Ticket #5487: fixes-latm-aac.2.patch

libs/libavcodec/aac_parser.c

@@ -28 +28 @@
 #define AAC_HEADER_SIZE 7
 
 
-static const int aac_sample_rates[16] = {
+const int aac_sample_rates[16] = {
     96000, 88200, 64000, 48000, 44100, 32000,
     24000, 22050, 16000, 12000, 11025, 8000, 7350
 };
 
-static const int aac_channels[8] = {
+const int aac_channels[8] = {
     0, 1, 2, 3, 4, 5, 6, 8
 };
 

libs/libavcodec/allcodecs.c

@@ -271 +271 @@
     REGISTER_ENCDEC  (LIBAMR_WB, libamr_wb);
     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);
@@ -281 +282 @@
 
     /* parsers */
     REGISTER_PARSER  (AAC, aac);
+    REGISTER_PARSER  (AAC, latm);
     REGISTER_PARSER  (AC3, ac3);
     REGISTER_PARSER  (CAVSVIDEO, cavsvideo);
     REGISTER_PARSER  (DCA, dca);

libs/libavcodec/avcodec.h

@@ -240 +240 @@
 #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,

libs/libavcodec/latmaac.c

@@ +1 @@
+/*
+ * 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 "bitstream.h"
+#include "neaacdec.h"
+
+extern const int aac_sample_rates[16];
+extern const uint8_t aac_channels[8];
+
+#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 = aac_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 = aac_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 * 8);
+
+    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;
+}
+
+
+
+#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;
+}
+
+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,
+};
+
+AVCodecParser latm_parser = {
+    { CODEC_ID_AAC_LATM },
+    sizeof(LATMParseContext),
+    NULL,
+    latm_parse,
+    ff_parse_close,
+    latm_split,
+};

libs/libavcodec/libavcodec.pro

@@ -317 +317 @@
 contains( CONFIG_LIBA52, yes )                  { SOURCES *= liba52.c }
 contains( CONFIG_LIBAMR, yes )                  { SOURCES *= libamr.c }
 contains( CONFIG_LIBFAAC, yes )                 { SOURCES *= libfaac.c }
-contains( CONFIG_LIBFAAD, yes )                 { SOURCES *= libfaad.c }
+contains( CONFIG_LIBFAAD, yes )                 { SOURCES *= libfaad.c latmaac.c }
 contains( CONFIG_LIBGSM, yes )                  { SOURCES *= libgsm.c }
 contains( CONFIG_LIBMP3LAME, yes )              { SOURCES *= libmp3lame.c }
 contains( CONFIG_LIBTHEORA, yes )               { SOURCES *= libtheoraenc.c }

libs/libavcodec/libfaad.c

@@ -325 +325 @@
 
 // FIXME - raw AAC files - maybe just one entry will be enough
 AAC_CODEC(CODEC_ID_AAC, libfaad);
+AAC_CODEC(CODEC_ID_AAC_LATM, aac_latm);
 #if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
 // If it's mp4 file - usually embeded into Qt Mov
 AAC_CODEC(CODEC_ID_MPEG4AAC, mpeg4aac);

libs/libavformat/mpeg.c

@@ -443 +443 @@
         } 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;

libs/libavformat/mpeg.h

@@ -50 +50 @@
 #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

libs/libavformat/mpegts.c

@@ -864 +864 @@
         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_PRIVATE_DATA:
@@ -1434 +1435 @@
             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;

libs/libavformat/mpegts.h

@@ -55 +55 @@
 #define STREAM_TYPE_DSMCC_B         0x0b
 #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
 

libs/libmythtv/mpeg/mpegtables.cpp

@@ -772 +772 @@
         return "audio-mp2-layer[1,2,3]"; // EIT, PMT
     else if (StreamID::MPEG1Audio==streamID)
         return "audio-mp1-layer[1,2,3]"; // EIT, PMT
+    else if (StreamID::MPEG2AudioAmd1==streamID)                                                                      
+        return "audio-aac-latm"; // EIT, PMT                                                           
     else if (StreamID::AACAudio==streamID)
         return "audio-aac"; // EIT, PMT
     else if (StreamID::DTSAudio==streamID)

libs/libmythtv/mpeg/mpegtables.h

@@ -155 +155 @@
     {
         return ((StreamID::MPEG1Audio == type) ||
                 (StreamID::MPEG2Audio == type) ||
+                (StreamID::MPEG2AudioAmd1 == type) ||
                 (StreamID::AACAudio   == type) ||
                 (StreamID::AC3Audio   == type) ||
                 (StreamID::DTSAudio   == type));