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gstreamer/ext/faad/gstfaad.c

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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.
*/
/**
* SECTION:element-faad
* @seealso: faac
*
* faad decodes AAC (MPEG-4 part 3) stream.
*
* <refsect2>
* <title>Example launch lines</title>
* |[
* gst-launch filesrc location=example.mp4 ! qtdemux ! faad ! audioconvert ! audioresample ! autoaudiosink
* ]| Play aac from mp4 file.
* |[
* gst-launch filesrc location=example.adts ! faad ! audioconvert ! audioresample ! autoaudiosink
* ]| Play standalone aac bitstream.
* </refsect2>
*/
#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.
*/
#if FAAD2_MINOR_VERSION < 7
#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
#endif /* FAAD2_MINOR_VERSION < 7 */
#include "gstfaad.h"
#if FAAD2_MINOR_VERSION < 7
#ifdef FAAD_IS_NEAAC
#undef NeAACDecInit
#undef NeAACDecInit2
#else
#undef faacDecInit
#undef faacDecInit2
#endif
extern long faacDecInit (faacDecHandle, guint8 *, guint32, guint32 *, guint8 *);
extern gint8 faacDecInit2 (faacDecHandle, guint8 *, guint32,
guint32 *, guint8 *);
#endif /* FAAD2_MINOR_VERSION < 7 */
GST_DEBUG_CATEGORY_STATIC (faad_debug);
#define GST_CAT_DEFAULT faad_debug
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_reset (GstFaad * faad);
static gboolean gst_faad_start (GstAudioDecoder * dec);
static gboolean gst_faad_stop (GstAudioDecoder * dec);
static gboolean gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps);
static gboolean gst_faad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length);
static GstFlowReturn gst_faad_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static void gst_faad_flush (GstAudioDecoder * dec, gboolean hard);
static gboolean gst_faad_open_decoder (GstFaad * faad);
static void gst_faad_close_decoder (GstFaad * faad);
GST_BOILERPLATE (GstFaad, gst_faad, GstAudioDecoder, GST_TYPE_AUDIO_DECODER);
static void
gst_faad_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_add_static_pad_template (element_class,
&sink_template);
gst_element_class_set_details_simple (element_class, "AAC audio decoder",
"Codec/Decoder/Audio",
"Free MPEG-2/4 AAC decoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
}
static void
gst_faad_class_init (GstFaadClass * klass)
{
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
base_class->start = GST_DEBUG_FUNCPTR (gst_faad_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_faad_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_faad_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (gst_faad_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faad_handle_frame);
base_class->flush = GST_DEBUG_FUNCPTR (gst_faad_flush);
}
static void
gst_faad_init (GstFaad * faad, GstFaadClass * klass)
{
gst_faad_reset (faad);
}
static void
gst_faad_reset_stream_state (GstFaad * faad)
{
if (faad->handle)
faacDecPostSeekReset (faad->handle, 0);
}
static void
gst_faad_reset (GstFaad * faad)
{
faad->samplerate = -1;
faad->channels = -1;
faad->init = FALSE;
faad->packetised = FALSE;
g_free (faad->channel_positions);
faad->channel_positions = NULL;
faad->last_header = 0;
gst_faad_reset_stream_state (faad);
}
static gboolean
gst_faad_start (GstAudioDecoder * dec)
{
GstFaad *faad = GST_FAAD (dec);
GST_DEBUG_OBJECT (dec, "start");
gst_faad_reset (faad);
/* call upon legacy upstream byte support (e.g. seeking) */
gst_audio_decoder_set_byte_time (dec, TRUE);
/* never mind a few errors */
gst_audio_decoder_set_max_errors (dec, 10);
return TRUE;
}
static gboolean
gst_faad_stop (GstAudioDecoder * dec)
{
GstFaad *faad = GST_FAAD (dec);
GST_DEBUG_OBJECT (dec, "stop");
gst_faad_reset (faad);
gst_faad_close_decoder (faad);
return TRUE;
}
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_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstFaad *faad = GST_FAAD (dec);
GstStructure *str = gst_caps_get_structure (caps, 0);
GstBuffer *buf;
const GValue *value;
/* clean up current decoder, rather than trying to reconfigure */
gst_faad_close_decoder (faad);
/* Assume raw stream */
faad->packetised = FALSE;
if ((value = gst_structure_get_value (str, "codec_data"))) {
#if FAAD2_MINOR_VERSION >= 7
unsigned long samplerate;
#else
guint32 samplerate;
#endif
guint8 channels;
guint8 *cdata;
guint csize;
/* We have codec data, means packetised stream */
faad->packetised = TRUE;
buf = gst_value_get_buffer (value);
g_return_val_if_fail (buf != NULL, FALSE);
cdata = GST_BUFFER_DATA (buf);
csize = GST_BUFFER_SIZE (buf);
if (csize < 2)
goto wrong_length;
GST_DEBUG_OBJECT (faad,
"codec_data: object_type=%d, sample_rate=%d, channels=%d",
((cdata[0] & 0xf8) >> 3),
(((cdata[0] & 0x07) << 1) | ((cdata[1] & 0x80) >> 7)),
((cdata[1] & 0x78) >> 3));
if (!gst_faad_open_decoder (faad))
goto open_failed;
/* someone forgot that char can be unsigned when writing the API */
if ((gint8) faacDecInit2 (faad->handle, cdata, csize, &samplerate,
&channels) < 0)
goto init_failed;
if (channels != ((cdata[1] & 0x78) >> 3)) {
/* https://bugs.launchpad.net/ubuntu/+source/faad2/+bug/290259 */
GST_WARNING_OBJECT (faad,
"buggy faad version, wrong nr of channels %d instead of %d", channels,
((cdata[1] & 0x78) >> 3));
}
GST_DEBUG_OBJECT (faad, "codec_data init: channels=%u, rate=%u", channels,
(guint32) 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;
} else if ((value = gst_structure_get_value (str, "framed")) &&
g_value_get_boolean (value) == TRUE) {
faad->packetised = TRUE;
faad->init = FALSE;
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 && !faad->init) {
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]);
}
}
return TRUE;
/* ERRORS */
wrong_length:
{
GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long");
gst_object_unref (faad);
return FALSE;
}
open_failed:
{
GST_DEBUG_OBJECT (faad, "failed to create decoder");
gst_object_unref (faad);
return FALSE;
}
init_failed:
{
GST_DEBUG_OBJECT (faad, "faacDecInit2() failed");
gst_object_unref (faad);
return FALSE;
}
}
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_OBJECT (faad,
"Unsupported FAAD channel position 0x%x encountered", fpos[n]);
*channel_map_failed = TRUE;
break;
}
}
}
return pos;
}
static gboolean
gst_faad_update_caps (GstFaad * faad, faacDecFrameInfo * info)
{
GstAudioChannelPosition *pos;
gboolean ret;
gboolean channel_map_failed;
GstCaps *caps;
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 (G_LIKELY (!fmt_change))
return TRUE;
/* 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 (GST_AUDIO_DECODER_SRC_PAD (faad), 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 (GstFaad * faad, const guint8 * data, guint size, gboolean next,
gint * off, gint * length)
{
guint n = 0;
gint snc;
gboolean ret = FALSE;
guint len = 0;
GST_LOG_OBJECT (faad, "Finding syncpoint");
/* check for too small a buffer */
if (size < 3)
goto exit;
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. */
GST_LOG_OBJECT (faad,
"Found one ADTS syncpoint at offset 0x%x, tracing next...", n);
if (size - n < 5) {
GST_LOG_OBJECT (faad, "Not enough data to parse ADTS header");
break;
}
len = ((data[n + 3] & 0x03) << 11) |
(data[n + 4] << 3) | ((data[n + 5] & 0xe0) >> 5);
if (n + len + 2 >= size) {
GST_LOG_OBJECT (faad, "Frame size %d, next frame is not within reach",
len);
if (next) {
break;
} else if (n + len <= size) {
GST_LOG_OBJECT (faad, "but have complete frame and no next frame; "
"accept ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
ret = TRUE;
break;
}
}
snc = GST_READ_UINT16_BE (&data[n + len]);
if ((snc & 0xfff6) == 0xfff0) {
GST_LOG_OBJECT (faad,
"Found ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
ret = TRUE;
break;
}
GST_LOG_OBJECT (faad, "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. */
GST_LOG_OBJECT (faad, "Found ADIF syncpoint at offset 0x%x", n);
ret = TRUE;
break;
}
}
exit:
*off = n;
if (ret) {
*length = len;
} else {
GST_LOG_OBJECT (faad, "Found no syncpoint");
}
return ret;
}
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_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{
GstFaad *faad;
const guint8 *data;
guint size;
gboolean sync, eos;
faad = GST_FAAD (dec);
size = gst_adapter_available (adapter);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
gst_audio_decoder_get_parse_state (dec, &sync, &eos);
if (faad->packetised) {
*offset = 0;
*length = size;
return GST_FLOW_OK;
} else {
data = gst_adapter_peek (adapter, size);
return gst_faad_sync (faad, data, size, !eos, offset, length) ?
GST_FLOW_OK : GST_FLOW_UNEXPECTED;
}
}
static GstFlowReturn
gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstFaad *faad;
GstFlowReturn ret = GST_FLOW_OK;
guint input_size;
guchar *input_data;
GstBuffer *outbuf;
faacDecFrameInfo info;
void *out;
faad = GST_FAAD (dec);
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
input_data = GST_BUFFER_DATA (buffer);
input_size = GST_BUFFER_SIZE (buffer);
init:
/* init if not already done during capsnego */
if (!faad->init) {
#if FAAD2_MINOR_VERSION >= 7
unsigned long rate;
#else
guint32 rate;
#endif
guint8 ch;
GST_DEBUG_OBJECT (faad, "initialising ...");
if (!gst_faad_open_decoder (faad))
goto open_failed;
/* 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",
(guint32) 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",
(guint32) rate, ch);
}
faad->init = TRUE;
/* make sure we create new caps below */
faad->samplerate = 0;
faad->channels = 0;
}
/* decode cycle */
info.error = 0;
do {
if (!faad->packetised) {
/* faad only really parses ADTS header at Init time, not when decoding,
* so monitor for changes and kick faad when needed */
if (GST_READ_UINT32_BE (input_data) >> 4 != faad->last_header >> 4) {
GST_DEBUG_OBJECT (faad, "ADTS header changed, forcing Init");
faad->last_header = GST_READ_UINT32_BE (input_data);
/* kick hard */
gst_faad_close_decoder (faad);
faad->init = FALSE;
goto init;
}
}
out = faacDecDecode (faad->handle, &info, input_data, input_size);
if (info.error > 0) {
/* give up on frame and bail out */
gst_audio_decoder_finish_frame (dec, NULL, 1);
goto decode_failed;
}
GST_LOG_OBJECT (faad, "%d bytes consumed, %d samples decoded",
(guint) info.bytesconsumed, (guint) info.samples);
if (out && info.samples > 0) {
if (!gst_faad_update_caps (faad, &info))
goto negotiation_failed;
/* C's lovely propensity for int overflow.. */
if (info.samples > G_MAXUINT / faad->bps)
goto sample_overflow;
/* note: info.samples is total samples, not per channel */
ret =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD
(faad), 0, info.samples * faad->bps,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (faad)), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
}
} while (FALSE);
out:
return ret;
/* ERRORS */
open_failed:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to open decoder"));
ret = GST_FLOW_ERROR;
goto out;
}
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;
}
decode_failed:
{
GST_AUDIO_DECODER_ERROR (faad, 1, STREAM, DECODE, (NULL),
("decoding error: %s", faacDecGetErrorMessage (info.error)), ret);
goto out;
}
negotiation_failed:
{
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL),
("Setting caps on source pad failed"));
ret = GST_FLOW_ERROR;
goto out;
}
sample_overflow:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Output buffer too large"));
ret = GST_FLOW_ERROR;
goto out;
}
}
static void
gst_faad_flush (GstAudioDecoder * dec, gboolean hard)
{
gst_faad_reset_stream_state (GST_FAAD (dec));
}
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 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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