+++ /dev/null
-Index: ffmpeg-r22950/libavcodec/allcodecs.c
-===================================================================
---- ffmpeg-r22950/libavcodec/allcodecs.c (revision 22950)
-+++ ffmpeg-r22950/libavcodec/allcodecs.c (working copy)
-@@ -334,6 +334,7 @@
- REGISTER_ENCDEC (LIBDIRAC, libdirac);
- REGISTER_ENCODER (LIBFAAC, libfaac);
- REGISTER_DECODER (LIBFAAD, libfaad);
-+ REGISTER_DECODER (LIBFAAD, libfaad2);
- REGISTER_ENCDEC (LIBGSM, libgsm);
- REGISTER_ENCDEC (LIBGSM_MS, libgsm_ms);
- REGISTER_ENCODER (LIBMP3LAME, libmp3lame);
-@@ -349,6 +350,7 @@
-
- /* parsers */
- REGISTER_PARSER (AAC, aac);
-+ REGISTER_PARSER (AAC, aac_latm);
- REGISTER_PARSER (AC3, ac3);
- REGISTER_PARSER (CAVSVIDEO, cavsvideo);
- REGISTER_PARSER (DCA, dca);
-Index: ffmpeg-r22950/libavcodec/avcodec.h
-===================================================================
---- ffmpeg-r22950/libavcodec/avcodec.h (revision 22950)
-+++ ffmpeg-r22950/libavcodec/avcodec.h (working copy)
-@@ -286,6 +286,7 @@
- CODEC_ID_MP2= 0x15000,
- CODEC_ID_MP3, ///< preferred ID for decoding MPEG audio layer 1, 2 or 3
- CODEC_ID_AAC,
-+ CODEC_ID_AAC_LATM,
- CODEC_ID_AC3,
- CODEC_ID_DTS,
- CODEC_ID_VORBIS,
-Index: ffmpeg-r22950/libavcodec/Makefile
-===================================================================
---- ffmpeg-r22950/libavcodec/Makefile (revision 22950)
-+++ ffmpeg-r22950/libavcodec/Makefile (working copy)
-@@ -505,7 +505,7 @@
- OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o
- OBJS-$(CONFIG_LIBDIRAC_ENCODER) += libdiracenc.o libdirac_libschro.o
- OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
--OBJS-$(CONFIG_LIBFAAD_DECODER) += libfaad.o
-+OBJS-$(CONFIG_LIBFAAD_DECODER) += libfaad.o latmaac.o
- OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
- OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
- OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
-@@ -529,7 +529,7 @@
-
- # parsers
- OBJS-$(CONFIG_AAC_PARSER) += aac_parser.o aac_ac3_parser.o \
-- mpeg4audio.o
-+ mpeg4audio.o latm_parser.o
- OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o \
- aac_ac3_parser.o
- OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs_parser.o
-Index: ffmpeg-r22950/libavcodec/latmaac.c
-===================================================================
---- ffmpeg-r22950/libavcodec/latmaac.c (revision 0)
-+++ ffmpeg-r22950/libavcodec/latmaac.c (revision 0)
-@@ -0,0 +1,642 @@
-+/*
-+ * copyright (c) 2008 Paul Kendall <paul@kcbbs.gen.nz>
-+ *
-+ * This file is part of FFmpeg.
-+ *
-+ * FFmpeg is free software; you can redistribute it and/or
-+ * modify it under the terms of the GNU Lesser General Public
-+ * License as published by the Free Software Foundation; either
-+ * version 2.1 of the License, or (at your option) any later version.
-+ *
-+ * FFmpeg 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
-+ * Lesser General Public License for more details.
-+ *
-+ * You should have received a copy of the GNU Lesser General Public
-+ * License along with FFmpeg; if not, write to the Free Software
-+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
-+ */
-+
-+/**
-+ * @file latmaac.c
-+ * LATM wrapped AAC decoder
-+ */
-+
-+#include <stdio.h>
-+#include <stdlib.h>
-+#include <string.h>
-+#include <math.h>
-+#include <sys/types.h>
-+
-+#include "parser.h"
-+#include "get_bits.h"
-+#include "put_bits.h"
-+#include "mpeg4audio.h"
-+#include "neaacdec.h"
-+
-+#define min(a,b) ((a)<(b) ? (a) : (b))
-+
-+
-+/*
-+ Note: This decoder filter is intended to decode LATM streams transferred
-+ in MPEG transport streams which are only supposed to contain one program.
-+ To do a more complex LATM demuxing a separate LATM demuxer should be used.
-+*/
-+
-+#define AAC_NONE 0 // mode not detected (or indicated in mediatype)
-+#define AAC_LATM 1 // LATM packets (ISO/IEC 14496-3 1.7.3 Multiplex layer)
-+
-+#define SYNC_LATM 0x2b7 // 11 bits
-+
-+#define MAX_SIZE 8*1024
-+
-+typedef struct AACConfig
-+{
-+ uint8_t extra[64]; // should be way enough
-+ int extrasize;
-+
-+ int audioObjectType;
-+ int samplingFrequencyIndex;
-+ int samplingFrequency;
-+ int channelConfiguration;
-+ int channels;
-+} AACConfig;
-+
-+typedef struct AACParser
-+{
-+ AACConfig config;
-+ uint8_t frameLengthType;
-+ uint16_t muxSlotLengthBytes;
-+
-+ uint8_t audio_mux_version;
-+ uint8_t audio_mux_version_A;
-+ int taraFullness;
-+ uint8_t config_crc;
-+ int64_t other_data_bits;
-+
-+ int mode;
-+ int offset; // byte offset in "buf" buffer
-+ uint8_t buf[MAX_SIZE]; // allocated buffer
-+ int count; // number of bytes written in buffer
-+} AACParser;
-+
-+typedef struct AACDecoder
-+{
-+ AACParser *parser;
-+ faacDecHandle aac_decoder;
-+ int open;
-+ uint32_t in_samplerate;
-+ uint8_t in_channels;
-+} AACDecoder;
-+
-+typedef struct {
-+ AACDecoder* decoder;
-+} FAACContext;
-+
-+static inline int64_t latm_get_value(GetBitContext *b)
-+{
-+ uint8_t bytesForValue = get_bits(b, 2);
-+ int64_t value = 0;
-+ int i;
-+ for (i=0; i<=bytesForValue; i++) {
-+ value <<= 8;
-+ value |= get_bits(b, 8);
-+ }
-+ return value;
-+}
-+
-+static void readGASpecificConfig(struct AACConfig *cfg, GetBitContext *b, PutBitContext *o)
-+{
-+ int framelen_flag;
-+ int dependsOnCoder;
-+ int ext_flag;
-+ int delay;
-+ int layerNr;
-+
-+ framelen_flag = get_bits(b, 1);
-+ put_bits(o, 1, framelen_flag);
-+ dependsOnCoder = get_bits(b, 1);
-+ put_bits(o, 1, dependsOnCoder);
-+
-+ if (dependsOnCoder) {
-+ delay = get_bits(b, 14);
-+ put_bits(o, 14, delay);
-+ }
-+ ext_flag = get_bits(b, 1);
-+ put_bits(o, 1, ext_flag);
-+ if (!cfg->channelConfiguration) {
-+ // program config element
-+ // TODO:
-+ }
-+
-+ if (cfg->audioObjectType == 6 || cfg->audioObjectType == 20) {
-+ layerNr = get_bits(b, 3);
-+ put_bits(o, 3, layerNr);
-+ }
-+ if (ext_flag) {
-+ if (cfg->audioObjectType == 22) {
-+ skip_bits(b, 5); // numOfSubFrame
-+ skip_bits(b, 11); // layer_length
-+
-+ put_bits(o, 16, 0);
-+ }
-+ if (cfg->audioObjectType == 17 ||
-+ cfg->audioObjectType == 19 ||
-+ cfg->audioObjectType == 20 ||
-+ cfg->audioObjectType == 23) {
-+
-+ skip_bits(b, 3); // stuff
-+ put_bits(o, 3, 0);
-+ }
-+
-+ skip_bits(b, 1); // extflag3
-+ put_bits(o, 1, 0);
-+ }
-+}
-+
-+static int readAudioSpecificConfig(struct AACConfig *cfg, GetBitContext *b)
-+{
-+ int ret;
-+ int sbr_present;
-+ PutBitContext o;
-+
-+ init_put_bits(&o, cfg->extra, sizeof(cfg->extra));
-+
-+ // returns the number of bits read
-+
-+ ret = 0;
-+ sbr_present = -1;
-+ // object
-+ cfg->audioObjectType = get_bits(b, 5);
-+ put_bits(&o, 5, cfg->audioObjectType);
-+ if (cfg->audioObjectType == 31) {
-+ uint8_t n = get_bits(b, 6);
-+ put_bits(&o, 6, n);
-+ cfg->audioObjectType = 32 + n;
-+ }
-+
-+ cfg->samplingFrequencyIndex = get_bits(b, 4);
-+ cfg->samplingFrequency = ff_mpeg4audio_sample_rates[cfg->samplingFrequencyIndex];
-+ put_bits(&o, 4, cfg->samplingFrequencyIndex);
-+ if (cfg->samplingFrequencyIndex == 0x0f) {
-+ uint32_t f = get_bits_long(b, 24);
-+ put_bits(&o, 24, f);
-+ cfg->samplingFrequency = f;
-+ }
-+ cfg->channelConfiguration = get_bits(b, 4);
-+ put_bits(&o, 4, cfg->channelConfiguration);
-+ cfg->channels = ff_mpeg4audio_channels[cfg->channelConfiguration];
-+
-+ if (cfg->audioObjectType == 5) {
-+ sbr_present = 1;
-+
-+ // TODO: parsing !!!!!!!!!!!!!!!!
-+ }
-+
-+ switch (cfg->audioObjectType) {
-+ case 1:
-+ case 2:
-+ case 3:
-+ case 4:
-+ case 6:
-+ case 7:
-+ case 17:
-+ case 19:
-+ case 20:
-+ case 21:
-+ case 22:
-+ case 23:
-+ readGASpecificConfig(cfg, b, &o);
-+ break;
-+ }
-+
-+ if (sbr_present == -1) {
-+ if (cfg->samplingFrequency <= 24000) {
-+ cfg->samplingFrequency *= 2;
-+ }
-+ }
-+
-+ // count the extradata
-+ ret = put_bits_count(&o);
-+ align_put_bits(&o);
-+ flush_put_bits(&o);
-+ cfg->extrasize = (ret + 7) >> 3;
-+ return ret;
-+}
-+
-+static void readStreamMuxConfig(struct AACParser *parser, GetBitContext *b)
-+{
-+ parser->audio_mux_version_A = 0;
-+ parser->audio_mux_version = get_bits(b, 1);
-+ if (parser->audio_mux_version == 1) { // audioMuxVersion
-+ parser->audio_mux_version_A = get_bits(b, 1);
-+ }
-+
-+ if (parser->audio_mux_version_A == 0) {
-+ int frame_length_type;
-+
-+ if (parser->audio_mux_version == 1) {
-+ parser->taraFullness = latm_get_value(b);
-+ }
-+ get_bits(b, 1); // allStreamSameTimeFraming = 1
-+ get_bits(b, 6); // numSubFrames = 0
-+ get_bits(b, 4); // numPrograms = 0
-+
-+ // for each program
-+ get_bits(b, 3); // numLayer = 0
-+
-+ // for each layer
-+ if (parser->audio_mux_version == 0) {
-+ // audio specific config.
-+ readAudioSpecificConfig(&parser->config, b);
-+ } else {
-+ int ascLen = latm_get_value(b);
-+ ascLen -= readAudioSpecificConfig(&parser->config, b);
-+
-+ // fill bits
-+ while (ascLen > 16) {
-+ skip_bits(b, 16);
-+ ascLen -= 16;
-+ }
-+ skip_bits(b, ascLen);
-+ }
-+
-+ // these are not needed... perhaps
-+ frame_length_type = get_bits(b, 3);
-+ parser->frameLengthType = frame_length_type;
-+ if (frame_length_type == 0) {
-+ get_bits(b, 8);
-+ } else if (frame_length_type == 1) {
-+ get_bits(b, 9);
-+ } else if (frame_length_type == 3 ||
-+ frame_length_type == 4 ||
-+ frame_length_type == 5) {
-+ int celp_table_index;
-+ celp_table_index = get_bits(b, 6);
-+ } else if (frame_length_type == 6 ||
-+ frame_length_type == 7) {
-+ int hvxc_table_index;
-+ hvxc_table_index = get_bits(b, 1);
-+ }
-+
-+ // other data
-+ parser->other_data_bits = 0;
-+ if (get_bits(b, 1)) {
-+ // other data present
-+ if (parser->audio_mux_version == 1) {
-+ parser->other_data_bits = latm_get_value(b);
-+ } else {
-+ // other data not present
-+ int esc, tmp;
-+ parser->other_data_bits = 0;
-+ do {
-+ parser->other_data_bits <<= 8;
-+ esc = get_bits(b, 1);
-+ tmp = get_bits(b, 8);
-+ parser->other_data_bits |= tmp;
-+ } while (esc);
-+ }
-+ }
-+
-+ // CRC
-+ if (get_bits(b, 1)) {
-+ parser->config_crc = get_bits(b, 8);
-+ }
-+ } else {
-+ // tbd
-+ }
-+}
-+
-+static void readPayloadLengthInfo(struct AACParser *parser, GetBitContext *b)
-+{
-+ uint8_t tmp;
-+ if (parser->frameLengthType == 0) {
-+ parser->muxSlotLengthBytes = 0;
-+ do {
-+ tmp = get_bits(b, 8);
-+ parser->muxSlotLengthBytes += tmp;
-+ } while (tmp == 255);
-+ } else {
-+ if (parser->frameLengthType == 5 ||
-+ parser->frameLengthType == 7 ||
-+ parser->frameLengthType == 3) {
-+ get_bits(b, 2);
-+ }
-+ }
-+}
-+
-+static void readAudioMuxElement(struct AACParser *parser, GetBitContext *b, uint8_t *payload, int *payloadsize)
-+{
-+ uint8_t use_same_mux = get_bits(b, 1);
-+ if (!use_same_mux) {
-+ readStreamMuxConfig(parser, b);
-+ }
-+
-+ if (parser->audio_mux_version_A == 0) {
-+ int j;
-+
-+ readPayloadLengthInfo(parser, b);
-+
-+ // copy data
-+ for (j=0; j<parser->muxSlotLengthBytes; j++) {
-+ *payload++ = get_bits(b, 8);
-+ }
-+ *payloadsize = parser->muxSlotLengthBytes;
-+
-+ // ignore otherdata
-+ } else {
-+ // TBD
-+ }
-+}
-+
-+static int readAudioSyncStream(struct AACParser *parser, GetBitContext *b, int size, uint8_t *payload, int *payloadsize)
-+{
-+ int muxlength;
-+ // ISO/IEC 14496-3 Table 1.28 - Syntax of AudioMuxElement()
-+ if (get_bits(b, 11) != 0x2b7) return -1; // not LATM
-+ muxlength = get_bits(b, 13);
-+
-+ if (3+muxlength > size) return 0; // not enough data
-+
-+ readAudioMuxElement(parser, b, payload, payloadsize);
-+
-+ // we don't parse anything else here...
-+ return (3+muxlength);
-+}
-+
-+
-+static void flush_buf(struct AACParser *parser, int offset) {
-+ int bytes_to_flush = min(parser->count, offset);
-+ int left = (parser->count - bytes_to_flush);
-+
-+ if (bytes_to_flush > 0) {
-+ if (left > 0) {
-+ memcpy(parser->buf, parser->buf+bytes_to_flush, left);
-+ parser->count = left;
-+ } else {
-+ parser->count = 0;
-+ }
-+ }
-+}
-+
-+static struct AACParser *latm_create_parser()
-+{
-+ struct AACParser *parser = (struct AACParser *)av_malloc(sizeof(struct AACParser));
-+ memset(parser, 0, sizeof(struct AACParser));
-+ return parser;
-+}
-+
-+static void latm_destroy_parser(struct AACParser *parser)
-+{
-+ av_free(parser);
-+}
-+
-+static void latm_flush(struct AACParser *parser)
-+{
-+ parser->offset = 0;
-+ parser->count = 0;
-+}
-+
-+static void latm_write_data(struct AACParser *parser, uint8_t *data, int len)
-+{
-+ // buffer overflow check... just ignore the data before
-+ if (parser->count + len > MAX_SIZE) {
-+ flush_buf(parser, parser->offset);
-+ parser->offset = 0;
-+ if (parser->count + len > MAX_SIZE) {
-+ int to_flush = (parser->count+len) - MAX_SIZE;
-+ flush_buf(parser, to_flush);
-+ }
-+ }
-+
-+ // append data
-+ memcpy(parser->buf+parser->count, data, len);
-+ parser->count += len;
-+}
-+
-+static int latm_parse_packet(struct AACParser *parser, uint8_t *data, int maxsize)
-+{
-+ /*
-+ Return value is either number of bytes parsed or
-+ -1 when failed.
-+ 0 = need more data.
-+ */
-+
-+ uint8_t *start = parser->buf + parser->offset;
-+ int bytes = parser->count - parser->offset;
-+ GetBitContext b;
-+ init_get_bits(&b, start, bytes);
-+
-+ if (parser->mode == AAC_LATM) {
-+ int outsize = 0;
-+ int ret = readAudioSyncStream(parser, &b, bytes, data, &outsize);
-+
-+ if (ret < 0) return -1;
-+ if (ret == 0) return 0;
-+
-+ // update the offset
-+ parser->offset += ret;
-+ return outsize;
-+ }
-+
-+ // check for syncwords
-+ while (bytes > 2) {
-+ if (show_bits(&b, 11) == SYNC_LATM) {
-+ // we must parse config first...
-+ int outsize = 0;
-+
-+ // check if there is a complete packet available...
-+ int ret = readAudioSyncStream(parser, &b, bytes, data, &outsize);
-+ if (ret < 0) return -1;
-+ if (ret == 0) return 0;
-+ parser->offset += ret;
-+
-+ parser->mode = AAC_LATM;
-+ return outsize;
-+ }
-+ skip_bits(&b, 8);
-+ parser->offset++;
-+ bytes--;
-+ }
-+ return 0;
-+}
-+
-+static void aac_filter_close(AACDecoder *decoder)
-+{
-+ if (decoder->aac_decoder) {
-+ NeAACDecClose(decoder->aac_decoder);
-+ decoder->aac_decoder = NULL;
-+ }
-+ decoder->open = 0;
-+}
-+
-+static int aac_decoder_open(AACDecoder *decoder)
-+{
-+ if (decoder->aac_decoder) return 0;
-+
-+ decoder->aac_decoder = NeAACDecOpen();
-+ if (!decoder->aac_decoder) return -1;
-+
-+ // are we going to initialize from decoder specific info ?
-+ if (decoder->parser->config.extrasize > 0) {
-+ char ret;
-+ unsigned long samplerate;
-+
-+ ret = NeAACDecInit2(decoder->aac_decoder, (unsigned char*)decoder->parser->config.extra, decoder->parser->config.extrasize, &samplerate, &decoder->in_channels);
-+ decoder->in_samplerate = samplerate;
-+ if (ret < 0) {
-+ aac_filter_close(decoder); // gone wrong ?
-+ return -1;
-+ }
-+ decoder->open = 1;
-+ } else {
-+ // we'll open the decoder later...
-+ decoder->open = 0;
-+ }
-+ return 0;
-+}
-+
-+static AACDecoder *aac_filter_create()
-+{
-+ AACDecoder *decoder = (AACDecoder *)av_malloc(sizeof(AACDecoder));
-+ decoder->parser = latm_create_parser();
-+ decoder->aac_decoder = NULL;
-+ decoder->open = 0;
-+ return (void *)decoder;
-+}
-+
-+static void aac_filter_destroy(AACDecoder *decoder)
-+{
-+ aac_filter_close(decoder);
-+ latm_destroy_parser(decoder->parser);
-+ av_free(decoder);
-+}
-+
-+static int aac_filter_receive(AACDecoder *decoder, void *out, int *out_size, uint8_t *data, int size)
-+{
-+ uint8_t tempbuf[32*1024];
-+ int ret;
-+ int consumed = size;
-+ int decoded;
-+ int max_size = *out_size;
-+
-+ *out_size = 0;
-+
-+ //-------------------------------------------------------------------------
-+ // Multiplex Parsing
-+ //-------------------------------------------------------------------------
-+
-+ latm_write_data(decoder->parser, data, size);
-+
-+ do {
-+ NeAACDecFrameInfo info;
-+ void *buf;
-+
-+ ret = latm_parse_packet(decoder->parser, tempbuf, sizeof(tempbuf));
-+ if (ret < 0) {
-+ latm_flush(decoder->parser);
-+ return consumed;
-+ }
-+ if (ret == 0) return consumed;
-+
-+ data = tempbuf;
-+ size = ret;
-+
-+ //-------------------------------------------------------------------------
-+ // Initialize decoder (if necessary)
-+ //-------------------------------------------------------------------------
-+ if (!decoder->open) {
-+ aac_filter_close(decoder);
-+ if (decoder->parser->mode == AAC_LATM) {
-+ ret = aac_decoder_open(decoder);
-+ if (ret < 0) return consumed;
-+ }
-+
-+ if(!decoder->open) return consumed;
-+ }
-+
-+ //-------------------------------------------------------------------------
-+ // Decode samples
-+ //-------------------------------------------------------------------------
-+ buf = NeAACDecDecode(decoder->aac_decoder, &info, data, size);
-+
-+ if (buf) {
-+ decoder->in_samplerate = info.samplerate;
-+ decoder->in_channels = info.channels;
-+
-+ //---------------------------------------------------------------------
-+ // Deliver decoded samples
-+ //---------------------------------------------------------------------
-+
-+ // kram dekoduje 16-bit. my vypustame 16-bit. takze by to malo byt okej
-+ decoded = info.samples * sizeof(short);
-+
-+ // napraskame tam sample
-+ *out_size += decoded;
-+ if(*out_size > max_size) {
-+ av_log(NULL, AV_LOG_ERROR, "overflow!\n");
-+ } else {
-+ memcpy(out, buf, decoded);
-+ out = (unsigned char *)out + decoded;
-+ }
-+ } else {
-+ // need more data
-+ break;
-+ }
-+
-+ } while (1); // decode all packets
-+ return consumed;
-+}
-+
-+static void aac_filter_getinfo(AACDecoder *decoder, int *sample_rate, int *channels)
-+{
-+ if(!decoder->open) return;
-+ *sample_rate = decoder->in_samplerate;
-+ *channels = decoder->in_channels;
-+}
-+
-+static int faac_decode_init(AVCodecContext *avctx)
-+{
-+ FAACContext *s = avctx->priv_data;
-+ avctx->sample_fmt = SAMPLE_FMT_S16;
-+ avctx->frame_size = 360;
-+ avctx->sample_rate = 48000;
-+ avctx->channels = 2;
-+ avctx->bit_rate = 8192 * 8 * avctx->sample_rate / avctx->frame_size;
-+ s->decoder = aac_filter_create();
-+ return 0;
-+}
-+
-+static int faac_decode_frame(AVCodecContext *avctx,
-+ void *data, int *data_size,
-+ AVPacket *avpkt)
-+{
-+ FAACContext *s = avctx->priv_data;
-+ int ret;
-+
-+ if (s->decoder == NULL) faac_decode_init(avctx);
-+ ret = aac_filter_receive(s->decoder, data, data_size, avpkt->data, avpkt->size);
-+ aac_filter_getinfo(s->decoder, &(avctx->sample_rate), &(avctx->channels));
-+ return ret;
-+}
-+
-+static int faac_decode_end(AVCodecContext *avctx)
-+{
-+ FAACContext *s = avctx->priv_data;
-+ if(s->decoder != NULL) {
-+ aac_filter_destroy(s->decoder);
-+ }
-+ return 0;
-+}
-+
-+AVCodec libfaad2_decoder = {
-+ .name = "AAC_LATM",
-+ .type = CODEC_TYPE_AUDIO,
-+ .id = CODEC_ID_AAC_LATM,
-+ .priv_data_size = sizeof (FAACContext),
-+ .init = faac_decode_init,
-+ .close = faac_decode_end,
-+ .decode = faac_decode_frame,
-+ .long_name = "AAC over LATM",
-+};
-Index: ffmpeg-r22950/libavcodec/latm_parser.c
-===================================================================
---- ffmpeg-r22950/libavcodec/latm_parser.c (revision 0)
-+++ ffmpeg-r22950/libavcodec/latm_parser.c (revision 0)
-@@ -0,0 +1,128 @@
-+/*
-+ * LATM parser
-+ * Copyright (c) 2008 Paul Kendall <paul@kcbbs.gen.nz>
-+ *
-+ * This file is part of FFmpeg.
-+ *
-+ * FFmpeg is free software; you can redistribute it and/or
-+ * modify it under the terms of the GNU Lesser General Public
-+ * License as published by the Free Software Foundation; either
-+ * version 2.1 of the License, or (at your option) any later version.
-+ *
-+ * FFmpeg 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
-+ * Lesser General Public License for more details.
-+ *
-+ * You should have received a copy of the GNU Lesser General Public
-+ * License along with FFmpeg; if not, write to the Free Software
-+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
-+ */
-+
-+/**
-+ * @file latm_parser.c
-+ * LATM parser
-+ */
-+
-+#include "parser.h"
-+
-+#define LATM_HEADER 0x56e000 // 0x2b7 (11 bits)
-+#define LATM_MASK 0xFFE000 // top 11 bits
-+#define LATM_SIZE_MASK 0x001FFF // bottom 13 bits
-+
-+typedef struct LATMParseContext{
-+ ParseContext pc;
-+ int count;
-+} LATMParseContext;
-+
-+/**
-+ * finds the end of the current frame in the bitstream.
-+ * @return the position of the first byte of the next frame, or -1
-+ */
-+static int latm_find_frame_end(AVCodecParserContext *s1, const uint8_t *buf,
-+ int buf_size) {
-+ LATMParseContext *s = s1->priv_data;
-+ ParseContext *pc = &s->pc;
-+ int pic_found, i;
-+ uint32_t state;
-+
-+ pic_found = pc->frame_start_found;
-+ state = pc->state;
-+
-+ i = 0;
-+ if(!pic_found){
-+ for(i=0; i<buf_size; i++){
-+ state = (state<<8) | buf[i];
-+ if((state & LATM_MASK) == LATM_HEADER){
-+ i++;
-+ s->count = - i;
-+ pic_found=1;
-+ break;
-+ }
-+ }
-+ }
-+
-+ if(pic_found){
-+ /* EOF considered as end of frame */
-+ if (buf_size == 0)
-+ return 0;
-+ if((state & LATM_SIZE_MASK) - s->count <= buf_size) {
-+ pc->frame_start_found = 0;
-+ pc->state = -1;
-+ return (state & LATM_SIZE_MASK) - s->count;
-+ }
-+ }
-+
-+ s->count += buf_size;
-+ pc->frame_start_found = pic_found;
-+ pc->state = state;
-+ return END_NOT_FOUND;
-+}
-+
-+static int latm_parse(AVCodecParserContext *s1,
-+ AVCodecContext *avctx,
-+ const uint8_t **poutbuf, int *poutbuf_size,
-+ const uint8_t *buf, int buf_size)
-+{
-+ LATMParseContext *s = s1->priv_data;
-+ ParseContext *pc = &s->pc;
-+ int next;
-+
-+ if(s1->flags & PARSER_FLAG_COMPLETE_FRAMES){
-+ next = buf_size;
-+ }else{
-+ next = latm_find_frame_end(s1, buf, buf_size);
-+
-+ if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
-+ *poutbuf = NULL;
-+ *poutbuf_size = 0;
-+ return buf_size;
-+ }
-+ }
-+ *poutbuf = buf;
-+ *poutbuf_size = buf_size;
-+ return next;
-+}
-+
-+static int latm_split(AVCodecContext *avctx,
-+ const uint8_t *buf, int buf_size)
-+{
-+ int i;
-+ uint32_t state= -1;
-+
-+ for(i=0; i<buf_size; i++){
-+ state= (state<<8) | buf[i];
-+ if((state & LATM_MASK) == LATM_HEADER)
-+ return i-2;
-+ }
-+ return 0;
-+}
-+
-+AVCodecParser aac_latm_parser = {
-+ { CODEC_ID_AAC_LATM },
-+ sizeof(LATMParseContext),
-+ NULL,
-+ latm_parse,
-+ ff_parse_close,
-+ latm_split,
-+};
-Index: ffmpeg-r22950/libavformat/mpegts.c
-===================================================================
---- ffmpeg-r22950/libavformat/mpegts.c (revision 22950)
-+++ ffmpeg-r22950/libavformat/mpegts.c (working copy)
-@@ -497,7 +497,7 @@
- { 0x04, AVMEDIA_TYPE_AUDIO, CODEC_ID_MP3 },
- { 0x0f, AVMEDIA_TYPE_AUDIO, CODEC_ID_AAC },
- { 0x10, AVMEDIA_TYPE_VIDEO, CODEC_ID_MPEG4 },
-- //{ 0x11, AVMEDIA_TYPE_AUDIO, CODEC_ID_AAC }, /* LATM syntax */
-+ { 0x11, AVMEDIA_TYPE_AUDIO, CODEC_ID_AAC_LATM }, /* LATM syntax */
- { 0x1b, AVMEDIA_TYPE_VIDEO, CODEC_ID_H264 },
- { 0xd1, AVMEDIA_TYPE_VIDEO, CODEC_ID_DIRAC },
- { 0xea, AVMEDIA_TYPE_VIDEO, CODEC_ID_VC1 },
-@@ -684,7 +684,7 @@
-
- if ((!pes->st && pes->stream->nb_streams == MAX_STREAMS) ||
- (pes->st && pes->st->discard == AVDISCARD_ALL) ||
-- code == 0x1be) /* padding_stream */
-+ code == 0x1be || code == 0x1fa) /* padding_stream */
- goto skip;
-
- /* stream not present in PMT */
-Index: ffmpeg-r22950/libavformat/mpegts.h
-===================================================================
---- ffmpeg-r22950/libavformat/mpegts.h (revision 22950)
-+++ ffmpeg-r22950/libavformat/mpegts.h (working copy)
-@@ -49,6 +49,7 @@
- #define STREAM_TYPE_PRIVATE_DATA 0x06
- #define STREAM_TYPE_AUDIO_AAC 0x0f
- #define STREAM_TYPE_VIDEO_MPEG4 0x10
-+#define STREAM_TYPE_AUDIO_AAC_LATM 0x11
- #define STREAM_TYPE_VIDEO_H264 0x1b
- #define STREAM_TYPE_VIDEO_VC1 0xea
- #define STREAM_TYPE_VIDEO_DIRAC 0xd1
-Index: ffmpeg-r22950/libavformat/mpeg.c
-===================================================================
---- ffmpeg-r22950/libavformat/mpeg.c (revision 22950)
-+++ ffmpeg-r22950/libavformat/mpeg.c (working copy)
-@@ -287,7 +287,7 @@
- /* find matching stream */
- if (!((startcode >= 0x1c0 && startcode <= 0x1df) ||
- (startcode >= 0x1e0 && startcode <= 0x1ef) ||
-- (startcode == 0x1bd) || (startcode == 0x1fd)))
-+ (startcode == 0x1bd) || (startcode == 0x1fa) || (startcode == 0x1fd)))
- goto redo;
- if (ppos) {
- *ppos = url_ftell(s->pb) - 4;
-@@ -454,6 +454,9 @@
- } else if(es_type == STREAM_TYPE_AUDIO_AAC){
- codec_id = CODEC_ID_AAC;
- type = AVMEDIA_TYPE_AUDIO;
-+ } else if(es_type == STREAM_TYPE_AUDIO_AAC_LATM){
-+ codec_id = CODEC_ID_AAC_LATM;
-+ type = CODEC_TYPE_AUDIO;
- } else if(es_type == STREAM_TYPE_VIDEO_MPEG4){
- codec_id = CODEC_ID_MPEG4;
- type = AVMEDIA_TYPE_VIDEO;
-Index: ffmpeg-r22950/libavformat/mpeg.h
-===================================================================
---- ffmpeg-r22950/libavformat/mpeg.h (revision 22950)
-+++ ffmpeg-r22950/libavformat/mpeg.h (working copy)
-@@ -53,6 +53,7 @@
- #define STREAM_TYPE_PRIVATE_DATA 0x06
- #define STREAM_TYPE_AUDIO_AAC 0x0f
- #define STREAM_TYPE_VIDEO_MPEG4 0x10
-+#define STREAM_TYPE_AUDIO_AAC_LATM 0x11
- #define STREAM_TYPE_VIDEO_H264 0x1b
-
- #define STREAM_TYPE_AUDIO_AC3 0x81
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