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838 lines
24 KiB
C
838 lines
24 KiB
C
/* GStreamer
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* Copyright (C) <2001> David I. Lehn <dlehn@users.sourceforge.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-a52dec
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*
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* Dolby Digital (AC-3) 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-1.0 dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioconvert ! audioresample ! autoaudiosink
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* ]| Play audio part of a dvd title.
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* |[
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* gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
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* ]| Decode and play a stand alone AC-3 file.
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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 <stdlib.h>
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#include "_stdint.h"
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#include <gst/gst.h>
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#include <a52dec/a52.h>
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#if !defined(A52_ACCEL_DETECT)
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# include <a52dec/mm_accel.h>
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#endif
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#include "gsta52dec.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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#ifdef LIBA52_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 (a52dec_debug);
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#define GST_CAT_DEFAULT (a52dec_debug)
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/* A52Dec args */
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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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ARG_MODE,
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ARG_LFE,
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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-ac3; audio/ac3; audio/x-private1-ac3")
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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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#define gst_a52dec_parent_class parent_class
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G_DEFINE_TYPE (GstA52Dec, gst_a52dec, GST_TYPE_AUDIO_DECODER);
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static gboolean gst_a52dec_start (GstAudioDecoder * dec);
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static gboolean gst_a52dec_stop (GstAudioDecoder * dec);
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static gboolean gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
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static gboolean gst_a52dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length);
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static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static void gst_a52dec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_a52dec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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#define GST_TYPE_A52DEC_MODE (gst_a52dec_mode_get_type())
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static GType
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gst_a52dec_mode_get_type (void)
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{
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static GType a52dec_mode_type = 0;
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static const GEnumValue a52dec_modes[] = {
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{A52_MONO, "Mono", "mono"},
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{A52_STEREO, "Stereo", "stereo"},
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{A52_3F, "3 Front", "3f"},
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{A52_2F1R, "2 Front, 1 Rear", "2f1r"},
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{A52_3F1R, "3 Front, 1 Rear", "3f1r"},
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{A52_2F2R, "2 Front, 2 Rear", "2f2r"},
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{A52_3F2R, "3 Front, 2 Rear", "3f2r"},
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{A52_DOLBY, "Dolby", "dolby"},
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{0, NULL, NULL},
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};
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if (!a52dec_mode_type) {
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a52dec_mode_type = g_enum_register_static ("GstA52DecMode", a52dec_modes);
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}
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return a52dec_mode_type;
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}
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static void
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gst_a52dec_class_init (GstA52DecClass * 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 = 0;
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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_a52dec_set_property;
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gobject_class->get_property = gst_a52dec_get_property;
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gstbase_class->start = GST_DEBUG_FUNCPTR (gst_a52dec_start);
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gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_a52dec_stop);
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gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_a52dec_set_format);
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gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_a52dec_parse);
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gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_a52dec_handle_frame);
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/**
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* GstA52Dec::drc
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*
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* Set to true to apply the recommended Dolby Digital 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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/**
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* GstA52Dec::mode
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*
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* Force a particular output channel configuration from the decoder. By default,
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* the channel downmix (if any) is chosen automatically based on the downstream
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* capabilities of the pipeline.
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
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g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)",
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GST_TYPE_A52DEC_MODE, A52_3F2R,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstA52Dec::lfe
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*
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* Whether to output the LFE (Low Frequency Emitter) channel of the audio stream.
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LFE,
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g_param_spec_boolean ("lfe", "LFE", "LFE", TRUE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
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gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
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gst_element_class_set_static_metadata (gstelement_class,
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"ATSC A/52 audio decoder", "Codec/Decoder/Audio",
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"Decodes ATSC A/52 encoded audio streams",
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"David I. Lehn <dlehn@users.sourceforge.net>");
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GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0,
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"AC3/A52 software decoder");
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/* If no CPU instruction based acceleration is available, end up using the
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* generic software djbfft based one when available in the used liba52 */
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#ifdef MM_ACCEL_DJBFFT
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klass->a52_cpuflags = MM_ACCEL_DJBFFT;
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#elif defined(A52_ACCEL_DETECT)
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klass->a52_cpuflags = A52_ACCEL_DETECT;
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#else
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klass->a52_cpuflags = 0;
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#endif
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#if HAVE_ORC && !defined(A52_ACCEL_DETECT)
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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->a52_cpuflags |= MM_ACCEL_X86_MMX;
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if (cpuflags & ORC_TARGET_MMX_3DNOW)
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klass->a52_cpuflags |= MM_ACCEL_X86_3DNOW;
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if (cpuflags & ORC_TARGET_MMX_MMXEXT)
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klass->a52_cpuflags |= MM_ACCEL_X86_MMXEXT;
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#endif
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GST_LOG ("CPU flags: a52=%08x, orc=%08x", klass->a52_cpuflags, cpuflags);
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}
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static void
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gst_a52dec_init (GstA52Dec * a52dec)
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{
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a52dec->request_channels = A52_CHANNEL;
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a52dec->dynamic_range_compression = FALSE;
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a52dec->state = NULL;
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a52dec->samples = NULL;
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(a52dec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (a52dec));
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/* retrieve and intercept base class chain.
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* Quite HACKish, but that's dvd specs/caps for you,
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* since one buffer needs to be split into 2 frames */
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a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
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gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
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GST_DEBUG_FUNCPTR (gst_a52dec_chain));
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}
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static gboolean
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gst_a52dec_start (GstAudioDecoder * dec)
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{
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GstA52Dec *a52dec = GST_A52DEC (dec);
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GstA52DecClass *klass;
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static GMutex init_mutex;
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GST_DEBUG_OBJECT (dec, "start");
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klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
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g_mutex_lock (&init_mutex);
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#if defined(A52_ACCEL_DETECT)
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a52dec->state = a52_init ();
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/* This line is just to avoid being accused of not using klass */
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a52_accel (klass->a52_cpuflags & A52_ACCEL_DETECT);
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#else
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a52dec->state = a52_init (klass->a52_cpuflags);
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#endif
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g_mutex_unlock (&init_mutex);
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if (!a52dec->state) {
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GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL),
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("failed to initialize a52 state"));
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return FALSE;
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}
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a52dec->samples = a52_samples (a52dec->state);
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a52dec->bit_rate = -1;
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a52dec->sample_rate = -1;
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a52dec->stream_channels = A52_CHANNEL;
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a52dec->using_channels = A52_CHANNEL;
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a52dec->level = 1;
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a52dec->bias = 0;
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a52dec->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_a52dec_stop (GstAudioDecoder * dec)
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{
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GstA52Dec *a52dec = GST_A52DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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a52dec->samples = NULL;
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if (a52dec->state) {
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a52_free (a52dec->state);
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a52dec->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_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
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gint * _offset, gint * len)
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{
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GstA52Dec *a52dec;
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const guint8 *data;
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gint av, size;
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gint length = 0, flags, sample_rate, bit_rate;
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GstFlowReturn result = GST_FLOW_EOS;
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a52dec = GST_A52DEC (bdec);
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size = av = gst_adapter_available (adapter);
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data = (const guint8 *) gst_adapter_map (adapter, av);
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/* find and read header */
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bit_rate = a52dec->bit_rate;
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sample_rate = a52dec->sample_rate;
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flags = 0;
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while (size >= 7) {
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length = a52_syncinfo ((guint8 *) data, &flags, &sample_rate, &bit_rate);
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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 (a52dec, "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 (a52dec, "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_a52dec_channels (int flags, GstAudioChannelPosition * pos)
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{
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gint chans = 0;
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if (flags & A52_LFE) {
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chans += 1;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE1;
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}
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}
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flags &= A52_CHANNEL_MASK;
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switch (flags) {
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case A52_3F2R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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}
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chans += 5;
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break;
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case A52_2F2R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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}
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chans += 4;
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break;
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case A52_3F1R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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}
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chans += 4;
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break;
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case A52_2F1R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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}
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chans += 3;
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break;
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case A52_3F:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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}
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chans += 3;
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break;
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case A52_CHANNEL: /* Dual mono. Should really be handled as 2 src pads */
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case A52_STEREO:
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case A52_DOLBY:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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}
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chans += 2;
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break;
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case A52_MONO:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_MONO;
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}
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chans += 1;
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break;
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default:
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/* error, caller should post error message */
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return 0;
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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_a52dec_reneg (GstA52Dec * a52dec)
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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_a52dec_channels (a52dec->using_channels, from);
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if (!channels)
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goto done;
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GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d",
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channels, a52dec->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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a52dec->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, a52dec->sample_rate, channels, (channels > 1 ? to : NULL));
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if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (a52dec), &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_a52dec_update_streaminfo (GstA52Dec * a52dec)
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{
|
|
GstTagList *taglist;
|
|
|
|
taglist = gst_tag_list_new_empty ();
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
|
|
(guint) a52dec->bit_rate, NULL);
|
|
|
|
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (a52dec), taglist,
|
|
GST_TAG_MERGE_REPLACE);
|
|
gst_tag_list_unref (taglist);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
|
|
{
|
|
GstA52Dec *a52dec;
|
|
gint channels, i;
|
|
gboolean need_reneg = FALSE;
|
|
gint chans;
|
|
gint length = 0, flags, sample_rate, bit_rate;
|
|
GstMapInfo map;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
GstBuffer *outbuf;
|
|
const gint num_blocks = 6;
|
|
|
|
a52dec = GST_A52DEC (bdec);
|
|
|
|
/* no fancy draining */
|
|
if (G_UNLIKELY (!buffer))
|
|
return GST_FLOW_OK;
|
|
|
|
/* parsed stuff already, so this should work out fine */
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
g_assert (map.size >= 7);
|
|
|
|
/* re-obtain some sync header info,
|
|
* should be same as during _parse and could also be cached there,
|
|
* but anyway ... */
|
|
bit_rate = a52dec->bit_rate;
|
|
sample_rate = a52dec->sample_rate;
|
|
flags = 0;
|
|
length = a52_syncinfo (map.data, &flags, &sample_rate, &bit_rate);
|
|
g_assert (length == map.size);
|
|
|
|
/* update stream information, renegotiate or re-streaminfo if needed */
|
|
need_reneg = FALSE;
|
|
if (a52dec->sample_rate != sample_rate) {
|
|
GST_DEBUG_OBJECT (a52dec, "sample rate changed");
|
|
need_reneg = TRUE;
|
|
a52dec->sample_rate = sample_rate;
|
|
}
|
|
|
|
if (flags) {
|
|
if (a52dec->stream_channels != (flags & (A52_CHANNEL_MASK | A52_LFE))) {
|
|
GST_DEBUG_OBJECT (a52dec, "stream channel flags changed, marking update");
|
|
a52dec->flag_update = TRUE;
|
|
}
|
|
a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
|
|
}
|
|
|
|
if (bit_rate != a52dec->bit_rate) {
|
|
a52dec->bit_rate = bit_rate;
|
|
gst_a52dec_update_streaminfo (a52dec);
|
|
}
|
|
|
|
/* 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.
|
|
*/
|
|
if (a52dec->request_channels != A52_CHANNEL) {
|
|
flags = a52dec->request_channels;
|
|
} else if (a52dec->flag_update) {
|
|
GstCaps *caps;
|
|
|
|
a52dec->flag_update = FALSE;
|
|
|
|
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
|
|
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 orig_channels = flags ? gst_a52dec_channels (flags, NULL) : 6;
|
|
gint fixed_channels = 0;
|
|
const int a52_channels[6] = {
|
|
A52_MONO,
|
|
A52_STEREO,
|
|
A52_STEREO | A52_LFE,
|
|
A52_2F2R,
|
|
A52_2F2R | A52_LFE,
|
|
A52_3F2R | A52_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",
|
|
orig_channels);
|
|
|
|
if (gst_structure_get_int (structure, "channels", &fixed_channels)
|
|
&& fixed_channels <= 6) {
|
|
if (fixed_channels < orig_channels)
|
|
flags = a52_channels[fixed_channels - 1];
|
|
} else {
|
|
flags = a52_channels[5];
|
|
}
|
|
|
|
gst_caps_unref (copy);
|
|
} else if (flags)
|
|
flags = a52dec->stream_channels;
|
|
else
|
|
flags = A52_3F2R | A52_LFE;
|
|
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
flags = a52dec->using_channels;
|
|
}
|
|
|
|
/* process */
|
|
flags |= A52_ADJUST_LEVEL;
|
|
a52dec->level = 1;
|
|
if (a52_frame (a52dec->state, map.data, &flags, &a52dec->level, a52dec->bias)) {
|
|
gst_buffer_unmap (buffer, &map);
|
|
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
|
|
("a52_frame error"), result);
|
|
goto exit;
|
|
}
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
channels = flags & (A52_CHANNEL_MASK | A52_LFE);
|
|
if (a52dec->using_channels != channels) {
|
|
need_reneg = TRUE;
|
|
a52dec->using_channels = channels;
|
|
}
|
|
|
|
/* negotiate if required */
|
|
if (need_reneg) {
|
|
GST_DEBUG_OBJECT (a52dec,
|
|
"a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
|
|
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
|
|
if (!gst_a52dec_reneg (a52dec))
|
|
goto failed_negotiation;
|
|
}
|
|
|
|
if (a52dec->dynamic_range_compression == FALSE) {
|
|
a52_dynrng (a52dec->state, NULL, NULL);
|
|
}
|
|
|
|
flags &= (A52_CHANNEL_MASK | A52_LFE);
|
|
chans = gst_a52dec_channels (flags, NULL);
|
|
if (!chans)
|
|
goto invalid_flags;
|
|
|
|
/* handle decoded data;
|
|
* each frame has 6 blocks, one block is 256 samples, ea */
|
|
outbuf =
|
|
gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
|
|
|
|
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
|
|
{
|
|
guint8 *ptr = map.data;
|
|
for (i = 0; i < num_blocks; i++) {
|
|
if (a52_block (a52dec->state)) {
|
|
/* also marks discont */
|
|
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
|
|
("error decoding block %d", i), result);
|
|
if (result != GST_FLOW_OK) {
|
|
gst_buffer_unmap (outbuf, &map);
|
|
goto exit;
|
|
}
|
|
} else {
|
|
gint n, c;
|
|
gint *reorder_map = a52dec->channel_reorder_map;
|
|
|
|
for (n = 0; n < 256; n++) {
|
|
for (c = 0; c < chans; c++) {
|
|
((sample_t *) ptr)[n * chans + reorder_map[c]] =
|
|
a52dec->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 (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
invalid_flags:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
|
|
("Invalid channel flags: %d", flags));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
|
|
{
|
|
GstA52Dec *a52dec = GST_A52DEC (bdec);
|
|
GstStructure *structure;
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
|
|
a52dec->dvdmode = TRUE;
|
|
else
|
|
a52dec->dvdmode = FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_a52dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
|
|
{
|
|
GstA52Dec *a52dec = GST_A52DEC (parent);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
gint first_access;
|
|
|
|
if (a52dec->dvdmode) {
|
|
gsize size;
|
|
guint8 data[2];
|
|
gint offset;
|
|
gint 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 = a52dec->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 = a52dec->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);
|
|
gst_buffer_unref (buf);
|
|
ret = a52dec->base_chain (pad, parent, subbuf);
|
|
}
|
|
} else {
|
|
ret = a52dec->base_chain (pad, parent, buf);
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
not_enough_data:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), 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 (a52dec), STREAM, DECODE, (NULL),
|
|
("Bad first_access parameter (%d) in buffer", first_access));
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstA52Dec *src = GST_A52DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DRC:
|
|
GST_OBJECT_LOCK (src);
|
|
src->dynamic_range_compression = g_value_get_boolean (value);
|
|
GST_OBJECT_UNLOCK (src);
|
|
break;
|
|
case ARG_MODE:
|
|
GST_OBJECT_LOCK (src);
|
|
src->request_channels &= ~A52_CHANNEL_MASK;
|
|
src->request_channels |= g_value_get_enum (value);
|
|
GST_OBJECT_UNLOCK (src);
|
|
break;
|
|
case ARG_LFE:
|
|
GST_OBJECT_LOCK (src);
|
|
src->request_channels &= ~A52_LFE;
|
|
src->request_channels |= g_value_get_boolean (value) ? A52_LFE : 0;
|
|
GST_OBJECT_UNLOCK (src);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstA52Dec *src = GST_A52DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DRC:
|
|
GST_OBJECT_LOCK (src);
|
|
g_value_set_boolean (value, src->dynamic_range_compression);
|
|
GST_OBJECT_UNLOCK (src);
|
|
break;
|
|
case ARG_MODE:
|
|
GST_OBJECT_LOCK (src);
|
|
g_value_set_enum (value, src->request_channels & A52_CHANNEL_MASK);
|
|
GST_OBJECT_UNLOCK (src);
|
|
break;
|
|
case ARG_LFE:
|
|
GST_OBJECT_LOCK (src);
|
|
g_value_set_boolean (value, src->request_channels & A52_LFE);
|
|
GST_OBJECT_UNLOCK (src);
|
|
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, "a52dec", GST_RANK_SECONDARY,
|
|
GST_TYPE_A52DEC))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
a52dec,
|
|
"Decodes ATSC A/52 encoded audio streams",
|
|
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
|