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gstreamer/ext/a52dec/gsta52dec.c
Sebastian Dröge 220b88fcc1 a52dec: Don't claim to support upstream renegotiation 
and use fixed caps on the srcpad. To correctly support
upstream renegotiation a52dec would need to check if the
caps of the downstream allocated buffer are the requested
caps or if the size is different.

Fixes bug #665989.
2011-12-13 14:54:18 +01:00

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/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-a52dec
*
* Dolby Digital (AC-3) audio decoder.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioresample ! audioconvert ! alsasink
* ]| Play audio track from a dvd.
* |[
* gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink
* ]| Decode a stand alone file and play it.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include "_stdint.h"
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <a52dec/a52.h>
#include <a52dec/mm_accel.h>
#include "gsta52dec.h"
#if HAVE_ORC
#include <orc/orc.h>
#endif
#ifdef LIBA52_DOUBLE
#define SAMPLE_WIDTH 64
#else
#define SAMPLE_WIDTH 32
#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-float, "
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
"width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstElement, GST_TYPE_ELEMENT);
static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buffer);
static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf);
static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_a52dec_change_state (GstElement * element,
GstStateChange transition);
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_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_static_pad_template (element_class, &sink_factory);
gst_element_class_add_static_pad_template (element_class, &src_factory);
gst_element_class_set_details_simple (element_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");
}
static void
gst_a52dec_class_init (GstA52DecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
guint cpuflags;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_a52dec_set_property;
gobject_class->get_property = gst_a52dec_get_property;
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state);
/**
* 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));
/* 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;
#else
klass->a52_cpuflags = 0;
#endif
#if HAVE_ORC
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;
#else
cpuflags = 0;
#endif
GST_LOG ("CPU flags: a52=%08x, liboil=%08x", klass->a52_cpuflags, cpuflags);
}
static void
gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class)
{
/* create the sink and src pads */
a52dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (a52dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_a52dec_sink_setcaps));
gst_pad_set_chain_function (a52dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_a52dec_chain));
gst_pad_set_event_function (a52dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_a52dec_sink_event));
gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
a52dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_pad_use_fixed_caps (a52dec->srcpad);
gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
a52dec->request_channels = A52_CHANNEL;
a52dec->dynamic_range_compression = FALSE;
a52dec->state = NULL;
a52dec->samples = NULL;
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
}
static gint
gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos)
{
gint chans = 0;
GstAudioChannelPosition *pos = NULL;
/* allocated just for safety. Number makes no sense */
if (_pos) {
pos = g_new (GstAudioChannelPosition, 6);
*_pos = pos;
}
if (flags & A52_LFE) {
chans += 1;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE;
}
}
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_FRONT_MONO;
}
chans += 1;
break;
default:
/* error, caller should post error message */
g_free (pos);
return 0;
}
return chans;
}
static void
clear_queued (GstA52Dec * dec)
{
g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->queued);
dec->queued = NULL;
}
static GstFlowReturn
flush_queued (GstA52Dec * dec)
{
GstFlowReturn ret = GST_FLOW_OK;
while (dec->queued) {
GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* iterate ouput queue an push downstream */
ret = gst_pad_push (dec->srcpad, buf);
dec->queued = g_list_delete_link (dec->queued, dec->queued);
}
return ret;
}
static GstFlowReturn
gst_a52dec_drain (GstA52Dec * dec)
{
GstFlowReturn ret = GST_FLOW_OK;
if (dec->segment.rate < 0.0) {
/* if we have some queued frames for reverse playback, flush
* them now */
ret = flush_queued (dec);
}
return ret;
}
static GstFlowReturn
gst_a52dec_push (GstA52Dec * a52dec,
GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
{
GstBuffer *buf;
int chans, n, c;
GstFlowReturn result;
flags &= (A52_CHANNEL_MASK | A52_LFE);
chans = gst_a52dec_channels (flags, NULL);
if (!chans) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
("invalid channel flags: %d", flags));
return GST_FLOW_ERROR;
}
result =
gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
if (result != GST_FLOW_OK)
return result;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
samples[c * 256 + n];
}
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
result = GST_FLOW_OK;
if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment,
a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
/* set discont when needed */
if (a52dec->discont) {
GST_LOG_OBJECT (a52dec, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
a52dec->discont = FALSE;
}
if (a52dec->segment.rate > 0.0) {
GST_DEBUG_OBJECT (a52dec,
"Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
result = gst_pad_push (srcpad, buf);
} else {
/* reverse playback, queue frame till later when we get a discont. */
GST_DEBUG_OBJECT (a52dec, "queued frame");
a52dec->queued = g_list_prepend (a52dec->queued, buf);
}
}
return result;
}
static gboolean
gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
{
GstAudioChannelPosition *pos;
gint channels = gst_a52dec_channels (a52dec->using_channels, &pos);
GstCaps *caps = NULL;
gboolean result = FALSE;
if (!channels)
goto done;
GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d",
channels, a52dec->sample_rate);
caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, SAMPLE_WIDTH,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, a52dec->sample_rate, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
if (!gst_pad_set_caps (pad, caps))
goto done;
result = TRUE;
done:
if (caps)
gst_caps_unref (caps);
return result;
}
static gboolean
gst_a52dec_sink_event (GstPad * pad, GstEvent * event)
{
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
gboolean ret = FALSE;
GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat fmt;
gboolean update;
gint64 start, end, pos;
gdouble rate, arate;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &fmt,
&start, &end, &pos);
/* drain queued buffers before activating the segment so that we can clip
* against the old segment first */
gst_a52dec_drain (a52dec);
if (fmt != 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 (fmt));
gst_event_unref (event);
a52dec->sent_segment = FALSE;
/* set some dummy values, FIXME: do proper conversion */
a52dec->time = start = pos = 0;
fmt = GST_FORMAT_TIME;
end = -1;
} else {
a52dec->time = start;
a52dec->sent_segment = TRUE;
GST_DEBUG_OBJECT (a52dec,
"Pushing newseg rate %g, applied rate %g, format %d, start %"
G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT ", pos %"
G_GINT64_FORMAT, rate, arate, fmt, start, end, pos);
ret = gst_pad_push_event (a52dec->srcpad, event);
}
gst_segment_set_newsegment (&a52dec->segment, update, rate, fmt, start,
end, pos);
break;
}
case GST_EVENT_TAG:
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
case GST_EVENT_EOS:
gst_a52dec_drain (a52dec);
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
case GST_EVENT_FLUSH_START:
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
clear_queued (a52dec);
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
default:
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
}
gst_object_unref (a52dec);
return ret;
}
static void
gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
{
GstTagList *taglist;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_AUDIO_CODEC, "Dolby Digital (AC-3)",
GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
GST_PAD (a52dec->srcpad), taglist);
}
static GstFlowReturn
gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
{
gint channels, i;
gboolean need_reneg = FALSE;
/* update stream information, renegotiate or re-streaminfo if needed */
need_reneg = FALSE;
if (a52dec->sample_rate != sample_rate) {
need_reneg = TRUE;
a52dec->sample_rate = sample_rate;
}
if (flags) {
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 (a52dec->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 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",
flags ? gst_a52dec_channels (flags, NULL) : 6);
if (gst_structure_get_int (structure, "channels", &channels)
&& channels <= 6)
flags = a52_channels[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, data, &flags, &a52dec->level, a52dec->bias)) {
GST_WARNING ("a52_frame error");
a52dec->discont = TRUE;
return GST_FLOW_OK;
}
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 ("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, a52dec->srcpad)) {
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
}
if (a52dec->dynamic_range_compression == FALSE) {
a52_dynrng (a52dec->state, NULL, NULL);
}
/* each frame consists of 6 blocks */
for (i = 0; i < 6; i++) {
if (a52_block (a52dec->state)) {
/* ignore errors but mark a discont */
GST_WARNING ("a52_block error %d", i);
a52dec->discont = TRUE;
} else {
GstFlowReturn ret;
/* push on */
ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
a52dec->samples, a52dec->time);
if (ret != GST_FLOW_OK)
return ret;
}
a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
}
return GST_FLOW_OK;
}
static gboolean
gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
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;
gst_object_unref (a52dec);
return TRUE;
}
static GstFlowReturn
gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
{
GstA52Dec *a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
GstFlowReturn ret;
gint first_access;
if (GST_BUFFER_IS_DISCONT (buf)) {
GST_LOG_OBJECT (a52dec, "received DISCONT");
gst_a52dec_drain (a52dec);
/* clear cache on discont and mark a discont in the element */
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
a52dec->discont = TRUE;
}
if (a52dec->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guchar *data = GST_BUFFER_DATA (buf);
gint offset;
gint 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_a52dec_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_a52dec_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_a52dec_chain_raw (pad, subbuf);
}
} else {
gst_buffer_ref (buf);
ret = gst_a52dec_chain_raw (pad, buf);
}
done:
gst_buffer_unref (buf);
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 GstFlowReturn
gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
{
GstA52Dec *a52dec;
guint8 *data;
guint size;
gint length = 0, flags, sample_rate, bit_rate;
GstFlowReturn result = GST_FLOW_OK;
a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
if (!a52dec->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 (a52dec->srcpad, gst_event_new_new_segment (FALSE,
segment.rate, segment.format, segment.start,
segment.duration, segment.start));
a52dec->sent_segment = TRUE;
}
/* merge with cache, if any. Also make sure timestamps match */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
a52dec->time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (a52dec,
"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 (a52dec->cache) {
buf = gst_buffer_join (a52dec->cache, buf);
a52dec->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* find and read header */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
while (size >= 7) {
length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
if (length == 0) {
/* no sync */
data++;
size--;
} else if (length <= size) {
GST_DEBUG ("Sync: %d", length);
if (flags != a52dec->prev_flags)
a52dec->flag_update = TRUE;
a52dec->prev_flags = flags;
result = gst_a52dec_handle_frame (a52dec, data,
length, flags, sample_rate, bit_rate);
if (result != GST_FLOW_OK) {
size = 0;
break;
}
size -= length;
data += length;
} else {
/* not enough data */
GST_LOG ("Not enough data available");
break;
}
}
/* keep cache */
if (length == 0) {
GST_LOG ("No sync found");
}
if (size > 0) {
a52dec->cache = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - size, size);
}
gst_buffer_unref (buf);
return result;
}
static GstStateChangeReturn
gst_a52dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstA52Dec *a52dec = GST_A52DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:{
GstA52DecClass *klass;
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
a52dec->state = a52_init (klass->a52_cpuflags);
if (!a52dec->state) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
("Failed to initialize a52 state"));
ret = GST_STATE_CHANGE_FAILURE;
}
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
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->time = 0;
a52dec->sent_segment = FALSE;
a52dec->flag_update = TRUE;
gst_segment_init (&a52dec->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:
a52dec->samples = NULL;
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
clear_queued (a52dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (a52dec->state) {
a52_free (a52dec->state);
a52dec->state = NULL;
}
break;
default:
break;
}
return ret;
}
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
/* ensure GstAudioChannelPosition type is registered */
if (!gst_audio_channel_position_get_type ())
return FALSE;
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);
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