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gstreamer/ext/faad/gstfaad.c
Michael Smith f910e0f4a5 ext/faad/gstfaad.c: Fix #334748: use fake_codec_data if the first bytes of the first buffer we process doesn't look l... 
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_setcaps),
(looks_like_valid_header), (gst_faad_chain):
Fix #334748: use fake_codec_data if the first bytes of the first
buffer we process doesn't look like plausible AAC data (e.g.
reserved values for rate, or channels). Fixes playback of Apple's
movie trailers.
2006-05-03 10:25:46 +00:00

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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're screwed there.
*
* 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. This hack will need updating if
* upstream ever releases something with the new API.
*/
#define faacDecInit faacDecInit_no_definition
#define faacDecInit2 faacDecInit2_no_definition
#include "gstfaad.h"
#undef faacDecInit
#undef faacDecInit2
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
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 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)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
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->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_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;
/* We have codec data, means packetised stream */
faad->packetised = TRUE;
buf = GST_BUFFER (gst_value_get_mini_object (value));
/* someone forgot that char can be unsigned when writing the API */
if ((gint8) faacDecInit2 (faad->handle,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), &samplerate,
&channels) < 0) {
GST_DEBUG ("faacDecInit2() failed");
return FALSE;
}
GST_DEBUG_OBJECT (faad, "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;
} 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;
}
/*
* 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 (guchar * fpos, guint num)
{
GstAudioChannelPosition *pos = g_new (GstAudioChannelPosition, num);
guint n;
gboolean unknown_channel = FALSE;
for (n = 0; n < num; 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:
unknown_channel = TRUE;
break;
}
}
if (unknown_channel) {
switch (num) {
case 1:{
GST_DEBUG ("FAAD reports unknown 1 channel mapping. Forcing to mono");
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
break;
}
case 2:{
GST_DEBUG ("FAAD reports unknown 2 channel mapping. Forcing to stereo");
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
}
default:{
GST_WARNING ("Unsupported FAAD channel position 0x%x encountered",
fpos[n]);
g_free (pos);
pos = NULL;
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;
for (n = 0; n < num_supported_positions; n++) {
supported_fpos[n] = n + FRONT_CHANNEL_CENTER;
}
supported_positions = gst_faad_chanpos_to_gst (supported_fpos,
num_supported_positions);
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;
pos = gst_faad_chanpos_to_gst (faad->channel_positions,
faad->channels);
if (!pos) {
gst_structure_free (str);
continue;
}
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 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_send_event (GST_PAD_PEER (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_event_default (pad, event))) {
res = gst_faad_do_raw_seek (faad, event);
}
gst_event_unref (event);
break;
}
default:
res = gst_pad_event_default (pad, 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));
/* FIXME: we should probably handle FLUSH */
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
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);
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));
} 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_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,
GstCaps ** p_caps)
{
GstAudioChannelPosition *pos;
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;
pos = gst_faad_chanpos_to_gst (faad->channel_positions, faad->channels);
if (!pos) {
GST_DEBUG_OBJECT (faad, "Could not map channel positions");
gst_caps_unref (caps);
return FALSE;
}
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
GST_DEBUG ("New output caps: %" GST_PTR_FORMAT, caps);
if (!gst_pad_set_caps (faad->srcpad, caps)) {
gst_caps_unref (caps);
return FALSE;
}
*p_caps = caps;
return TRUE;
}
/*
* 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");
/* 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)
{
guint32 rate;
guint32 channels;
if (input_size < 2)
return FALSE;
rate = ((input_data[0] & 0x7) << 1) | ((input_data[1] & 0x80) >> 7);
channels = (input_data[1] & 0x78) >> 3;
if (rate == 0xd || rate == 0xe) /* Reserved values */
return FALSE;
if (channels == 0) /* Extended specifier: never seen one of these */
return FALSE;
return TRUE;
}
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;
GstCaps *caps = NULL;
faacDecFrameInfo info;
void *out;
gboolean run_loop = TRUE;
guint sync_off;
faad = GST_FAAD (gst_pad_get_parent (pad));
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) {
guint32 rate;
guint8 ch;
if (!faad->packetised)
goto decode_error;
/* decode error? try again using faacDecInit2
* fabricated private codec data from sink caps */
gst_faad_close_decoder (faad);
if (!gst_faad_open_decoder (faad))
goto init2_failed;
GST_DEBUG_OBJECT (faad, "decoding error, reopening with faacDecInit2()");
if ((gint8) faacDecInit2 (faad->handle, faad->fake_codec_data, 2,
&rate, &ch) < 0) {
goto init2_failed;
}
GST_DEBUG_OBJECT (faad, "faacDecInit2(): rate=%d,channels=%d", rate, ch);
/* let's try again */
info.error = 0;
out = faacDecDecode (faad->handle, &info, input_data + skip_bytes,
input_size - skip_bytes);
if (info.error)
goto decode_error;
}
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, &caps)) {
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL),
("Setting caps on source pad failed"));
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;
/* note: info.samples is total samples, not per channel */
ret =
gst_pad_alloc_buffer_and_set_caps (faad->srcpad, 0, bufsize, caps,
&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);
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 ((ret = gst_pad_push (faad->srcpad, outbuf)) != GST_FLOW_OK &&
ret != GST_FLOW_NOT_LINKED)
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:
if (caps)
gst_caps_unref (caps);
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;
}
decode_error:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to decode buffer: %s", faacDecGetErrorMessage (info.error)));
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)
{
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;
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;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_faad_close_decoder (faad);
if (faad->tempbuf) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
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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