mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-15 21:06:32 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/ext/faad/gstfaad.c
Mark Nauwelaerts 8a6c490962 faad: reflow opening and closing decoder 
In particular, this reconfigures quite forcefully when renegotiation is needed.

Fixes #631501.
2010-10-14 11:27:17 +02:00

1422 lines
39 KiB
C
Raw Blame History

/* 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_base_init (GstFaadClass * klass);
static void gst_faad_class_init (GstFaadClass * klass);
static void gst_faad_init (GstFaad * faad);
static void gst_faad_reset (GstFaad * faad);
static void gst_faad_finalize (GObject * object);
static void clear_queued (GstFaad * faad);
static gboolean gst_faad_setcaps (GstPad * pad, GstCaps * caps);
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_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)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_faad_finalize);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faad_change_state);
}
static void
gst_faad_init (GstFaad * faad)
{
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_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);
faad->adapter = gst_adapter_new ();
gst_faad_reset (faad);
}
static void
gst_faad_reset_stream_state (GstFaad * faad)
{
faad->sync_flush = 0;
gst_adapter_clear (faad->adapter);
clear_queued (faad);
if (faad->handle)
faacDecPostSeekReset (faad->handle, 0);
}
static void
gst_faad_reset (GstFaad * faad)
{
gst_segment_init (&faad->segment, GST_FORMAT_TIME);
faad->samplerate = -1;
faad->channels = -1;
faad->init = FALSE;
faad->packetised = FALSE;
g_free (faad->channel_positions);
faad->channel_positions = NULL;
faad->next_ts = GST_CLOCK_TIME_NONE;
faad->prev_ts = 0;
faad->bytes_in = 0;
faad->sum_dur_out = 0;
faad->error_count = 0;
faad->last_header = 0;
gst_faad_reset_stream_state (faad);
}
static void
gst_faad_finalize (GObject * object)
{
GstFaad *faad = GST_FAAD (object);
g_object_unref (faad->adapter);
G_OBJECT_CLASS (parent_class)->finalize (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;
/* 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;
gst_faad_send_tags (faad);
gst_adapter_clear (faad->adapter);
} 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]);
}
}
gst_object_unref (faad);
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 void
clear_queued (GstFaad * faad)
{
g_list_foreach (faad->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (faad->queued);
faad->queued = NULL;
g_list_foreach (faad->gather, (GFunc) gst_mini_object_unref, NULL);
g_list_free (faad->gather);
faad->gather = NULL;
g_list_foreach (faad->decode, (GFunc) gst_mini_object_unref, NULL);
g_list_free (faad->decode);
faad->decode = 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;
GST_DEBUG_OBJECT (faad, "draining");
if (faad->segment.rate < 0.0) {
/* also decode tail = head of previous fragment to fill this one */
while (faad->decode) {
GstBuffer *buf = GST_BUFFER_CAST (faad->decode->data);
GST_DEBUG_OBJECT (faad, "processing delayed decode buffer");
gst_faad_chain (faad->sinkpad, buf);
faad->decode = g_list_delete_link (faad->decode, faad->decode);
}
/* if we have some queued frames for reverse playback, flush
* them now */
ret = flush_queued (faad);
/* move non-decoded leading buffers gathered in previous run
* to decode queue for this run */
faad->decode = g_list_reverse (faad->gather);
faad->gather = NULL;
} else {
/* squeeze any possible remaining frames that are pending sync */
gst_faad_chain (faad->sinkpad, NULL);
}
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:
gst_faad_reset_stream_state (faad);
res = gst_pad_push_event (faad->srcpad, event);
break;
case GST_EVENT_EOS:
gst_faad_drain (faad);
gst_faad_reset_stream_state (faad);
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_OBJECT (faad,
"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_OBJECT (faad, "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_OBJECT (faad,
"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_OBJECT (faad,
"Sending new NEWSEGMENT event, time %" GST_TIME_FORMAT
" - %" GST_TIME_FORMAT, GST_TIME_ARGS (new_start),
GST_TIME_ARGS (new_end));
faad->next_ts = GST_CLOCK_TIME_NONE;
faad->prev_ts = new_start;
}
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;
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 (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 (GstFaad * faad, guint8 * data, guint size, gboolean next,
guint * off)
{
guint n = 0;
gint snc;
gboolean ret = FALSE;
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. */
guint len;
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)
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;
}
#define FAAD_MAX_ERROR 10
#define FAAD_MAX_SYNC 10 * 8 * 1024
static GstFlowReturn
gst_faad_chain (GstPad * pad, GstBuffer * buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
guint input_size;
guint available;
guchar *input_data;
GstFaad *faad;
GstBuffer *outbuf;
faacDecFrameInfo info;
void *out;
gboolean run_loop = TRUE;
guint sync_off;
GstClockTime ts;
gboolean next;
faad = GST_FAAD (gst_pad_get_parent (pad));
if (G_LIKELY (buffer)) {
GST_LOG_OBJECT (faad, "buffer of size %d with ts: %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
if (GST_BUFFER_IS_DISCONT (buffer)) {
gst_faad_drain (faad);
gst_faad_reset_stream_state (faad);
faad->discont = TRUE;
}
gst_adapter_push (faad->adapter, buffer);
buffer = NULL;
next = TRUE;
} else {
next = FALSE;
}
available = gst_adapter_available (faad->adapter);
input_size = available;
if (G_UNLIKELY (!available))
goto out;
ts = gst_adapter_prev_timestamp (faad->adapter, NULL);
if (GST_CLOCK_TIME_IS_VALID (ts) && (ts != faad->prev_ts)) {
faad->prev_ts = ts;
} else {
/* nothing new */
ts = GST_CLOCK_TIME_NONE;
}
if (!GST_CLOCK_TIME_IS_VALID (faad->next_ts))
faad->next_ts = faad->prev_ts;
input_data = (guchar *) gst_adapter_peek (faad->adapter, available);
if (!faad->packetised) {
if (!gst_faad_sync (faad, input_data, input_size, next, &sync_off)) {
faad->sync_flush += sync_off;
input_size -= sync_off;
if (faad->sync_flush > FAAD_MAX_SYNC)
goto parse_failed;
else
goto out;
} else {
faad->sync_flush = 0;
input_data += sync_off;
input_size -= sync_off;
}
}
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;
gst_faad_send_tags (faad);
/* make sure we create new caps below */
faad->samplerate = 0;
faad->channels = 0;
}
/* decode cycle */
info.bytesconsumed = input_size;
info.error = 0;
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;
}
/* 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) {
/* mark discont for the next buffer */
faad->discont = TRUE;
/* flush a bit, arranges for resync next time */
input_size--;
faad->error_count++;
/* do not bail out at once, but know when to stop */
if (faad->error_count > FAAD_MAX_ERROR)
goto decode_failed;
else {
GST_WARNING_OBJECT (faad, "decoding error: %s",
faacDecGetErrorMessage (info.error));
goto out;
}
}
/* ok again */
faad->error_count = 0;
GST_LOG_OBJECT (faad, "%d bytes consumed, %d samples decoded",
(guint) info.bytesconsumed, (guint) info.samples);
if (info.bytesconsumed > input_size)
info.bytesconsumed = input_size;
input_size -= info.bytesconsumed;
input_data += info.bytesconsumed;
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;
/* 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,
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);
faad->bytes_in += info.bytesconsumed;
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_LOG_OBJECT (faad, "queued frame");
faad->queued = g_list_prepend (faad->queued, outbuf);
ret = GST_FLOW_OK;
}
if (ret != GST_FLOW_OK)
goto out;
}
}
} else {
if (faad->packetised && faad->segment.rate < 0.0) {
/* leading non-decoded frames used as tail
* for next preceding fragment */
outbuf = gst_adapter_take_buffer (faad->adapter, available);
available = 0;
outbuf = gst_buffer_make_metadata_writable (outbuf);
GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_DISCONT);
faad->gather = g_list_prepend (faad->gather, outbuf);
}
}
/* adjust to incoming new timestamp, if any, after decoder delay */
if (GST_CLOCK_TIME_IS_VALID (ts)) {
faad->next_ts = ts;
ts = GST_CLOCK_TIME_NONE;
}
}
out:
/* in raw case: (pretend) all consumed */
if (faad->packetised)
input_size = 0;
gst_adapter_flush (faad->adapter, available - input_size);
gst_object_unref (faad);
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_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("decoding error: %s", faacDecGetErrorMessage (info.error)));
ret = GST_FLOW_ERROR;
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;
}
parse_failed:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("failed to parse non-packetized stream"));
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:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
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:
gst_faad_reset (faad);
gst_faad_close_decoder (faad);
break;
case GST_STATE_CHANGE_READY_TO_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)
Reference in a new issue View git blame Copy permalink