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gstreamer/ext/faad/gstfaad.c
Michael Smith e765addd73 Fix compile on systems with broken faad headers. 
Original commit message from CVS:
Fix compile on systems with broken faad headers.
2005-11-11 18:54:14 +00:00

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/* GStreamer FAAD (Free AAC Decoder) plugin
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.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 faadDecInit_no_definition
#define faacDecInit2 faadDecInit2_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 GstElementDetails faad_details = {
"Free AAC Decoder (FAAD)",
"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_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_faad_chain (GstPad * pad, GstBuffer * buffer);
static GstStateChangeReturn gst_faad_change_state (GstElement * element,
GstStateChange transition);
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_template (gst_static_pad_template_get (&sink_template),
"sink");
gst_element_add_pad (GST_ELEMENT (faad), faad->sinkpad);
gst_pad_set_event_function (faad->sinkpad,
GST_DEBUG_FUNCPTR (gst_faad_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_template (gst_static_pad_template_get (&src_template),
"src");
gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad);
gst_pad_use_fixed_caps (faad->srcpad);
gst_pad_set_getcaps_function (faad->srcpad,
GST_DEBUG_FUNCPTR (gst_faad_srcgetcaps));
}
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"))) {
guint samplerate;
guchar 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;
}
#if 0
faad->samplerate = samplerate;
faad->channels = channels;
#endif
/* 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 {
faad->init = FALSE;
}
faad->need_channel_setup = TRUE;
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;
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:
GST_WARNING ("Unsupported FAAD channel position 0x%x encountered",
fpos[n]);
g_free (pos);
return NULL;
}
}
return pos;
}
/*
static GstPadLinkReturn
gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
{
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
GstStructure *str = gst_caps_get_structure (caps, 0);
const GValue *value;
GstBuffer *buf;
// Assume raw stream
faad->packetised = FALSE;
if ((value = gst_structure_get_value (str, "codec_data"))) {
gulong samplerate;
guchar channels;
// We have codec data, means packetised stream
faad->packetised = TRUE;
buf = g_value_get_boxed (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)
return GST_PAD_LINK_REFUSED;
//faad->samplerate = samplerate;
//faad->channels = channels;
faad->init = TRUE;
if (faad->tempbuf) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
} else {
faad->init = FALSE;
}
faad->need_channel_setup = TRUE;
// if there's no decoderspecificdata, it's all fine. We cannot know
// * much more at this point...
return GST_PAD_LINK_OK;
}
*/
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 != -1) {
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 != -1) {
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_event (GstPad * pad, GstEvent * event)
{
GstFaad *faad;
gboolean res = TRUE;
faad = GST_FAAD (gst_pad_get_parent (pad));
GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
/* FIXME: we should probably handle FLUSH and also
* SEEK in the case where we are not in a container
* (when our newsegment was in BYTES) */
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
GST_STREAM_LOCK (pad);
if (faad->tempbuf != NULL) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
res = gst_pad_push_event (faad->srcpad, event);
GST_STREAM_UNLOCK (pad);
break;
case GST_EVENT_NEWSEGMENT:
{
GstFormat fmt;
gboolean is_update;
gint64 start, end, base;
gdouble rate;
gst_event_parse_newsegment (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) {
GstEvent *new_ev;
guint64 new_start = 0;
guint64 new_end = GST_CLOCK_TIME_NONE;
GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_BYTES (%"
G_GUINT64_FORMAT " - %" G_GUINT64_FORMAT ")", start, end);
if (faad->bytes_in > 0 && faad->sum_dur_out > 0) {
/* try to convert based on the average bitrate so far */
new_start = (faad->sum_dur_out * start) / faad->bytes_in;
if (new_end != (guint64) - 1) {
new_end = (faad->sum_dur_out * end) / faad->bytes_in;
}
} else {
GST_DEBUG
("no average bitrate yet, sending newsegment with start at 0");
}
new_ev =
gst_event_new_newsegment (is_update, rate, GST_FORMAT_TIME,
new_start, new_end, base);
gst_event_unref (event);
event = new_ev;
GST_DEBUG ("Sending new NEWSEGMENT event, time %" GST_TIME_FORMAT
" - %" GST_TIME_FORMAT, GST_TIME_ARGS (new_start),
GST_TIME_ARGS (new_end));
}
GST_STREAM_LOCK (pad);
res = gst_pad_push_event (faad->srcpad, event);
GST_STREAM_UNLOCK (pad);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
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);
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;
}
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;
faad = GST_FAAD (gst_pad_get_parent (pad));
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_DEBUG ("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;
}
/* init if not already done during capsnego */
if (!faad->init) {
guint32 samplerate;
guchar channels;
glong init_res;
init_res = faacDecInit (faad->handle,
GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), &samplerate,
&channels);
if (init_res < 0) {
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to init decoder from stream"));
gst_object_unref (faad);
return GST_FLOW_UNEXPECTED;
}
skip_bytes = init_res;
faad->init = TRUE;
/* make sure we create new caps below */
faad->samplerate = 0;
faad->channels = 0;
}
/* decode cycle */
input_data = GST_BUFFER_DATA (buffer);
input_size = GST_BUFFER_SIZE (buffer);
info.bytesconsumed = input_size - skip_bytes;
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) {
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to decode buffer: %s", faacDecGetErrorMessage (info.error)));
ret = GST_FLOW_ERROR;
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, &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 && GST_PAD_PEER (faad->srcpad)) {
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 (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);
faad->next_ts += GST_BUFFER_DURATION (outbuf);
faad->sum_dur_out += GST_BUFFER_DURATION (outbuf);
GST_DEBUG ("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;
}
}
}
/* 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);
}
}
faad->bytes_in += input_size;
out:
if (caps)
gst_caps_unref (caps);
gst_buffer_unref (buffer);
gst_object_unref (faad);
return ret;
}
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 (!(faad->handle = faacDecOpen ()))
return GST_STATE_CHANGE_FAILURE;
else {
faacDecConfiguration *conf;
conf = faacDecGetCurrentConfiguration (faad->handle);
conf->defObjectType = LC;
/* conf->dontUpSampleImplicitSBR = 1; */
conf->outputFormat = FAAD_FMT_16BIT;
if (faacDecSetConfiguration (faad->handle, conf) == 0)
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;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
faacDecClose (faad->handle);
faad->handle = NULL;
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, GST_ORIGIN)
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