Commit 136e19e1 authored by Janne Grunau's avatar Janne Grunau
Browse files

Add single stream LATM/LOAS decoder

The decoder is just a wrapper around the AAC decoder.
based on patch by Paul Kendall { paul <ät> kcbbs gen nz }

Originally committed as revision 25642 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 6c003e6d
......@@ -50,6 +50,7 @@ version <next>:
- transpose filter added
- ffmpeg -force_key_frames option added
- demuxer for receiving raw rtp:// URLs without an SDP description
- single stream LATM/LOAS decoder
version 0.6:
......
......@@ -1188,6 +1188,7 @@ rdft_select="fft"
# decoders / encoders / hardware accelerators
aac_decoder_select="mdct rdft"
aac_encoder_select="mdct"
aac_latm_decoder_select="aac_decoder aac_latm_parser"
ac3_decoder_select="mdct ac3_parser"
alac_encoder_select="lpc"
amrnb_decoder_select="lsp"
......
......@@ -576,6 +576,7 @@ OBJS-$(CONFIG_H264_PARSER) += h264_parser.o h264.o \
h264_loopfilter.o h264_cabac.o \
h264_cavlc.o h264_ps.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_AAC_LATM_PARSER) += latm_parser.o
OBJS-$(CONFIG_MJPEG_PARSER) += mjpeg_parser.o
OBJS-$(CONFIG_MLP_PARSER) += mlp_parser.o mlp.o
OBJS-$(CONFIG_MPEG4VIDEO_PARSER) += mpeg4video_parser.o h263.o \
......
......@@ -3,6 +3,10 @@
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
......@@ -2098,6 +2102,261 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
return 0;
}
#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
struct LATMContext {
AACContext aac_ctx; ///< containing AACContext
int initialized; ///< initilized after a valid extradata was seen
// parser data
int audio_mux_version_A; ///< LATM syntax version
int frame_length_type; ///< 0/1 variable/fixed frame length
int frame_length; ///< frame length for fixed frame length
};
static inline uint32_t latm_get_value(GetBitContext *b)
{
int length = get_bits(b, 2);
return get_bits_long(b, (length+1)*8);
}
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
GetBitContext *gb)
{
AVCodecContext *avctx = latmctx->aac_ctx.avctx;
MPEG4AudioConfig m4ac;
int config_start_bit = get_bits_count(gb);
int bits_consumed, esize;
if (config_start_bit % 8) {
av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
"config not byte aligned.\n", 1);
return AVERROR_INVALIDDATA;
} else {
bits_consumed =
decode_audio_specific_config(NULL, avctx, &m4ac,
gb->buffer + (config_start_bit / 8),
get_bits_left(gb) / 8);
if (bits_consumed < 0)
return AVERROR_INVALIDDATA;
esize = (bits_consumed+7) / 8;
if (avctx->extradata_size <= esize) {
av_free(avctx->extradata);
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
}
avctx->extradata_size = esize;
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
}
return bits_consumed;
}
static int read_stream_mux_config(struct LATMContext *latmctx,
GetBitContext *gb)
{
int ret, audio_mux_version = get_bits(gb, 1);
latmctx->audio_mux_version_A = 0;
if (audio_mux_version)
latmctx->audio_mux_version_A = get_bits(gb, 1);
if (!latmctx->audio_mux_version_A) {
if (audio_mux_version)
latm_get_value(gb); // taraFullness
skip_bits(gb, 1); // allStreamSameTimeFraming
skip_bits(gb, 6); // numSubFrames
// numPrograms
if (get_bits(gb, 4)) { // numPrograms
av_log_missing_feature(latmctx->aac_ctx.avctx,
"multiple programs are not supported\n", 1);
return AVERROR_PATCHWELCOME;
}
// for each program (which there is only on in DVB)
// for each layer (which there is only on in DVB)
if (get_bits(gb, 3)) { // numLayer
av_log_missing_feature(latmctx->aac_ctx.avctx,
"multiple layers are not supported\n", 1);
return AVERROR_PATCHWELCOME;
}
// for all but first stream: use_same_config = get_bits(gb, 1);
if (!audio_mux_version) {
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
return ret;
} else {
int ascLen = latm_get_value(gb);
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
return ret;
ascLen -= ret;
skip_bits_long(gb, ascLen);
}
latmctx->frame_length_type = get_bits(gb, 3);
switch (latmctx->frame_length_type) {
case 0:
skip_bits(gb, 8); // latmBufferFullness
break;
case 1:
latmctx->frame_length = get_bits(gb, 9);
break;
case 3:
case 4:
case 5:
skip_bits(gb, 6); // CELP frame length table index
break;
case 6:
case 7:
skip_bits(gb, 1); // HVXC frame length table index
break;
}
if (get_bits(gb, 1)) { // other data
if (audio_mux_version) {
latm_get_value(gb); // other_data_bits
} else {
int esc;
do {
esc = get_bits(gb, 1);
skip_bits(gb, 8);
} while (esc);
}
}
if (get_bits(gb, 1)) // crc present
skip_bits(gb, 8); // config_crc
}
return 0;
}
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
{
uint8_t tmp;
if (ctx->frame_length_type == 0) {
int mux_slot_length = 0;
do {
tmp = get_bits(gb, 8);
mux_slot_length += tmp;
} while (tmp == 255);
return mux_slot_length;
} else if (ctx->frame_length_type == 1) {
return ctx->frame_length;
} else if (ctx->frame_length_type == 3 ||
ctx->frame_length_type == 5 ||
ctx->frame_length_type == 7) {
skip_bits(gb, 2); // mux_slot_length_coded
}
return 0;
}
static int read_audio_mux_element(struct LATMContext *latmctx,
GetBitContext *gb)
{
int err;
uint8_t use_same_mux = get_bits(gb, 1);
if (!use_same_mux) {
if ((err = read_stream_mux_config(latmctx, gb)) < 0)
return err;
} else if (!latmctx->aac_ctx.avctx->extradata) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
"no decoder config found\n");
return AVERROR(EAGAIN);
}
if (latmctx->audio_mux_version_A == 0) {
int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
return AVERROR_INVALIDDATA;
} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
"frame length mismatch %d << %d\n",
mux_slot_length_bytes * 8, get_bits_left(gb));
return AVERROR_INVALIDDATA;
}
}
return 0;
}
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
AVPacket *avpkt)
{
struct LATMContext *latmctx = avctx->priv_data;
int muxlength, err;
GetBitContext gb;
if (avpkt->size == 0)
return 0;
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
// check for LOAS sync word
if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
return AVERROR_INVALIDDATA;
muxlength = get_bits(&gb, 13);
// not enough data, the parser should have sorted this
if (muxlength+3 > avpkt->size)
return AVERROR_INVALIDDATA;
if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
return err;
if (!latmctx->initialized) {
if (!avctx->extradata) {
*out_size = 0;
return avpkt->size;
} else {
if ((err = aac_decode_init(avctx)) < 0)
return err;
latmctx->initialized = 1;
}
}
if (show_bits(&gb, 12) == 0xfff) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
"ADTS header detected, probably as result of configuration "
"misparsing\n");
return AVERROR_INVALIDDATA;
}
if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
return err;
return muxlength;
}
av_cold static int latm_decode_init(AVCodecContext *avctx)
{
struct LATMContext *latmctx = avctx->priv_data;
int ret;
ret = aac_decode_init(avctx);
if (avctx->extradata_size > 0) {
latmctx->initialized = !ret;
} else {
latmctx->initialized = 0;
}
return ret;
}
AVCodec aac_decoder = {
"aac",
AVMEDIA_TYPE_AUDIO,
......@@ -2113,3 +2372,23 @@ AVCodec aac_decoder = {
},
.channel_layouts = aac_channel_layout,
};
/*
Note: This decoder filter is intended to decode LATM streams transferred
in MPEG transport streams which only contain one program.
To do a more complex LATM demuxing a separate LATM demuxer should be used.
*/
AVCodec aac_latm_decoder = {
.name = "aac_latm",
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_AAC_LATM,
.priv_data_size = sizeof(struct LATMContext),
.init = latm_decode_init,
.close = aac_decode_close,
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum SampleFormat[]) {
SAMPLE_FMT_S16,SAMPLE_FMT_NONE
},
.channel_layouts = aac_channel_layout,
};
......@@ -220,6 +220,7 @@ void avcodec_register_all(void)
/* audio codecs */
REGISTER_ENCDEC (AAC, aac);
REGISTER_DECODER (AAC_LATM, aac_latm);
REGISTER_ENCDEC (AC3, ac3);
REGISTER_ENCDEC (ALAC, alac);
REGISTER_DECODER (ALS, als);
......@@ -366,6 +367,7 @@ void avcodec_register_all(void)
/* parsers */
REGISTER_PARSER (AAC, aac);
REGISTER_PARSER (AAC_LATM, aac_latm);
REGISTER_PARSER (AC3, ac3);
REGISTER_PARSER (CAVSVIDEO, cavsvideo);
REGISTER_PARSER (DCA, dca);
......
......@@ -31,7 +31,7 @@
#include "libavutil/cpu.h"
#define LIBAVCODEC_VERSION_MAJOR 52
#define LIBAVCODEC_VERSION_MINOR 93
#define LIBAVCODEC_VERSION_MINOR 94
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
......@@ -376,6 +376,7 @@ enum CodecID {
CODEC_ID_ATRAC1,
CODEC_ID_BINKAUDIO_RDFT,
CODEC_ID_BINKAUDIO_DCT,
CODEC_ID_AAC_LATM,
/* subtitle codecs */
CODEC_ID_DVD_SUBTITLE= 0x17000,
......
/*
* 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
* AAC LATM parser
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <sys/types.h>
#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;
}
AVCodecParser aac_latm_parser = {
{ CODEC_ID_AAC_LATM },
sizeof(LATMParseContext),
NULL,
latm_parse,
ff_parse_close
};
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment