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gstreamer/ext/faad/gstfaad.c
Wim Taymans f638d690fe ext/faad/gstfaad.c: Always drain before activating the new segment. 
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_sink_event):
Always drain before activating the new segment.
2008-06-02 17:06:34 +00:00

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43 KiB
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/* GStreamer FAAD (Free AAC Decoder) plugin
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
/* These are the correct types for these functions, as defined in the source,
* with types changed to match glib types, since those are defined for us.
* However, upstream FAAD is distributed with a broken header file that defined
* these wrongly (in a way which was broken on 64 bit systems).
*
* Upstream CVS still has the bug, but has also renamed all the public symbols
* for Better Corporate Branding (or whatever), so we need to take that
* (FAAD_IS_NEAAC) into account as well.
*
* We must call them using these definitions. Most distributions now have the
* corrected header file (they distribute a patch along with the source),
* but not all, hence this Truly Evil Hack.
*
* Note: The prototypes don't need to be defined conditionaly, as the cpp will
* do that for us.
*/
#ifdef FAAD_IS_NEAAC
#define NeAACDecInit NeAACDecInit_no_definition
#define NeAACDecInit2 NeAACDecInit2_no_definition
#else
#define faacDecInit faacDecInit_no_definition
#define faacDecInit2 faacDecInit2_no_definition
#endif
#include "gstfaad.h"
#ifdef FAAD_IS_NEAAC
#undef NeAACDecInit
#undef NeAACDecInit2
#else
#undef faacDecInit
#undef faacDecInit2
#endif
extern long faacDecInit (faacDecHandle, guint8 *, guint32, guint32 *, guint8 *);
extern int8_t faacDecInit2 (faacDecHandle, guint8 *, guint32,
guint32 *, guint8 *);
GST_DEBUG_CATEGORY_STATIC (faad_debug);
#define GST_CAT_DEFAULT faad_debug
#define MAX_DECODE_ERRORS 5
static const GstElementDetails faad_details =
GST_ELEMENT_DETAILS ("AAC audio decoder",
"Codec/Decoder/Audio",
"Free MPEG-2/4 AAC decoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }")
);
#define STATIC_INT_CAPS(bpp) \
"audio/x-raw-int, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (bool) TRUE, " \
"width = (int) " G_STRINGIFY (bpp) ", " \
"depth = (int) " G_STRINGIFY (bpp) ", " \
"rate = (int) [ 8000, 96000 ], " \
"channels = (int) [ 1, 8 ]"
#if 0
#define STATIC_FLOAT_CAPS(bpp) \
"audio/x-raw-float, " \
"endianness = (int) BYTE_ORDER, " \
"depth = (int) " G_STRINGIFY (bpp) ", " \
"rate = (int) [ 8000, 96000 ], " \
"channels = (int) [ 1, 8 ]"
#endif
/*
* All except 16-bit integer are disabled until someone fixes FAAD.
* FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
* for 1 frame (1024 samples) of 6 channel (5.1) 16-bit integer 16bpp
* audio, but not for any other. You'll get random segfaults, crashes
* and even valgrind goes crazy.
*/
#define STATIC_CAPS \
STATIC_INT_CAPS (16)
#if 0
#define NOTUSED "; " \
STATIC_INT_CAPS (24) \
"; " \
STATIC_INT_CAPS (32) \
"; " \
STATIC_FLOAT_CAPS (32) \
"; " \
STATIC_FLOAT_CAPS (64)
#endif
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (STATIC_CAPS)
);
static void gst_faad_base_init (GstFaadClass * klass);
static void gst_faad_class_init (GstFaadClass * klass);
static void gst_faad_init (GstFaad * faad);
static void gst_faad_dispose (GObject * object);
static gboolean gst_faad_setcaps (GstPad * pad, GstCaps * caps);
static GstCaps *gst_faad_srcgetcaps (GstPad * pad);
static gboolean gst_faad_src_event (GstPad * pad, GstEvent * event);
static gboolean gst_faad_sink_event (GstPad * pad, GstEvent * event);
static gboolean gst_faad_src_query (GstPad * pad, GstQuery * query);
static GstFlowReturn gst_faad_chain (GstPad * pad, GstBuffer * buffer);
static GstStateChangeReturn gst_faad_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_faad_src_convert (GstFaad * faad, GstFormat src_format,
gint64 src_val, GstFormat dest_format, gint64 * dest_val);
static gboolean gst_faad_open_decoder (GstFaad * faad);
static void gst_faad_close_decoder (GstFaad * faad);
static GstElementClass *parent_class; /* NULL */
GType
gst_faad_get_type (void)
{
static GType gst_faad_type = 0;
if (!gst_faad_type) {
static const GTypeInfo gst_faad_info = {
sizeof (GstFaadClass),
(GBaseInitFunc) gst_faad_base_init,
NULL,
(GClassInitFunc) gst_faad_class_init,
NULL,
NULL,
sizeof (GstFaad),
0,
(GInstanceInitFunc) gst_faad_init,
};
gst_faad_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstFaad", &gst_faad_info, 0);
}
return gst_faad_type;
}
static void
gst_faad_base_init (GstFaadClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details (element_class, &faad_details);
GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
}
static void
gst_faad_class_init (GstFaadClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_faad_dispose);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faad_change_state);
}
static void
gst_faad_init (GstFaad * faad)
{
faad->handle = NULL;
faad->samplerate = -1;
faad->channels = -1;
faad->tempbuf = NULL;
faad->need_channel_setup = TRUE;
faad->channel_positions = NULL;
faad->init = FALSE;
faad->next_ts = 0;
faad->prev_ts = GST_CLOCK_TIME_NONE;
faad->bytes_in = 0;
faad->sum_dur_out = 0;
faad->packetised = FALSE;
faad->error_count = 0;
faad->segment = gst_segment_new ();
faad->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_element_add_pad (GST_ELEMENT (faad), faad->sinkpad);
gst_pad_set_event_function (faad->sinkpad,
GST_DEBUG_FUNCPTR (gst_faad_sink_event));
gst_pad_set_setcaps_function (faad->sinkpad,
GST_DEBUG_FUNCPTR (gst_faad_setcaps));
gst_pad_set_chain_function (faad->sinkpad,
GST_DEBUG_FUNCPTR (gst_faad_chain));
faad->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (faad->srcpad);
gst_pad_set_getcaps_function (faad->srcpad,
GST_DEBUG_FUNCPTR (gst_faad_srcgetcaps));
gst_pad_set_query_function (faad->srcpad,
GST_DEBUG_FUNCPTR (gst_faad_src_query));
gst_pad_set_event_function (faad->srcpad,
GST_DEBUG_FUNCPTR (gst_faad_src_event));
gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad);
}
static void
gst_faad_dispose (GObject * object)
{
GstFaad *faad = GST_FAAD (object);
if (faad->segment) {
gst_segment_free (faad->segment);
faad->segment = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_faad_send_tags (GstFaad * faad)
{
GstTagList *tags;
tags = gst_tag_list_new ();
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "MPEG-4 AAC audio", NULL);
gst_element_found_tags (GST_ELEMENT (faad), tags);
}
static gint
aac_rate_idx (gint rate)
{
if (92017 <= rate)
return 0;
else if (75132 <= rate)
return 1;
else if (55426 <= rate)
return 2;
else if (46009 <= rate)
return 3;
else if (37566 <= rate)
return 4;
else if (27713 <= rate)
return 5;
else if (23004 <= rate)
return 6;
else if (18783 <= rate)
return 7;
else if (13856 <= rate)
return 8;
else if (11502 <= rate)
return 9;
else if (9391 <= rate)
return 10;
else
return 11;
}
static gboolean
gst_faad_setcaps (GstPad * pad, GstCaps * caps)
{
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
GstStructure *str = gst_caps_get_structure (caps, 0);
GstBuffer *buf;
const GValue *value;
/* Assume raw stream */
faad->packetised = FALSE;
if ((value = gst_structure_get_value (str, "codec_data"))) {
guint32 samplerate;
guint8 channels;
guint8 *cdata;
guint csize;
/* We have codec data, means packetised stream */
faad->packetised = TRUE;
buf = gst_value_get_buffer (value);
cdata = GST_BUFFER_DATA (buf);
csize = GST_BUFFER_SIZE (buf);
if (csize < 2)
goto wrong_length;
/* someone forgot that char can be unsigned when writing the API */
if ((gint8) faacDecInit2 (faad->handle, cdata, csize, &samplerate,
&channels) < 0)
goto init_failed;
GST_DEBUG_OBJECT (faad, "codec_data init: channels=%u, rate=%u", channels,
samplerate);
/* not updating these here, so they are updated in the
* chain function, and new caps are created etc. */
faad->samplerate = 0;
faad->channels = 0;
faad->init = TRUE;
if (faad->tempbuf) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
} else if ((value = gst_structure_get_value (str, "framed")) &&
g_value_get_boolean (value) == TRUE) {
faad->packetised = TRUE;
GST_DEBUG_OBJECT (faad, "we have packetized audio");
} else {
faad->init = FALSE;
}
faad->fake_codec_data[0] = 0;
faad->fake_codec_data[1] = 0;
if (faad->packetised) {
gint rate, channels;
if (gst_structure_get_int (str, "rate", &rate) &&
gst_structure_get_int (str, "channels", &channels)) {
gint rate_idx, profile;
profile = 3; /* 0=MAIN, 1=LC, 2=SSR, 3=LTP */
rate_idx = aac_rate_idx (rate);
faad->fake_codec_data[0] = ((profile + 1) << 3) | ((rate_idx & 0xE) >> 1);
faad->fake_codec_data[1] = ((rate_idx & 0x1) << 7) | (channels << 3);
GST_LOG_OBJECT (faad, "created fake codec data (%u,%u): 0x%x 0x%x", rate,
channels, (int) faad->fake_codec_data[0],
(int) faad->fake_codec_data[1]);
}
}
faad->need_channel_setup = TRUE;
if (!faad->packetised)
gst_faad_send_tags (faad);
return TRUE;
/* ERRORS */
wrong_length:
{
GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long");
return FALSE;
}
init_failed:
{
GST_DEBUG_OBJECT (faad, "faacDecInit2() failed");
return FALSE;
}
}
/*
* Channel identifier conversion - caller g_free()s result!
*/
/*
static guchar *
gst_faad_chanpos_from_gst (GstAudioChannelPosition * pos, guint num)
{
guchar *fpos = g_new (guchar, num);
guint n;
for (n = 0; n < num; n++) {
switch (pos[n]) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
fpos[n] = FRONT_CHANNEL_LEFT;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
fpos[n] = FRONT_CHANNEL_RIGHT;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
case GST_AUDIO_CHANNEL_POSITION_FRONT_MONO:
fpos[n] = FRONT_CHANNEL_CENTER;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT:
fpos[n] = SIDE_CHANNEL_LEFT;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT:
fpos[n] = SIDE_CHANNEL_RIGHT;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT:
fpos[n] = BACK_CHANNEL_LEFT;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT:
fpos[n] = BACK_CHANNEL_RIGHT;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER:
fpos[n] = BACK_CHANNEL_CENTER;
break;
case GST_AUDIO_CHANNEL_POSITION_LFE:
fpos[n] = LFE_CHANNEL;
break;
default:
GST_WARNING ("Unsupported GST channel position 0x%x encountered",
pos[n]);
g_free (fpos);
return NULL;
}
}
return fpos;
}
*/
static GstAudioChannelPosition *
gst_faad_chanpos_to_gst (GstFaad * faad, guchar * fpos, guint num,
gboolean * channel_map_failed)
{
GstAudioChannelPosition *pos;
guint n;
gboolean unknown_channel = FALSE;
*channel_map_failed = FALSE;
/* special handling for the common cases for mono and stereo */
if (num == 1 && fpos[0] == FRONT_CHANNEL_CENTER) {
GST_DEBUG_OBJECT (faad, "mono common case; won't set channel positions");
return NULL;
} else if (num == 2 && fpos[0] == FRONT_CHANNEL_LEFT
&& fpos[1] == FRONT_CHANNEL_RIGHT) {
GST_DEBUG_OBJECT (faad, "stereo common case; won't set channel positions");
return NULL;
}
pos = g_new (GstAudioChannelPosition, num);
for (n = 0; n < num; n++) {
GST_DEBUG_OBJECT (faad, "faad channel %d as %d", n, fpos[n]);
switch (fpos[n]) {
case FRONT_CHANNEL_LEFT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
break;
case FRONT_CHANNEL_RIGHT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case FRONT_CHANNEL_CENTER:
/* argh, mono = center */
if (num == 1)
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
else
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
break;
case SIDE_CHANNEL_LEFT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
break;
case SIDE_CHANNEL_RIGHT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
break;
case BACK_CHANNEL_LEFT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
break;
case BACK_CHANNEL_RIGHT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
case BACK_CHANNEL_CENTER:
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
break;
case LFE_CHANNEL:
pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE;
break;
default:
GST_DEBUG_OBJECT (faad, "unknown channel %d at %d", fpos[n], n);
unknown_channel = TRUE;
break;
}
}
if (unknown_channel) {
g_free (pos);
pos = NULL;
switch (num) {
case 1:{
GST_DEBUG_OBJECT (faad,
"FAAD reports unknown 1 channel mapping. Forcing to mono");
break;
}
case 2:{
GST_DEBUG_OBJECT (faad,
"FAAD reports unknown 2 channel mapping. Forcing to stereo");
break;
}
default:{
GST_WARNING ("Unsupported FAAD channel position 0x%x encountered",
fpos[n]);
*channel_map_failed = TRUE;
break;
}
}
}
return pos;
}
static GstCaps *
gst_faad_srcgetcaps (GstPad * pad)
{
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
static GstAudioChannelPosition *supported_positions = NULL;
static gint num_supported_positions = LFE_CHANNEL - FRONT_CHANNEL_CENTER + 1;
GstCaps *templ;
if (!supported_positions) {
guchar *supported_fpos = g_new0 (guchar, num_supported_positions);
gint n;
gboolean map_failed;
for (n = 0; n < num_supported_positions; n++) {
supported_fpos[n] = n + FRONT_CHANNEL_CENTER;
}
supported_positions = gst_faad_chanpos_to_gst (faad, supported_fpos,
num_supported_positions, &map_failed);
g_free (supported_fpos);
}
if (faad->handle != NULL && faad->channels != -1 && faad->samplerate != -1) {
GstCaps *caps = gst_caps_new_empty ();
GstStructure *str;
gint fmt[] = {
FAAD_FMT_16BIT,
#if 0
FAAD_FMT_24BIT,
FAAD_FMT_32BIT,
FAAD_FMT_FLOAT,
FAAD_FMT_DOUBLE,
#endif
-1
}
, n;
for (n = 0; fmt[n] != -1; n++) {
switch (fmt[n]) {
case FAAD_FMT_16BIT:
str = gst_structure_new ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
break;
#if 0
case FAAD_FMT_24BIT:
str = gst_structure_new ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 24, "depth", G_TYPE_INT, 24, NULL);
break;
case FAAD_FMT_32BIT:
str = gst_structure_new ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 32, "depth", G_TYPE_INT, 32, NULL);
break;
case FAAD_FMT_FLOAT:
str = gst_structure_new ("audio/x-raw-float",
"depth", G_TYPE_INT, 32, NULL);
break;
case FAAD_FMT_DOUBLE:
str = gst_structure_new ("audio/x-raw-float",
"depth", G_TYPE_INT, 64, NULL);
break;
#endif
default:
str = NULL;
break;
}
if (!str)
continue;
if (faad->samplerate > 0) {
gst_structure_set (str, "rate", G_TYPE_INT, faad->samplerate, NULL);
} else {
gst_structure_set (str, "rate", GST_TYPE_INT_RANGE, 8000, 96000, NULL);
}
if (faad->channels > 0) {
gst_structure_set (str, "channels", G_TYPE_INT, faad->channels, NULL);
/* put channel information here */
if (faad->channel_positions) {
GstAudioChannelPosition *pos;
gboolean map_failed;
pos = gst_faad_chanpos_to_gst (faad, faad->channel_positions,
faad->channels, &map_failed);
if (map_failed) {
gst_structure_free (str);
continue;
}
if (pos) {
gst_audio_set_channel_positions (str, pos);
g_free (pos);
}
} else {
gst_audio_set_structure_channel_positions_list (str,
supported_positions, num_supported_positions);
}
} else {
gst_structure_set (str, "channels", GST_TYPE_INT_RANGE, 1, 8, NULL);
/* we set channel positions later */
}
gst_structure_set (str, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
gst_caps_append_structure (caps, str);
}
if (faad->channels == -1) {
gst_audio_set_caps_channel_positions_list (caps,
supported_positions, num_supported_positions);
}
gst_object_unref (faad);
return caps;
}
/* template with channel positions */
templ = gst_caps_copy (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
gst_audio_set_caps_channel_positions_list (templ,
supported_positions, num_supported_positions);
gst_object_unref (faad);
return templ;
}
/*
static GstPadLinkReturn
gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
{
GstStructure *structure;
const gchar *mimetype;
gint fmt = -1;
gint depth, rate, channels;
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
structure = gst_caps_get_structure (caps, 0);
if (!faad->handle || (faad->samplerate == -1 || faad->channels == -1) ||
!faad->channel_positions) {
return GST_PAD_LINK_DELAYED;
}
mimetype = gst_structure_get_name (structure);
// Samplerate and channels are normally provided through
// * the getcaps function
if (!gst_structure_get_int (structure, "channels", &channels) ||
!gst_structure_get_int (structure, "rate", &rate) ||
rate != faad->samplerate || channels != faad->channels) {
return GST_PAD_LINK_REFUSED;
}
// Another internal checkup.
if (faad->need_channel_setup) {
GstAudioChannelPosition *pos;
guchar *fpos;
guint i;
pos = gst_audio_get_channel_positions (structure);
if (!pos) {
return GST_PAD_LINK_DELAYED;
}
fpos = gst_faad_chanpos_from_gst (pos, faad->channels);
g_free (pos);
if (!fpos)
return GST_PAD_LINK_REFUSED;
for (i = 0; i < faad->channels; i++) {
if (fpos[i] != faad->channel_positions[i]) {
g_free (fpos);
return GST_PAD_LINK_REFUSED;
}
}
g_free (fpos);
}
if (!strcmp (mimetype, "audio/x-raw-int")) {
gint width;
if (!gst_structure_get_int (structure, "depth", &depth) ||
!gst_structure_get_int (structure, "width", &width))
return GST_PAD_LINK_REFUSED;
if (depth != width)
return GST_PAD_LINK_REFUSED;
switch (depth) {
case 16:
fmt = FAAD_FMT_16BIT;
break;
#if 0
case 24:
fmt = FAAD_FMT_24BIT;
break;
case 32:
fmt = FAAD_FMT_32BIT;
break;
#endif
}
} else {
if (!gst_structure_get_int (structure, "depth", &depth))
return GST_PAD_LINK_REFUSED;
switch (depth) {
#if 0
case 32:
fmt = FAAD_FMT_FLOAT;
break;
case 64:
fmt = FAAD_FMT_DOUBLE;
break;
#endif
}
}
if (fmt != -1) {
faacDecConfiguration *conf;
conf = faacDecGetCurrentConfiguration (faad->handle);
conf->outputFormat = fmt;
if (faacDecSetConfiguration (faad->handle, conf) == 0)
return GST_PAD_LINK_REFUSED;
// FIXME: handle return value, how?
faad->bps = depth / 8;
return GST_PAD_LINK_OK;
}
return GST_PAD_LINK_REFUSED;
}*/
static void
clear_queued (GstFaad * faad)
{
g_list_foreach (faad->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (faad->queued);
faad->queued = NULL;
}
static GstFlowReturn
flush_queued (GstFaad * faad)
{
GstFlowReturn ret = GST_FLOW_OK;
while (faad->queued) {
GstBuffer *buf = GST_BUFFER_CAST (faad->queued->data);
GST_LOG_OBJECT (faad, "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 (faad->srcpad, buf);
faad->queued = g_list_delete_link (faad->queued, faad->queued);
}
return ret;
}
static GstFlowReturn
gst_faad_drain (GstFaad * faad)
{
GstFlowReturn ret = GST_FLOW_OK;
if (faad->segment->rate < 0.0) {
/* if we have some queued frames for reverse playback, flush
* them now */
ret = flush_queued (faad);
}
return ret;
}
static gboolean
gst_faad_do_raw_seek (GstFaad * faad, GstEvent * event)
{
GstSeekFlags flags;
GstSeekType start_type, end_type;
GstFormat format;
gdouble rate;
gint64 start, start_time;
gst_event_parse_seek (event, &rate, &format, &flags, &start_type,
&start_time, &end_type, NULL);
if (rate != 1.0 ||
format != GST_FORMAT_TIME ||
start_type != GST_SEEK_TYPE_SET || end_type != GST_SEEK_TYPE_NONE) {
return FALSE;
}
if (!gst_faad_src_convert (faad, GST_FORMAT_TIME, start_time,
GST_FORMAT_BYTES, &start)) {
return FALSE;
}
event = gst_event_new_seek (1.0, GST_FORMAT_BYTES, flags,
GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE, -1);
GST_DEBUG_OBJECT (faad, "seeking to %" GST_TIME_FORMAT " at byte offset %"
G_GINT64_FORMAT, GST_TIME_ARGS (start_time), start);
return gst_pad_push_event (faad->sinkpad, event);
}
static gboolean
gst_faad_src_event (GstPad * pad, GstEvent * event)
{
GstFaad *faad;
gboolean res;
faad = GST_FAAD (gst_pad_get_parent (pad));
GST_LOG_OBJECT (faad, "Handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:{
/* try upstream first, there might be a demuxer */
gst_event_ref (event);
if (!(res = gst_pad_push_event (faad->sinkpad, event))) {
res = gst_faad_do_raw_seek (faad, event);
}
gst_event_unref (event);
break;
}
default:
res = gst_pad_push_event (faad->sinkpad, event);
break;
}
gst_object_unref (faad);
return res;
}
static gboolean
gst_faad_sink_event (GstPad * pad, GstEvent * event)
{
GstFaad *faad;
gboolean res = TRUE;
faad = GST_FAAD (gst_pad_get_parent (pad));
GST_LOG_OBJECT (faad, "Handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
if (faad->tempbuf != NULL) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
clear_queued (faad);
res = gst_pad_push_event (faad->srcpad, event);
break;
case GST_EVENT_EOS:
gst_faad_drain (faad);
if (faad->tempbuf != NULL) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
res = gst_pad_push_event (faad->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
GstFormat fmt;
gboolean is_update;
gint64 start, end, base;
gdouble rate;
gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
&end, &base);
/* drain queued buffers before we activate the new segment */
gst_faad_drain (faad);
if (fmt == GST_FORMAT_TIME) {
GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
GST_TIME_ARGS (end));
gst_segment_set_newsegment (faad->segment, is_update, rate, fmt, start,
end, base);
} else if (fmt == GST_FORMAT_BYTES) {
gint64 new_start = 0;
gint64 new_end = -1;
GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_BYTES (%"
G_GUINT64_FORMAT " - %" G_GUINT64_FORMAT ")", start, end);
if (gst_faad_src_convert (faad, GST_FORMAT_BYTES, start,
GST_FORMAT_TIME, &new_start)) {
if (end != -1) {
gst_faad_src_convert (faad, GST_FORMAT_BYTES, end,
GST_FORMAT_TIME, &new_end);
}
} else {
GST_DEBUG
("no average bitrate yet, sending newsegment with start at 0");
}
gst_event_unref (event);
event = gst_event_new_new_segment (is_update, rate,
GST_FORMAT_TIME, new_start, new_end, new_start);
gst_segment_set_newsegment (faad->segment, is_update, rate,
GST_FORMAT_TIME, new_start, new_end, new_start);
GST_DEBUG ("Sending new NEWSEGMENT event, time %" GST_TIME_FORMAT
" - %" GST_TIME_FORMAT, GST_TIME_ARGS (new_start),
GST_TIME_ARGS (new_end));
faad->next_ts = new_start;
faad->prev_ts = GST_CLOCK_TIME_NONE;
}
res = gst_pad_push_event (faad->srcpad, event);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (faad);
return res;
}
static gboolean
gst_faad_src_convert (GstFaad * faad, GstFormat src_format, gint64 src_val,
GstFormat dest_format, gint64 * dest_val)
{
guint64 bytes_in, time_out, val;
if (src_format == dest_format) {
if (dest_val)
*dest_val = src_val;
return TRUE;
}
GST_OBJECT_LOCK (faad);
bytes_in = faad->bytes_in;
time_out = faad->sum_dur_out;
GST_OBJECT_UNLOCK (faad);
if (bytes_in == 0 || time_out == 0)
return FALSE;
/* convert based on the average bitrate so far */
if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME) {
val = gst_util_uint64_scale (src_val, time_out, bytes_in);
} else if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES) {
val = gst_util_uint64_scale (src_val, bytes_in, time_out);
} else {
return FALSE;
}
if (dest_val)
*dest_val = (gint64) val;
return TRUE;
}
static gboolean
gst_faad_src_query (GstPad * pad, GstQuery * query)
{
gboolean res = FALSE;
GstFaad *faad;
GstPad *peer = NULL;
faad = GST_FAAD (gst_pad_get_parent (pad));
GST_LOG_OBJECT (faad, "processing %s query", GST_QUERY_TYPE_NAME (query));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:{
GstFormat format;
gint64 len_bytes, duration;
/* try upstream first, in case there's a demuxer */
if ((res = gst_pad_query_default (pad, query)))
break;
gst_query_parse_duration (query, &format, NULL);
if (format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (faad, "query failed: can't handle format %s",
gst_format_get_name (format));
break;
}
peer = gst_pad_get_peer (faad->sinkpad);
if (peer == NULL)
break;
format = GST_FORMAT_BYTES;
if (!gst_pad_query_duration (peer, &format, &len_bytes)) {
GST_DEBUG_OBJECT (faad, "query failed: failed to get upstream length");
break;
}
res = gst_faad_src_convert (faad, GST_FORMAT_BYTES, len_bytes,
GST_FORMAT_TIME, &duration);
if (res) {
gst_query_set_duration (query, GST_FORMAT_TIME, duration);
GST_LOG_OBJECT (faad, "duration estimate: %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
}
break;
}
case GST_QUERY_POSITION:{
GstFormat format;
gint64 pos_bytes, pos;
/* try upstream first, in case there's a demuxer */
if ((res = gst_pad_query_default (pad, query)))
break;
gst_query_parse_position (query, &format, NULL);
if (format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (faad, "query failed: can't handle format %s",
gst_format_get_name (format));
break;
}
peer = gst_pad_get_peer (faad->sinkpad);
if (peer == NULL)
break;
format = GST_FORMAT_BYTES;
if (!gst_pad_query_position (peer, &format, &pos_bytes)) {
GST_OBJECT_LOCK (faad);
pos = faad->next_ts;
GST_OBJECT_UNLOCK (faad);
res = TRUE;
} else {
res = gst_faad_src_convert (faad, GST_FORMAT_BYTES, pos_bytes,
GST_FORMAT_TIME, &pos);
}
if (res) {
gst_query_set_position (query, GST_FORMAT_TIME, pos);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
if (peer)
gst_object_unref (peer);
gst_object_unref (faad);
return res;
}
static gboolean
gst_faad_update_caps (GstFaad * faad, faacDecFrameInfo * info)
{
GstAudioChannelPosition *pos;
gboolean ret;
gboolean channel_map_failed;
GstCaps *caps;
/* store new negotiation information */
faad->samplerate = info->samplerate;
faad->channels = info->channels;
g_free (faad->channel_positions);
faad->channel_positions = g_memdup (info->channel_position, faad->channels);
caps = gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, faad->samplerate,
"channels", G_TYPE_INT, faad->channels, NULL);
faad->bps = 16 / 8;
channel_map_failed = FALSE;
pos =
gst_faad_chanpos_to_gst (faad, faad->channel_positions, faad->channels,
&channel_map_failed);
if (channel_map_failed) {
GST_DEBUG_OBJECT (faad, "Could not map channel positions");
gst_caps_unref (caps);
return FALSE;
}
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
}
GST_DEBUG_OBJECT (faad, "New output caps: %" GST_PTR_FORMAT, caps);
ret = gst_pad_set_caps (faad->srcpad, caps);
gst_caps_unref (caps);
return ret;
}
/*
* Find syncpoint in ADTS/ADIF stream. Doesn't work for raw,
* packetized streams. Be careful when calling.
* Returns FALSE on no-sync, fills offset/length if one/two
* syncpoints are found, only returns TRUE when it finds two
* subsequent syncpoints (similar to mp3 typefinding in
* gst/typefind/) for ADTS because 12 bits isn't very reliable.
*/
static gboolean
gst_faad_sync (GstBuffer * buf, guint * off)
{
guint8 *data = GST_BUFFER_DATA (buf);
guint size = GST_BUFFER_SIZE (buf), n;
gint snc;
GST_DEBUG ("Finding syncpoint");
/* check for too small a buffer */
if (size < 3)
return FALSE;
/* FIXME: for no-sync, we go over the same data for every new buffer.
* We should save the information somewhere. */
for (n = 0; n < size - 3; n++) {
snc = GST_READ_UINT16_BE (&data[n]);
if ((snc & 0xfff6) == 0xfff0) {
/* we have an ADTS syncpoint. Parse length and find
* next syncpoint. */
guint len;
GST_DEBUG ("Found one ADTS syncpoint at offset 0x%x, tracing next...", n);
if (size - n < 5) {
GST_DEBUG ("Not enough data to parse ADTS header");
return FALSE;
}
*off = n;
len = ((data[n + 3] & 0x03) << 11) |
(data[n + 4] << 3) | ((data[n + 5] & 0xe0) >> 5);
if (n + len + 2 >= size) {
GST_DEBUG ("Next frame is not within reach");
return FALSE;
}
snc = GST_READ_UINT16_BE (&data[n + len]);
if ((snc & 0xfff6) == 0xfff0) {
GST_DEBUG ("Found ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
return TRUE;
}
GST_DEBUG ("No next frame found... (should be at 0x%x)", n + len);
} else if (!memcmp (&data[n], "ADIF", 4)) {
/* we have an ADIF syncpoint. 4 bytes is enough. */
*off = n;
GST_DEBUG ("Found ADIF syncpoint at offset 0x%x", n);
return TRUE;
}
}
GST_DEBUG ("Found no syncpoint");
return FALSE;
}
static gboolean
looks_like_valid_header (guint8 * input_data, guint input_size)
{
if (input_size < 4)
return FALSE;
if (input_data[0] == 'A'
&& input_data[1] == 'D' && input_data[2] == 'I' && input_data[3] == 'F')
/* ADIF type header */
return TRUE;
if (input_data[0] == 0xff && (input_data[1] >> 4) == 0xf)
/* ADTS type header */
return TRUE;
return FALSE;
}
static GstFlowReturn
gst_faad_chain (GstPad * pad, GstBuffer * buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
guint input_size;
guint skip_bytes = 0;
guchar *input_data;
GstFaad *faad;
GstBuffer *outbuf;
faacDecFrameInfo info;
void *out;
gboolean run_loop = TRUE;
guint sync_off;
faad = GST_FAAD (gst_pad_get_parent (pad));
if (GST_BUFFER_IS_DISCONT (buffer)) {
gst_faad_drain (faad);
faacDecPostSeekReset (faad->handle, 0);
if (faad->tempbuf != NULL) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
faad->discont = TRUE;
}
GST_OBJECT_LOCK (faad);
faad->bytes_in += GST_BUFFER_SIZE (buffer);
GST_OBJECT_UNLOCK (faad);
if (GST_BUFFER_TIMESTAMP (buffer) != GST_CLOCK_TIME_NONE) {
/* some demuxers send multiple buffers in a row
* with the same timestamp (e.g. matroskademux) */
if (GST_BUFFER_TIMESTAMP (buffer) != faad->prev_ts) {
faad->next_ts = GST_BUFFER_TIMESTAMP (buffer);
faad->prev_ts = GST_BUFFER_TIMESTAMP (buffer);
}
GST_LOG_OBJECT (faad, "Timestamp on incoming buffer: %" GST_TIME_FORMAT
", next_ts: %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (faad->next_ts));
}
/* buffer + remaining data */
if (faad->tempbuf) {
buffer = gst_buffer_join (faad->tempbuf, buffer);
faad->tempbuf = NULL;
}
input_data = GST_BUFFER_DATA (buffer);
input_size = GST_BUFFER_SIZE (buffer);
if (!faad->packetised) {
if (!gst_faad_sync (buffer, &sync_off)) {
goto next;
} else {
input_data += sync_off;
input_size -= sync_off;
}
}
/* init if not already done during capsnego */
if (!faad->init) {
guint32 rate;
guint8 ch;
GST_DEBUG_OBJECT (faad, "initialising ...");
/* We check if the first data looks like it might plausibly contain
* appropriate initialisation info... if not, we use our fake_codec_data
*/
if (looks_like_valid_header (input_data, input_size) || !faad->packetised) {
if (faacDecInit (faad->handle, input_data, input_size, &rate, &ch) < 0)
goto init_failed;
GST_DEBUG_OBJECT (faad, "faacDecInit() ok: rate=%u,channels=%u", rate,
ch);
} else {
if ((gint8) faacDecInit2 (faad->handle, faad->fake_codec_data, 2,
&rate, &ch) < 0) {
goto init2_failed;
}
GST_DEBUG_OBJECT (faad, "faacDecInit2() ok: rate=%u,channels=%u", rate,
ch);
}
skip_bytes = 0;
faad->init = TRUE;
/* make sure we create new caps below */
faad->samplerate = 0;
faad->channels = 0;
gst_faad_send_tags (faad);
}
/* decode cycle */
info.bytesconsumed = input_size - skip_bytes;
info.error = 0;
if (!faad->packetised) {
/* We must check that ourselves for raw stream */
run_loop = (input_size >= FAAD_MIN_STREAMSIZE);
}
while ((input_size > 0) && run_loop) {
if (faad->packetised) {
/* Only one packet per buffer, no matter how much is really consumed */
run_loop = FALSE;
} else {
if (input_size < FAAD_MIN_STREAMSIZE || info.bytesconsumed <= 0) {
break;
}
}
out = faacDecDecode (faad->handle, &info, input_data + skip_bytes,
input_size - skip_bytes);
if (info.error > 0) {
GST_WARNING_OBJECT (faad, "decoding error: %s",
faacDecGetErrorMessage (info.error));
/* mark discont for the next buffer */
faad->discont = TRUE;
goto out;
}
if (info.bytesconsumed > input_size)
info.bytesconsumed = input_size;
input_size -= info.bytesconsumed;
input_data += info.bytesconsumed;
if (out && info.samples > 0) {
gboolean fmt_change = FALSE;
/* see if we need to renegotiate */
if (info.samplerate != faad->samplerate ||
info.channels != faad->channels || !faad->channel_positions) {
fmt_change = TRUE;
} else {
gint i;
for (i = 0; i < info.channels; i++) {
if (info.channel_position[i] != faad->channel_positions[i])
fmt_change = TRUE;
}
}
if (fmt_change) {
if (!gst_faad_update_caps (faad, &info)) {
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL),
("Setting caps on source pad failed"));
ret = GST_FLOW_ERROR;
goto out;
}
}
if (info.samples > G_MAXUINT / faad->bps) {
/* C's lovely propensity for int overflow.. */
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Output buffer too large"));
ret = GST_FLOW_ERROR;
goto out;
}
/* play decoded data */
if (info.samples > 0) {
guint bufsize = info.samples * faad->bps;
guint num_samples = info.samples / faad->channels;
GST_DEBUG_OBJECT (faad, "decoded %d samples", (guint) info.samples);
/* note: info.samples is total samples, not per channel */
ret =
gst_pad_alloc_buffer_and_set_caps (faad->srcpad, 0, bufsize,
GST_PAD_CAPS (faad->srcpad), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
GST_BUFFER_OFFSET (outbuf) =
GST_CLOCK_TIME_TO_FRAMES (faad->next_ts, faad->samplerate);
GST_BUFFER_TIMESTAMP (outbuf) = faad->next_ts;
GST_BUFFER_DURATION (outbuf) =
GST_FRAMES_TO_CLOCK_TIME (num_samples, faad->samplerate);
GST_OBJECT_LOCK (faad);
faad->next_ts += GST_BUFFER_DURATION (outbuf);
faad->sum_dur_out += GST_BUFFER_DURATION (outbuf);
GST_OBJECT_UNLOCK (faad);
if ((outbuf = gst_audio_buffer_clip (outbuf, faad->segment,
faad->samplerate, faad->bps * faad->channels))) {
GST_LOG_OBJECT (faad,
"pushing buffer, off=%" G_GUINT64_FORMAT ", ts=%" GST_TIME_FORMAT,
GST_BUFFER_OFFSET (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
if (faad->discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
faad->discont = FALSE;
}
if (faad->segment->rate > 0.0) {
ret = gst_pad_push (faad->srcpad, outbuf);
} else {
/* reverse playback, queue frame till later when we get a discont. */
GST_DEBUG_OBJECT (faad, "queued frame");
faad->queued = g_list_prepend (faad->queued, outbuf);
ret = GST_FLOW_OK;
}
if (ret != GST_FLOW_OK)
goto out;
}
}
}
}
next:
/* Keep the leftovers in raw stream */
if (input_size > 0 && !faad->packetised) {
if (input_size < GST_BUFFER_SIZE (buffer)) {
faad->tempbuf = gst_buffer_create_sub (buffer,
GST_BUFFER_SIZE (buffer) - input_size, input_size);
} else {
faad->tempbuf = buffer;
gst_buffer_ref (buffer);
}
}
out:
gst_buffer_unref (buffer);
gst_object_unref (faad);
return ret;
/* ERRORS */
init_failed:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to init decoder from stream"));
ret = GST_FLOW_ERROR;
goto out;
}
init2_failed:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("%s() failed", (faad->handle) ? "faacDecInit2" : "faacDecOpen"));
ret = GST_FLOW_ERROR;
goto out;
}
}
static gboolean
gst_faad_open_decoder (GstFaad * faad)
{
faacDecConfiguration *conf;
faad->handle = faacDecOpen ();
if (faad->handle == NULL) {
GST_WARNING_OBJECT (faad, "faacDecOpen() failed");
return FALSE;
}
conf = faacDecGetCurrentConfiguration (faad->handle);
conf->defObjectType = LC;
conf->dontUpSampleImplicitSBR = 1;
conf->outputFormat = FAAD_FMT_16BIT;
if (faacDecSetConfiguration (faad->handle, conf) == 0) {
GST_WARNING_OBJECT (faad, "faacDecSetConfiguration() failed");
return FALSE;
}
return TRUE;
}
static void
gst_faad_close_decoder (GstFaad * faad)
{
if (faad->handle) {
faacDecClose (faad->handle);
faad->handle = NULL;
}
}
static GstStateChangeReturn
gst_faad_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstFaad *faad = GST_FAAD (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_faad_open_decoder (faad))
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (faad->segment, GST_FORMAT_TIME);
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
faad->samplerate = -1;
faad->channels = -1;
faad->need_channel_setup = TRUE;
faad->init = FALSE;
g_free (faad->channel_positions);
faad->channel_positions = NULL;
faad->next_ts = 0;
faad->prev_ts = GST_CLOCK_TIME_NONE;
faad->bytes_in = 0;
faad->sum_dur_out = 0;
faad->error_count = 0;
if (faad->tempbuf) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
clear_queued (faad);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_faad_close_decoder (faad);
break;
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"faad",
"Free AAC Decoder (FAAD)",
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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