/* GStreamer
* Copyright (C) <2001> David I. Lehn <dlehn@users.sourceforge.net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-a52dec
*
* Dolby Digital (AC-3) audio decoder.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioconvert ! audioresample ! autoaudiosink
* ]| Play audio part of a dvd title.
* |[
* gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
* ]| Decode and play a stand alone AC-3 file.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#ifdef HAVE_STDINT_H
#include <stdint.h>
#endif
#include <gst/gst.h>
#include <a52dec/a52.h>
#if !defined(A52_ACCEL_DETECT)
# include <a52dec/mm_accel.h>
#endif
#include "gsta52dec.h"
#if HAVE_ORC
#include <orc/orc.h>
#endif
#ifdef LIBA52_DOUBLE
#define SAMPLE_WIDTH 64
#define SAMPLE_FORMAT GST_AUDIO_NE(F64)
#define SAMPLE_TYPE GST_AUDIO_FORMAT_F64
#else
#define SAMPLE_WIDTH 32
#define SAMPLE_FORMAT GST_AUDIO_NE(F32)
#define SAMPLE_TYPE GST_AUDIO_FORMAT_F32
#endif
GST_DEBUG_CATEGORY_STATIC (a52dec_debug);
#define GST_CAT_DEFAULT (a52dec_debug)
/* A52Dec args */
enum
{
ARG_0,
ARG_DRC,
ARG_MODE,
ARG_LFE,
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-ac3; audio/ac3; audio/x-private1-ac3")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " SAMPLE_FORMAT ", "
"layout = (string) interleaved, "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
#define gst_a52dec_parent_class parent_class
G_DEFINE_TYPE (GstA52Dec, gst_a52dec, GST_TYPE_AUDIO_DECODER);
static gboolean gst_a52dec_start (GstAudioDecoder * dec);
static gboolean gst_a52dec_stop (GstAudioDecoder * dec);
static gboolean gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
static GstFlowReturn gst_a52dec_parse (GstAudioDecoder * dec,
GstAdapter * adapter, gint * offset, gint * length);
static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
static void gst_a52dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_a52dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
#define GST_TYPE_A52DEC_MODE (gst_a52dec_mode_get_type())
static GType
gst_a52dec_mode_get_type (void)
{
static GType a52dec_mode_type = 0;
static const GEnumValue a52dec_modes[] = {
{A52_MONO, "Mono", "mono"},
{A52_STEREO, "Stereo", "stereo"},
{A52_3F, "3 Front", "3f"},
{A52_2F1R, "2 Front, 1 Rear", "2f1r"},
{A52_3F1R, "3 Front, 1 Rear", "3f1r"},
{A52_2F2R, "2 Front, 2 Rear", "2f2r"},
{A52_3F2R, "3 Front, 2 Rear", "3f2r"},
{A52_DOLBY, "Dolby", "dolby"},
{0, NULL, NULL},
};
if (!a52dec_mode_type) {
a52dec_mode_type = g_enum_register_static ("GstA52DecMode", a52dec_modes);
}
return a52dec_mode_type;
}
static void
gst_a52dec_class_init (GstA52DecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioDecoderClass *gstbase_class;
guint cpuflags = 0;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbase_class = (GstAudioDecoderClass *) klass;
gobject_class->set_property = gst_a52dec_set_property;
gobject_class->get_property = gst_a52dec_get_property;
gstbase_class->start = GST_DEBUG_FUNCPTR (gst_a52dec_start);
gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_a52dec_stop);
gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_a52dec_set_format);
gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_a52dec_parse);
gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_a52dec_handle_frame);
/**
* GstA52Dec::drc
*
* Set to true to apply the recommended Dolby Digital dynamic range compression
* to the audio stream. Dynamic range compression makes loud sounds
* softer and soft sounds louder, so you can more easily listen
* to the stream without disturbing other people.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
g_param_spec_boolean ("drc", "Dynamic Range Compression",
"Use Dynamic Range Compression", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstA52Dec::mode
*
* Force a particular output channel configuration from the decoder. By default,
* the channel downmix (if any) is chosen automatically based on the downstream
* capabilities of the pipeline.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)",
GST_TYPE_A52DEC_MODE, A52_3F2R,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstA52Dec::lfe
*
* Whether to output the LFE (Low Frequency Emitter) channel of the audio stream.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LFE,
g_param_spec_boolean ("lfe", "LFE", "LFE", TRUE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
gst_element_class_set_static_metadata (gstelement_class,
"ATSC A/52 audio decoder", "Codec/Decoder/Audio",
"Decodes ATSC A/52 encoded audio streams",
"David I. Lehn <dlehn@users.sourceforge.net>");
GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0,
"AC3/A52 software decoder");
/* If no CPU instruction based acceleration is available, end up using the
* generic software djbfft based one when available in the used liba52 */
#ifdef MM_ACCEL_DJBFFT
klass->a52_cpuflags = MM_ACCEL_DJBFFT;
#elif defined(A52_ACCEL_DETECT)
klass->a52_cpuflags = A52_ACCEL_DETECT;
#else
klass->a52_cpuflags = 0;
#endif
#if HAVE_ORC && !defined(A52_ACCEL_DETECT)
cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx"));
if (cpuflags & ORC_TARGET_MMX_MMX)
klass->a52_cpuflags |= MM_ACCEL_X86_MMX;
if (cpuflags & ORC_TARGET_MMX_3DNOW)
klass->a52_cpuflags |= MM_ACCEL_X86_3DNOW;
if (cpuflags & ORC_TARGET_MMX_MMXEXT)
klass->a52_cpuflags |= MM_ACCEL_X86_MMXEXT;
#endif
GST_LOG ("CPU flags: a52=%08x, orc=%08x", klass->a52_cpuflags, cpuflags);
}
static void
gst_a52dec_init (GstA52Dec * a52dec)
{
a52dec->request_channels = A52_CHANNEL;
a52dec->dynamic_range_compression = FALSE;
a52dec->state = NULL;
a52dec->samples = NULL;
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(a52dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (a52dec));
/* retrieve and intercept base class chain.
* Quite HACKish, but that's dvd specs/caps for you,
* since one buffer needs to be split into 2 frames */
a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
GST_DEBUG_FUNCPTR (gst_a52dec_chain));
}
static gboolean
gst_a52dec_start (GstAudioDecoder * dec)
{
GstA52Dec *a52dec = GST_A52DEC (dec);
GstA52DecClass *klass;
static GMutex init_mutex;
GST_DEBUG_OBJECT (dec, "start");
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
g_mutex_lock (&init_mutex);
#if defined(A52_ACCEL_DETECT)
a52dec->state = a52_init ();
/* This line is just to avoid being accused of not using klass */
a52_accel (klass->a52_cpuflags & A52_ACCEL_DETECT);
#else
a52dec->state = a52_init (klass->a52_cpuflags);
#endif
g_mutex_unlock (&init_mutex);
if (!a52dec->state) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL),
("failed to initialize a52 state"));
return FALSE;
}
a52dec->samples = a52_samples (a52dec->state);
a52dec->bit_rate = -1;
a52dec->sample_rate = -1;
a52dec->stream_channels = A52_CHANNEL;
a52dec->using_channels = A52_CHANNEL;
a52dec->level = 1;
a52dec->bias = 0;
a52dec->flag_update = TRUE;
/* call upon legacy upstream byte support (e.g. seeking) */
gst_audio_decoder_set_estimate_rate (dec, TRUE);
return TRUE;
}
static gboolean
gst_a52dec_stop (GstAudioDecoder * dec)
{
GstA52Dec *a52dec = GST_A52DEC (dec);
GST_DEBUG_OBJECT (dec, "stop");
a52dec->samples = NULL;
if (a52dec->state) {
a52_free (a52dec->state);
a52dec->state = NULL;
}
return TRUE;
}
static GstFlowReturn
gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
gint * _offset, gint * len)
{
GstA52Dec *a52dec;
const guint8 *data;
gint av, size;
gint length = 0, flags, sample_rate, bit_rate;
GstFlowReturn result = GST_FLOW_EOS;
a52dec = GST_A52DEC (bdec);
size = av = gst_adapter_available (adapter);
data = (const guint8 *) gst_adapter_map (adapter, av);
/* find and read header */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
while (size >= 7) {
length = a52_syncinfo ((guint8 *) data, &flags, &sample_rate, &bit_rate);
if (length == 0) {
/* shift window to re-find sync */
data++;
size--;
} else if (length <= size) {
GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
result = GST_FLOW_OK;
break;
} else {
GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
length, size);
break;
}
}
gst_adapter_unmap (adapter);
*_offset = av - size;
*len = length;
return result;
}
static gint
gst_a52dec_channels (int flags, GstAudioChannelPosition * pos)
{
gint chans = 0;
if (flags & A52_LFE) {
chans += 1;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE1;
}
}
flags &= A52_CHANNEL_MASK;
switch (flags) {
case A52_3F2R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
chans += 5;
break;
case A52_2F2R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
chans += 4;
break;
case A52_3F1R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
chans += 4;
break;
case A52_2F1R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
chans += 3;
break;
case A52_3F:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
chans += 3;
break;
case A52_CHANNEL: /* Dual mono. Should really be handled as 2 src pads */
case A52_STEREO:
case A52_DOLBY:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
chans += 2;
break;
case A52_MONO:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_MONO;
}
chans += 1;
break;
default:
/* error, caller should post error message */
return 0;
}
return chans;
}
static gboolean
gst_a52dec_reneg (GstA52Dec * a52dec)
{
gint channels;
gboolean result = FALSE;
GstAudioChannelPosition from[6], to[6];
GstAudioInfo info;
channels = gst_a52dec_channels (a52dec->using_channels, from);
if (!channels)
goto done;
GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d",
channels, a52dec->sample_rate);
memcpy (to, from, sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (to, channels);
gst_audio_get_channel_reorder_map (channels, from, to,
a52dec->channel_reorder_map);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
SAMPLE_TYPE, a52dec->sample_rate, channels, (channels > 1 ? to : NULL));
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (a52dec), &info))
goto done;
result = TRUE;
done:
return result;
}
static void
gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
{
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);
gst_buffer_unref (outbuf);
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);