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|
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 <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 "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 <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: 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)
@@ -259,4 +259,5 @@
CODEC_ID_AAC,
+ 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
|