Added Paul Kendalls ffmpeg AAC-LATM codec. Allows HB to decode AAC-LATM

MPEG-TS DVB streams.


git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1550 b64f7644-9d1e-0410-96f1-a4d463321fa5

2 files changed, 896 insertions, 0 deletions

diff --git a/contrib/Jamfile b/contrib/Jamfile
index 9e226769a..a723f1b7b 100644
--- a/contrib/Jamfile
+++ b/contrib/Jamfile
@@ -78,6 +78,7 @@ LibFaad2 $(SUBDIR)/lib/libfaad.a : $(SUBDIR)/faad2.tar.gz ;
 rule LibAvCodec
 {
     FFMPEG_PATCH = "$(PATCH) -p0 < ../patch-ffmpeg.patch" ;
+    FFMPEG_PATCH += " && $(PATCH) -p0 < ../patch-ffmpeg-latm.patch " ;
     if $(OS) = CYGWIN
     {
         FFMPEG_PATCH += " && $(PATCH) -p1 < ../patch-ffmpeg-cygwin.patch " ;
diff --git a/contrib/patch-ffmpeg-latm.patch b/contrib/patch-ffmpeg-latm.patch
new file mode 100644
index 000000000..6f7c925b8
--- /dev/null
+++ b/contrib/patch-ffmpeg-latm.patch
@@ -0,0 +1,895 @@
+Index: libavcodec/Makefile
+===================================================================
+--- libavcodec/Makefile	(revision 14016)
++++ libavcodec/Makefile	(working copy)
+@@ -322,7 +322,7 @@
+ OBJS-$(CONFIG_LIBDIRAC_DECODER)        += libdiracdec.o
+ OBJS-$(CONFIG_LIBDIRAC_ENCODER)        += libdiracenc.o libdirac_libschro.o
+ OBJS-$(CONFIG_LIBFAAC)                 += libfaac.o
+-OBJS-$(CONFIG_LIBFAAD)                 += libfaad.o
++OBJS-$(CONFIG_LIBFAAD)                 += libfaad.o latmaac.o
+ OBJS-$(CONFIG_LIBGSM)                  += libgsm.o
+ OBJS-$(CONFIG_LIBMP3LAME)              += libmp3lame.o
+ OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER) += libschroedingerdec.o libschroedinger.o libdirac_libschro.o
+@@ -333,7 +333,7 @@
+ OBJS-$(CONFIG_LIBXVID)                 += libxvidff.o libxvid_rc.o
+ 
+ 
+-OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o mpeg4audio.o
++OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.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
+ OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o
+Index: libavcodec/latmaac.c
+===================================================================
+--- libavcodec/latmaac.c	(revision 0)
++++ libavcodec/latmaac.c	(revision 0)
+@@ -0,0 +1,624 @@
++/*
++ * copyright (c) 2008 Paul Kendall <[email protected]>
++ *
++ * 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 "bitstream.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 = get_bits(b, 1);
++    put_bits(o, 1, framelen_flag);
++    int dependsOnCoder = get_bits(b, 1);
++    put_bits(o, 1, dependsOnCoder);
++    int ext_flag;
++    int delay;
++    int layerNr;
++
++    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)
++{
++    PutBitContext o;
++    init_put_bits(&o, cfg->extra, sizeof(cfg->extra));
++
++    // returns the number of bits read
++    int ret = 0;
++    int 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) {
++        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
++        int 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 = get_bits(b, 6);
++        } else if (frame_length_type == 6 ||
++            frame_length_type == 7) {
++            int 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
++                parser->other_data_bits = 0;
++                int esc, tmp;
++                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)
++{
++    // ISO/IEC 14496-3 Table 1.28 - Syntax of AudioMuxElement()
++    if (get_bits(b, 11) != 0x2b7) return -1;        // not LATM
++    int 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 = NeAACDecInit2(decoder->aac_decoder, (unsigned char*)decoder->parser->config.extra, decoder->parser->config.extrasize, &decoder->in_samplerate, &decoder->in_channels);
++        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;
++}
++
++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;
++}
++
++void aac_filter_destroy(AACDecoder *decoder)
++{
++    aac_filter_close(decoder);
++    latm_destroy_parser(decoder->parser);
++    av_free(decoder);
++}
++
++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 {
++        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
++        //-------------------------------------------------------------------------
++        NeAACDecFrameInfo    info;
++        void *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;
++}
++
++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->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,
++                             uint8_t *buf, int buf_size)
++{
++    FAACContext *s = avctx->priv_data;
++    int ret;
++
++    if (s->decoder == NULL) faac_decode_init(avctx);
++    ret = aac_filter_receive(s->decoder, data, data_size, buf, buf_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: libavcodec/latm_parser.c
+===================================================================
+--- libavcodec/latm_parser.c	(revision 0)
++++ libavcodec/latm_parser.c	(revision 0)
+@@ -0,0 +1,128 @@
++/*
++ * LATM parser
++ * Copyright (c) 2008 Paul Kendall <[email protected]>
++ *
++ * 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: libavcodec/allcodecs.c
+===================================================================
+--- libavcodec/allcodecs.c	(revision 14016)
++++ libavcodec/allcodecs.c	(working copy)
+@@ -280,6 +280,7 @@
+     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);
+@@ -294,6 +295,7 @@
+ 
+     /* parsers */
+     REGISTER_PARSER  (AAC, aac);
++    REGISTER_PARSER  (AAC, aac_latm);
+     REGISTER_PARSER  (AC3, ac3);
+     REGISTER_PARSER  (CAVSVIDEO, cavsvideo);
+     REGISTER_PARSER  (DCA, dca);
+Index: libavcodec/avcodec.h
+===================================================================
+--- libavcodec/avcodec.h	(revision 14016)
++++ libavcodec/avcodec.h	(working copy)
+@@ -259,6 +259,7 @@
+ #if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
+     CODEC_ID_MPEG4AAC,
+ #endif
++    CODEC_ID_AAC_LATM,
+     CODEC_ID_AC3,
+     CODEC_ID_DTS,
+     CODEC_ID_VORBIS,
+Index: libavformat/mpegts.c
+===================================================================
+--- libavformat/mpegts.c	(revision 14016)
++++ libavformat/mpegts.c	(working copy)
+@@ -611,6 +611,7 @@
+         case STREAM_TYPE_VIDEO_H264:
+         case STREAM_TYPE_VIDEO_VC1:
+         case STREAM_TYPE_AUDIO_AAC:
++        case STREAM_TYPE_AUDIO_AAC_LATM:
+         case STREAM_TYPE_AUDIO_AC3:
+         case STREAM_TYPE_AUDIO_DTS:
+         case STREAM_TYPE_AUDIO_HDMV_DTS:
+@@ -832,7 +833,7 @@
+                     code = pes->header[3] | 0x100;
+                     if (!((code >= 0x1c0 && code <= 0x1df) ||
+                           (code >= 0x1e0 && code <= 0x1ef) ||
+-                          (code == 0x1bd) || (code == 0x1fd)))
++                          (code == 0x1bd) || (code == 0x1fa) || (code == 0x1fd)))
+                         goto skip;
+                     if (!pes->st) {
+                         /* allocate stream */
+@@ -948,6 +949,10 @@
+         codec_type = CODEC_TYPE_AUDIO;
+         codec_id = CODEC_ID_AAC;
+         break;
++    case STREAM_TYPE_AUDIO_AAC_LATM:
++        codec_type = CODEC_TYPE_AUDIO;
++        codec_id = CODEC_ID_AAC_LATM;
++        break;
+     case STREAM_TYPE_AUDIO_AC3:
+         codec_type = CODEC_TYPE_AUDIO;
+         codec_id = CODEC_ID_AC3;
+Index: libavformat/mpegts.h
+===================================================================
+--- libavformat/mpegts.h	(revision 14016)
++++ libavformat/mpegts.h	(working copy)
+@@ -50,6 +50,7 @@
+ #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
+ 
+Index: libavformat/mpeg.c
+===================================================================
+--- libavformat/mpeg.c	(revision 14016)
++++ libavformat/mpeg.c	(working copy)
+@@ -281,7 +281,7 @@
+     /* 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;
+@@ -439,6 +439,9 @@
+         } else if(es_type == STREAM_TYPE_AUDIO_AAC){
+             codec_id = CODEC_ID_AAC;
+             type = CODEC_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 = CODEC_TYPE_VIDEO;
+Index: libavformat/mpeg.h
+===================================================================
+--- libavformat/mpeg.h	(revision 14016)
++++ 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