "%s event", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_NEWSEGMENT:{
|
|
GstFormat format;
|
|
gboolean update;
|
|
gint64 start, end, pos;
|
|
gdouble rate;
|
|
|
|
gst_event_parse_new_segment (event, &update, &rate, &format, &start, &end,
|
|
&pos);
|
|
|
|
/* drain queued buffers before activating the segment so that we can clip
|
|
* against the old segment first */
|
|
gst_dtsdec_drain (dtsdec);
|
|
|
|
if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) {
|
|
GST_WARNING ("No time in newsegment event %p (format is %s)",
|
|
event, gst_format_get_name (format));
|
|
gst_event_unref (event);
|
|
dtsdec->sent_segment = FALSE;
|
|
/* set some dummy values, FIXME: do proper conversion */
|
|
dtsdec->time = start = pos = 0;
|
|
format = GST_FORMAT_TIME;
|
|
end = -1;
|
|
} else {
|
|
dtsdec->time = start;
|
|
dtsdec->sent_segment = TRUE;
|
|
ret = gst_pad_push_event (dtsdec->srcpad, event);
|
|
}
|
|
|
|
gst_segment_set_newsegment (&dtsdec->segment, update, rate, format, start,
|
|
end, pos);
|
|
break;
|
|
}
|
|
case GST_EVENT_TAG:
|
|
ret = gst_pad_push_event (dtsdec->srcpad, event);
|
|
break;
|
|
case GST_EVENT_EOS:
|
|
gst_dtsdec_drain (dtsdec);
|
|
ret = gst_pad_push_event (dtsdec->srcpad, event);
|
|
break;
|
|
case GST_EVENT_FLUSH_START:
|
|
ret = gst_pad_push_event (dtsdec->srcpad, event);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
if (dtsdec->cache) {
|
|
gst_buffer_unref (dtsdec->cache);
|
|
dtsdec->cache = NULL;
|
|
}
|
|
clear_queued (dtsdec);
|
|
gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
|
|
ret = gst_pad_push_event (dtsdec->srcpad, event);
|
|
break;
|
|
default:
|
|
ret = gst_pad_push_event (dtsdec->srcpad, event);
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (dtsdec);
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
|
|
{
|
|
GstTagList *taglist;
|
|
|
|
taglist = gst_tag_list_new ();
|
|
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
|
|
GST_TAG_AUDIO_CODEC, "DTS DCA", NULL);
|
|
|
|
if (dts->bit_rate > 3) {
|
|
/* 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_element_found_tags_for_pad (GST_ELEMENT (dts), dts->srcpad, taglist);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
|
|
guint length, gint flags, gint sample_rate, gint bit_rate)
|
|
{
|
|
gint channels, i, num_blocks;
|
|
gboolean need_renegotiation = FALSE;
|
|
|
|
/* 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 (dts->srcpad);
|
|
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_WARNING_OBJECT (dts, "dts_frame error");
|
|
dts->discont = TRUE;
|
|
return GST_FLOW_OK;
|
|
}
|
|
channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
|
|
if (dts->using_channels != channels) {
|
|
need_renegotiation = TRUE;
|
|
dts->using_channels = channels;
|
|
}
|
|
|
|
/* negotiate if required */
|
|
if (need_renegotiation) {
|
|
GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
|
|
dts->sample_rate, dts->stream_channels, dts->using_channels);
|
|
if (!gst_dtsdec_renegotiate (dts)) {
|
|
GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
if (dts->dynamic_range_compression == FALSE) {
|
|
dca_dynrng (dts->state, NULL, NULL);
|
|
}
|
|
|
|
/* handle decoded data, one block is 256 samples */
|
|
num_blocks = dca_blocks_num (dts->state);
|
|
for (i = 0; i < num_blocks; i++) {
|
|
if (dca_block (dts->state)) {
|
|
/* Ignore errors, but mark a discont */
|
|
GST_WARNING_OBJECT (dts, "dts_block error %d", i);
|
|
dts->discont = TRUE;
|
|
} else {
|
|
GstFlowReturn ret;
|
|
|
|
/* push on */
|
|
ret = gst_dtsdec_push (dts, dts->srcpad, dts->using_channels,
|
|
dts->samples, dts->time);
|
|
if (ret != GST_FLOW_OK)
|
|
return ret;
|
|
}
|
|
dts->time += GST_SECOND * 256 / dts->sample_rate;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
|
|
GstStructure *structure;
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
|
|
dts->dvdmode = TRUE;
|
|
else
|
|
dts->dvdmode = FALSE;
|
|
|
|
gst_object_unref (dts);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
|
|
gint first_access;
|
|
|
|
if (GST_BUFFER_IS_DISCONT (buf)) {
|
|
GST_LOG_OBJECT (dts, "received DISCONT");
|
|
gst_dtsdec_drain (dts);
|
|
/* clear cache on discont and mark a discont in the element */
|
|
if (dts->cache) {
|
|
gst_buffer_unref (dts->cache);
|
|
dts->cache = NULL;
|
|
}
|
|
dts->discont = TRUE;
|
|
}
|
|
|
|
if (dts->dvdmode) {
|
|
gint size = GST_BUFFER_SIZE (buf);
|
|
guint8 *data = GST_BUFFER_DATA (buf);
|
|
gint offset, len;
|
|
GstBuffer *subbuf;
|
|
|
|
if (size < 2)
|
|
goto not_enough_data;
|
|
|
|
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_create_sub (buf, offset, len);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
|
|
ret = gst_dtsdec_chain_raw (pad, subbuf);
|
|
if (ret != GST_FLOW_OK)
|
|
goto done;
|
|
|
|
offset += len;
|
|
len = size - offset;
|
|
|
|
if (len > 0) {
|
|
subbuf = gst_buffer_create_sub (buf, offset, len);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
ret = gst_dtsdec_chain_raw (pad, subbuf);
|
|
}
|
|
} else {
|
|
/* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
|
|
subbuf = gst_buffer_create_sub (buf, offset, size - offset);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
|
|
ret = gst_dtsdec_chain_raw (pad, subbuf);
|
|
}
|
|
} else {
|
|
gst_buffer_ref (buf);
|
|
ret = gst_dtsdec_chain_raw (pad, buf);
|
|
}
|
|
|
|
done:
|
|
gst_buffer_unref (buf);
|
|
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 GstFlowReturn
|
|
gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstDtsDec *dts;
|
|
guint8 *data;
|
|
gint size;
|
|
gint length = 0, flags, sample_rate, bit_rate, frame_length;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
|
|
dts = GST_DTSDEC (GST_PAD_PARENT (pad));
|
|
|
|
if (!dts->sent_segment) {
|
|
GstSegment segment;
|
|
|
|
/* Create a basic segment. Usually, we'll get a new-segment sent by
|
|
* another element that will know more information (a demuxer). If we're
|
|
* just looking at a raw AC3 stream, we won't - so we need to send one
|
|
* here, but we don't know much info, so just send a minimal TIME
|
|
* new-segment event
|
|
*/
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
gst_pad_push_event (dts->srcpad, gst_event_new_new_segment (FALSE,
|
|
segment.rate, segment.format, segment.start,
|
|
segment.duration, segment.start));
|
|
dts->sent_segment = TRUE;
|
|
}
|
|
|
|
/* merge with cache, if any. Also make sure timestamps match */
|
|
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
|
|
dts->time = GST_BUFFER_TIMESTAMP (buf);
|
|
GST_DEBUG_OBJECT (dts,
|
|
"Received buffer with ts %" GST_TIME_FORMAT " duration %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
}
|
|
|
|
if (dts->cache) {
|
|
buf = gst_buffer_join (dts->cache, buf);
|
|
dts->cache = NULL;
|
|
}
|
|
data = GST_BUFFER_DATA (buf);
|
|
size = GST_BUFFER_SIZE (buf);
|
|
|
|
/* find and read header */
|
|
bit_rate = dts->bit_rate;
|
|
sample_rate = dts->sample_rate;
|
|
flags = 0;
|
|
while (size >= 7) {
|
|
length = dca_syncinfo (dts->state, data, &flags,
|
|
&sample_rate, &bit_rate, &frame_length);
|
|
|
|
if (length == 0) {
|
|
/* shift window to re-find sync */
|
|
data++;
|
|
size--;
|
|
} else if (length <= size) {
|
|
GST_DEBUG ("Sync: frame size %d", length);
|
|
|
|
if (flags != dts->prev_flags)
|
|
dts->flag_update = TRUE;
|
|
dts->prev_flags = flags;
|
|
|
|
result = gst_dtsdec_handle_frame (dts, data, length,
|
|
flags, sample_rate, bit_rate);
|
|
if (result != GST_FLOW_OK) {
|
|
size = 0;
|
|
break;
|
|
}
|
|
size -= length;
|
|
data += length;
|
|
} else {
|
|
GST_LOG ("Not enough data available (needed %d had %d)", length, size);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* keep cache */
|
|
if (length == 0) {
|
|
GST_LOG ("No sync found");
|
|
}
|
|
|
|
if (size > 0) {
|
|
dts->cache = gst_buffer_create_sub (buf,
|
|
GST_BUFFER_SIZE (buf) - size, size);
|
|
}
|
|
|
|
gst_buffer_unref (buf);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstDtsDec *dts = GST_DTSDEC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:{
|
|
GstDtsDecClass *klass;
|
|
|
|
klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
|
|
dts->state = dca_init (klass->dts_cpuflags);
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
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->time = 0;
|
|
dts->sent_segment = FALSE;
|
|
dts->flag_update = TRUE;
|
|
gst_segment_init (&dts->segment, GST_FORMAT_UNDEFINED);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
dts->samples = NULL;
|
|
if (dts->cache) {
|
|
gst_buffer_unref (dts->cache);
|
|
dts->cache = NULL;
|
|
}
|
|
clear_queued (dts);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
dca_free (dts->state);
|
|
dts->state = NULL;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
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 ARG_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 ARG_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)
|
|
{
|
|
#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);
|