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FFmpeg
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Merge remote-tracking branch 'qatar/master' 

* qatar/master:
  build: Fix Ogg demuxer dependencies
  build: Fix FLAC demuxer dependencies
  flac: Move flac functions shared between libraries to flac common code
  build: Fix CAF demuxer dependencies
  build: Fix MP2 muxer dependencies
  build: Add missing build rules for the ISMV muxer
  configure: Drop redundant mxf_d10 test dependency declaration
  Support AAC encoding via the external library fdk-aac
  libavcodec: Add more AAC profiles
  dct/fft-test: use a replacement getopt() if the system has none present.

Conflicts:
	Changelog
	libavcodec/Makefile
	libavcodec/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
tags/n1.0
Michael Niedermayer 13 years ago
parent
f9c823df13 0da29727ea
commit
620c6292b1
15 changed files with 602 additions and 91 deletions
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  1. +1
    -0
      Changelog
  2. +84
    -0
      compat/getopt.c
  3. +7
    -1
      configure
  4. +10
    -2
      doc/general.texi
  5. +16
    -14
      libavcodec/Makefile
  6. +1
    -0
      libavcodec/allcodecs.c
  7. +4
    -0
      libavcodec/avcodec.h
  8. +7
    -0
      libavcodec/dct-test.c
  9. +6
    -0
      libavcodec/fft-test.c
  10. +74
    -0
      libavcodec/flac.c
  11. +0
    -73
      libavcodec/flacdec.c
  12. +384
    -0
      libavcodec/libfdk-aacenc.c
  13. +4
    -0
      libavcodec/options_table.h
  14. +1
    -1
      libavcodec/version.h
  15. +3
    -0
      libavformat/Makefile

+ 1
- 0
Changelog View File

@@ -17,6 +17,7 @@ version next:
- Microsoft ATC Screen decoder
- RTSP listen mode
- TechSmith Screen Codec 2 decoder
- AAC encoding via libfdk-aac
- showwaves filter
- LucasArts SMUSH playback support
- SAMI demuxer and decoder


+ 84
- 0
compat/getopt.c View File

@@ -0,0 +1,84 @@
/*
* 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
*/

/*
* This file was copied from the following newsgroup posting:
*
* Newsgroups: mod.std.unix
* Subject: public domain AT&T getopt source
* Date: 3 Nov 85 19:34:15 GMT
*
* Here's something you've all been waiting for: the AT&T public domain
* source for getopt(3). It is the code which was given out at the 1985
* UNIFORUM conference in Dallas. I obtained it by electronic mail
* directly from AT&T. The people there assure me that it is indeed
* in the public domain.
*/

#define EOF (-1)

static int opterr = 1;
static int optind = 1;
static int optopt;
static char *optarg;

#undef fprintf

static int getopt(int argc, char *argv[], char *opts)
{
static int sp = 1;
int c;
char *cp;

if (sp == 1)
if (optind >= argc ||
argv[optind][0] != '-' || argv[optind][1] == '\0')
return EOF;
else if (!strcmp(argv[optind], "--")) {
optind++;
return EOF;
}
optopt = c = argv[optind][sp];
if (c == ':' || (cp = strchr(opts, c)) == NULL) {
fprintf(stderr, ": illegal option -- %c\n", c);
if (argv[optind][++sp] == '\0') {
optind++;
sp = 1;
}
return '?';
}
if (*++cp == ':') {
if (argv[optind][sp+1] != '\0')
optarg = &argv[optind++][sp+1];
else if(++optind >= argc) {
fprintf(stderr, ": option requires an argument -- %c\n", c);
sp = 1;
return '?';
} else
optarg = argv[optind++];
sp = 1;
} else {
if (argv[optind][++sp] == '\0') {
sp = 1;
optind++;
}
optarg = NULL;
}

return c;
}

+ 7
- 1
configure View File

@@ -178,6 +178,7 @@ External library support:
--enable-libdc1394 enable IIDC-1394 grabbing using libdc1394
and libraw1394 [no]
--enable-libfaac enable FAAC support via libfaac [no]
--enable-libfdk-aac enable AAC support via libfdk-aac [no]
--enable-libfreetype enable libfreetype [no]
--enable-libgsm enable GSM support via libgsm [no]
--enable-libiec61883 enable iec61883 via libiec61883 [no]
@@ -1053,6 +1054,7 @@ CONFIG_LIST="
libcelt
libdc1394
libfaac
libfdk_aac
libfreetype
libgsm
libiec61883
@@ -1210,6 +1212,7 @@ HAVE_LIST="
fork
getaddrinfo
gethrtime
getopt
GetProcessAffinityMask
GetProcessMemoryInfo
GetProcessTimes
@@ -1613,6 +1616,7 @@ h264_parser_select="golomb h264dsp h264pred"
libaacplus_encoder_deps="libaacplus"
libcelt_decoder_deps="libcelt"
libfaac_encoder_deps="libfaac"
libfdk_aac_encoder_deps="libfdk_aac"
libgsm_decoder_deps="libgsm"
libgsm_encoder_deps="libgsm"
libgsm_ms_decoder_deps="libgsm"
@@ -1797,7 +1801,6 @@ test_deps(){
done
}

mxf_d10_test_deps="avfilter"
seek_lavf_mxf_d10_test_deps="mxf_d10_test"

test_deps _muxer _demuxer \
@@ -3178,6 +3181,7 @@ check_func fcntl
check_func fork
check_func getaddrinfo $network_extralibs
check_func gethrtime
check_func getopt
check_func getrusage
check_struct "sys/time.h sys/resource.h" "struct rusage" ru_maxrss
check_func gettimeofday
@@ -3308,6 +3312,7 @@ enabled libcelt && require libcelt celt/celt.h celt_decode -lcelt0 &&
{ check_lib celt/celt.h celt_decoder_create_custom -lcelt0 ||
die "ERROR: libcelt version must be >= 0.11.0."; }
enabled libfaac && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac
enabled libfdk_aac && require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac
enabled libfreetype && require_pkg_config freetype2 "ft2build.h freetype/freetype.h" FT_Init_FreeType
enabled libgsm && require libgsm gsm/gsm.h gsm_create -lgsm
enabled libilbc && require libilbc ilbc.h WebRtcIlbcfix_InitDecode -lilbc
@@ -3684,6 +3689,7 @@ echo "libcdio support ${libcdio-no}"
echo "libcelt enabled ${libcelt-no}"
echo "libdc1394 support ${libdc1394-no}"
echo "libfaac enabled ${libfaac-no}"
echo "libfdk-aac enabled ${libfdk_aac-no}"
echo "libgsm enabled ${libgsm-no}"
echo "libiec61883 support ${libiec61883-no}"
echo "libilbc enabled ${libilbc-no}"


+ 10
- 2
doc/general.texi View File

@@ -26,8 +26,8 @@ instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp

@section OpenCORE and VisualOn libraries

Spun off Google Android sources, OpenCore and VisualOn libraries provide
encoders for a number of audio codecs.
Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
libraries provide encoders for a number of audio codecs.

@float NOTE
OpenCORE and VisualOn libraries are under the Apache License 2.0
@@ -63,6 +63,14 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.

@subsection Fraunhofer AAC library

Libav can make use of the Fraunhofer AAC library for AAC encoding.

Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libfdk-aac} to configure to enable it.

@section LAME

FFmpeg can make use of the LAME library for MP3 encoding.


+ 16
- 14
libavcodec/Makefile View File

@@ -624,22 +624,23 @@ OBJS-$(CONFIG_VIMA_DECODER) += vima.o adpcm_data.o
# libavformat dependencies
OBJS-$(CONFIG_ADTS_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_ADX_DEMUXER) += adx.o
OBJS-$(CONFIG_CAF_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_CAF_DEMUXER) += mpeg4audio.o mpegaudiodata.o \
ac3tab.o
OBJS-$(CONFIG_DV_DEMUXER) += dv_profile.o
OBJS-$(CONFIG_DV_MUXER) += dv_profile.o timecode.o
OBJS-$(CONFIG_FLAC_DEMUXER) += flacdec.o flacdata.o flac.o vorbis_data.o \
OBJS-$(CONFIG_FLAC_DEMUXER) += flac.o flacdata.o vorbis_data.o \
vorbis_parser.o xiph.o
OBJS-$(CONFIG_FLAC_MUXER) += flacdec.o flacdata.o flac.o vorbis_data.o
OBJS-$(CONFIG_FLAC_MUXER) += flac.o flacdata.o vorbis_data.o
OBJS-$(CONFIG_FLV_DEMUXER) += mpeg4audio.o
OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
OBJS-$(CONFIG_IFF_DEMUXER) += iff.o
OBJS-$(CONFIG_ISMV_MUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_LATM_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o vorbis_data.o \
flacdec.o flacdata.o flac.o
flac.o flacdata.o
OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o \
flacdec.o flacdata.o flac.o \
mpegaudiodata.o vorbis_data.o
OBJS-$(CONFIG_MATROSKA_MUXER) += mpeg4audio.o mpegaudiodata.o \
flac.o flacdata.o vorbis_data.o xiph.o
OBJS-$(CONFIG_MP2_MUXER) += mpegaudiodata.o mpegaudiodecheader.o
OBJS-$(CONFIG_MP3_MUXER) += mpegaudiodata.o mpegaudiodecheader.o
OBJS-$(CONFIG_MOV_DEMUXER) += mpeg4audio.o mpegaudiodata.o ac3tab.o timecode.o
@@ -648,22 +649,23 @@ OBJS-$(CONFIG_MPEGTS_MUXER) += mpegvideo.o mpeg4audio.o
OBJS-$(CONFIG_MPEGTS_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MXF_MUXER) += timecode.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
OBJS-$(CONFIG_OGG_DEMUXER) += flacdec.o flacdata.o flac.o \
dirac.o mpeg12data.o vorbis_parser.o \
xiph.o vorbis_data.o
OBJS-$(CONFIG_OGG_MUXER) += xiph.o flacdec.o flacdata.o flac.o \
OBJS-$(CONFIG_OGG_DEMUXER) += xiph.o flac.o flacdata.o \
mpeg12data.o vorbis_parser.o \
dirac.o vorbis_data.o
OBJS-$(CONFIG_OGG_MUXER) += xiph.o flac.o flacdata.o \
vorbis_data.o
OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o mpegvideo.o xiph.o
OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_WEBM_MUXER) += xiph.o mpeg4audio.o \
flacdec.o flacdata.o flac.o \
mpegaudiodata.o vorbis_data.o
OBJS-$(CONFIG_WEBM_MUXER) += mpeg4audio.o mpegaudiodata.o \
xiph.o flac.o flacdata.o \
vorbis_data.o
OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o

# external codec libraries
OBJS-$(CONFIG_LIBAACPLUS_ENCODER) += libaacplus.o
OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o audio_frame_queue.o
OBJS-$(CONFIG_LIBFDK_AAC_ENCODER) += libfdk-aacenc.o audio_frame_queue.o
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o


+ 1
- 0
libavcodec/allcodecs.c View File

@@ -417,6 +417,7 @@ void avcodec_register_all(void)
/* external libraries */
REGISTER_DECODER (LIBCELT, libcelt);
REGISTER_ENCODER (LIBFAAC, libfaac);
REGISTER_ENCODER (LIBFDK_AAC, libfdk_aac);
REGISTER_ENCDEC (LIBGSM, libgsm);
REGISTER_ENCDEC (LIBGSM_MS, libgsm_ms);
REGISTER_ENCDEC (LIBILBC, libilbc);


+ 4
- 0
libavcodec/avcodec.h View File

@@ -2825,6 +2825,10 @@ typedef struct AVCodecContext {
#define FF_PROFILE_AAC_LOW 1
#define FF_PROFILE_AAC_SSR 2
#define FF_PROFILE_AAC_LTP 3
#define FF_PROFILE_AAC_HE 4
#define FF_PROFILE_AAC_HE_V2 28
#define FF_PROFILE_AAC_LD 22
#define FF_PROFILE_AAC_ELD 38

#define FF_PROFILE_DTS 20
#define FF_PROFILE_DTS_ES 30


+ 7
- 0
libavcodec/dct-test.c View File

@@ -25,10 +25,13 @@
* Started from sample code by Juan J. Sierralta P.
*/

#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#if HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <math.h>

#include "libavutil/cpu.h"
@@ -519,6 +522,10 @@ static void help(void)
"-t speed test\n");
}

#if !HAVE_GETOPT
#include "compat/getopt.c"
#endif

int main(int argc, char **argv)
{
int test_idct = 0, test_248_dct = 0;


+ 6
- 0
libavcodec/fft-test.c View File

@@ -34,7 +34,9 @@
#include "rdft.h"
#endif
#include <math.h>
#if HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <stdlib.h>
#include <string.h>

@@ -229,6 +231,10 @@ enum tf_transform {
TRANSFORM_DCT,
};

#if !HAVE_GETOPT
#include "compat/getopt.c"
#endif

int main(int argc, char **argv)
{
FFTComplex *tab, *tab1, *tab_ref;


+ 74
- 0
libavcodec/flac.c View File

@@ -20,6 +20,9 @@
*/

#include "libavutil/crc.h"
#include "libavutil/log.h"
#include "bytestream.h"
#include "get_bits.h"
#include "flac.h"
#include "flacdata.h"

@@ -150,3 +153,74 @@ int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)

return count;
}

int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,
enum FLACExtradataFormat *format,
uint8_t **streaminfo_start)
{
if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
return 0;
}
if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
/* extradata contains STREAMINFO only */
if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
FLAC_STREAMINFO_SIZE-avctx->extradata_size);
}
*format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
*streaminfo_start = avctx->extradata;
} else {
if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
return 0;
}
*format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
*streaminfo_start = &avctx->extradata[8];
}
return 1;
}

void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
const uint8_t *buffer)
{
GetBitContext gb;
init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);

skip_bits(&gb, 16); /* skip min blocksize */
s->max_blocksize = get_bits(&gb, 16);
if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
s->max_blocksize);
s->max_blocksize = 16;
}

skip_bits(&gb, 24); /* skip min frame size */
s->max_framesize = get_bits_long(&gb, 24);

s->samplerate = get_bits_long(&gb, 20);
s->channels = get_bits(&gb, 3) + 1;
s->bps = get_bits(&gb, 5) + 1;

avctx->channels = s->channels;
avctx->sample_rate = s->samplerate;
avctx->bits_per_raw_sample = s->bps;

s->samples = get_bits_long(&gb, 32) << 4;
s->samples |= get_bits(&gb, 4);

skip_bits_long(&gb, 64); /* md5 sum */
skip_bits_long(&gb, 64); /* md5 sum */
}

void avpriv_flac_parse_block_header(const uint8_t *block_header,
int *last, int *type, int *size)
{
int tmp = bytestream_get_byte(&block_header);
if (last)
*last = tmp & 0x80;
if (type)
*type = tmp & 0x7F;
if (size)
*size = bytestream_get_be24(&block_header);
}

+ 0
- 73
libavcodec/flacdec.c View File

@@ -75,33 +75,6 @@ static const int64_t flac_channel_layouts[6] = {

static void allocate_buffers(FLACContext *s);

int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,
enum FLACExtradataFormat *format,
uint8_t **streaminfo_start)
{
if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
return 0;
}
if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
/* extradata contains STREAMINFO only */
if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
FLAC_STREAMINFO_SIZE-avctx->extradata_size);
}
*format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
*streaminfo_start = avctx->extradata;
} else {
if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
return 0;
}
*format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
*streaminfo_start = &avctx->extradata[8];
}
return 1;
}

static void flac_set_bps(FLACContext *s)
{
enum AVSampleFormat req = s->avctx->request_sample_fmt;
@@ -175,52 +148,6 @@ static void allocate_buffers(FLACContext *s)
}
}

void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
const uint8_t *buffer)
{
GetBitContext gb;
init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);

skip_bits(&gb, 16); /* skip min blocksize */
s->max_blocksize = get_bits(&gb, 16);
if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
s->max_blocksize);
s->max_blocksize = 16;
}

skip_bits(&gb, 24); /* skip min frame size */
s->max_framesize = get_bits_long(&gb, 24);

s->samplerate = get_bits_long(&gb, 20);
s->channels = get_bits(&gb, 3) + 1;
s->bps = get_bits(&gb, 5) + 1;

avctx->channels = s->channels;
avctx->sample_rate = s->samplerate;
avctx->bits_per_raw_sample = s->bps;

s->samples = get_bits_long(&gb, 32) << 4;
s->samples |= get_bits(&gb, 4);

skip_bits_long(&gb, 64); /* md5 sum */
skip_bits_long(&gb, 64); /* md5 sum */

dump_headers(avctx, s);
}

void avpriv_flac_parse_block_header(const uint8_t *block_header,
int *last, int *type, int *size)
{
int tmp = bytestream_get_byte(&block_header);
if (last)
*last = tmp & 0x80;
if (type)
*type = tmp & 0x7F;
if (size)
*size = bytestream_get_be24(&block_header);
}

/**
* Parse the STREAMINFO from an inline header.
* @param s the flac decoding context


+ 384
- 0
libavcodec/libfdk-aacenc.c View File

@@ -0,0 +1,384 @@
/*
* AAC encoder wrapper
* Copyright (c) 2012 Martin Storsjo
*
* 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
*/

#include <fdk-aac/aacenc_lib.h>

#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "libavutil/audioconvert.h"
#include "libavutil/opt.h"

typedef struct AACContext {
const AVClass *class;
HANDLE_AACENCODER handle;
int afterburner;
int eld_sbr;
int signaling;

AudioFrameQueue afq;
} AACContext;

static const AVOption aac_enc_options[] = {
{ "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { 1 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ "eld_sbr", "Enable SBR for ELD (for SBR in other configurations, use the -profile parameter)", offsetof(AACContext, eld_sbr), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ "signaling", "SBR/PS signaling style", offsetof(AACContext, signaling), AV_OPT_TYPE_INT, { -1 }, -1, 2, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "default", "Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)", 0, AV_OPT_TYPE_CONST, { -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ NULL }
};

static const AVClass aac_enc_class = {
"libfdk_aac", av_default_item_name, aac_enc_options, LIBAVUTIL_VERSION_INT
};

static const char *aac_get_error(AACENC_ERROR err)
{
switch (err) {
case AACENC_OK:
return "No error";
case AACENC_INVALID_HANDLE:
return "Invalid handle";
case AACENC_MEMORY_ERROR:
return "Memory allocation error";
case AACENC_UNSUPPORTED_PARAMETER:
return "Unsupported parameter";
case AACENC_INVALID_CONFIG:
return "Invalid config";
case AACENC_INIT_ERROR:
return "Initialization error";
case AACENC_INIT_AAC_ERROR:
return "AAC library initialization error";
case AACENC_INIT_SBR_ERROR:
return "SBR library initialization error";
case AACENC_INIT_TP_ERROR:
return "Transport library initialization error";
case AACENC_INIT_META_ERROR:
return "Metadata library initialization error";
case AACENC_ENCODE_ERROR:
return "Encoding error";
case AACENC_ENCODE_EOF:
return "End of file";
default:
return "Unknown error";
}
}

static int aac_encode_close(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;

if (s->handle)
aacEncClose(&s->handle);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);

return 0;
}

static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
int ret = AVERROR(EINVAL);
AACENC_InfoStruct info = { 0 };
CHANNEL_MODE mode;
AACENC_ERROR err;
int aot = FF_PROFILE_AAC_LOW + 1;
int sce = 0, cpe = 0;

if ((err = aacEncOpen(&s->handle, 0, avctx->channels)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to open the encoder: %s\n",
aac_get_error(err));
goto error;
}

if (avctx->profile != FF_PROFILE_UNKNOWN)
aot = avctx->profile + 1;

if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the AOT %d: %s\n",
aot, aac_get_error(err));
goto error;
}

if (aot == FF_PROFILE_AAC_ELD + 1 && s->eld_sbr) {
if ((err = aacEncoder_SetParam(s->handle, AACENC_SBR_MODE,
1)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to enable SBR for ELD: %s\n",
aac_get_error(err));
goto error;
}
}

if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE,
avctx->sample_rate)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the sample rate %d: %s\n",
avctx->sample_rate, aac_get_error(err));
goto error;
}

switch (avctx->channels) {
case 1: mode = MODE_1; sce = 1; cpe = 0; break;
case 2: mode = MODE_2; sce = 0; cpe = 1; break;
case 3: mode = MODE_1_2; sce = 1; cpe = 1; break;
case 4: mode = MODE_1_2_1; sce = 2; cpe = 1; break;
case 5: mode = MODE_1_2_2; sce = 1; cpe = 2; break;
case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break;
default:
av_log(avctx, AV_LOG_ERROR,
"Unsupported number of channels %d\n", avctx->channels);
goto error;
}

if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE,
mode)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR,
"Unable to set channel mode %d: %s\n", mode, aac_get_error(err));
goto error;
}

if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER,
1)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR,
"Unable to set wav channel order %d: %s\n",
mode, aac_get_error(err));
goto error;
}

if (avctx->flags & CODEC_FLAG_QSCALE) {
int mode = avctx->global_quality;
if (mode < 1 || mode > 5) {
av_log(avctx, AV_LOG_WARNING,
"VBR quality %d out of range, should be 1-5\n", mode);
mode = av_clip(mode, 1, 5);
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATEMODE,
mode)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the VBR bitrate mode %d: %s\n",
mode, aac_get_error(err));
goto error;
}
} else {
if (avctx->bit_rate <= 0) {
if (avctx->profile == FF_PROFILE_AAC_HE_V2) {
sce = 1;
cpe = 0;
}
avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44;
if (avctx->profile == FF_PROFILE_AAC_HE ||
avctx->profile == FF_PROFILE_AAC_HE_V2 ||
s->eld_sbr)
avctx->bit_rate /= 2;
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE,
avctx->bit_rate)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the bitrate %d: %s\n",
avctx->bit_rate, aac_get_error(err));
goto error;
}
}

/* Choose bitstream format - if global header is requested, use
* raw access units, otherwise use ADTS. */
if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,
avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : 2)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
aac_get_error(err));
goto error;
}

/* If no signaling mode is chosen, use explicit hierarchical signaling
* if using mp4 mode (raw access units, with global header) and
* implicit signaling if using ADTS. */
if (s->signaling < 0)
s->signaling = avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;

if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE,
s->signaling)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set signaling mode %d: %s\n",
s->signaling, aac_get_error(err));
goto error;
}

if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER,
s->afterburner)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n",
s->afterburner, aac_get_error(err));
goto error;
}

if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n",
aac_get_error(err));
return AVERROR(EINVAL);
}

if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n",
aac_get_error(err));
goto error;
}

#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
avctx->frame_size = info.frameLength;
avctx->delay = info.encoderDelay;
ff_af_queue_init(avctx, &s->afq);

if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
avctx->extradata_size = info.confSize;
avctx->extradata = av_mallocz(avctx->extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
ret = AVERROR(ENOMEM);
goto error;
}

memcpy(avctx->extradata, info.confBuf, info.confSize);
}
return 0;
error:
aac_encode_close(avctx);
return ret;
}

static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AACContext *s = avctx->priv_data;
AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
AACENC_InArgs in_args = { 0 };
AACENC_OutArgs out_args = { 0 };
int in_buffer_identifier = IN_AUDIO_DATA;
int in_buffer_size, in_buffer_element_size;
int out_buffer_identifier = OUT_BITSTREAM_DATA;
int out_buffer_size, out_buffer_element_size;
void *in_ptr, *out_ptr;
int ret;
AACENC_ERROR err;

/* handle end-of-stream small frame and flushing */
if (!frame) {
in_args.numInSamples = -1;
} else {
in_ptr = frame->data[0];
in_buffer_size = 2 * avctx->channels * frame->nb_samples;
in_buffer_element_size = 2;

in_args.numInSamples = avctx->channels * frame->nb_samples;
in_buf.numBufs = 1;
in_buf.bufs = &in_ptr;
in_buf.bufferIdentifiers = &in_buffer_identifier;
in_buf.bufSizes = &in_buffer_size;
in_buf.bufElSizes = &in_buffer_element_size;

/* add current frame to the queue */
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}

/* The maximum packet size is 6144 bits aka 768 bytes per channel. */
if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}

out_ptr = avpkt->data;
out_buffer_size = avpkt->size;
out_buffer_element_size = 1;
out_buf.numBufs = 1;
out_buf.bufs = &out_ptr;
out_buf.bufferIdentifiers = &out_buffer_identifier;
out_buf.bufSizes = &out_buffer_size;
out_buf.bufElSizes = &out_buffer_element_size;

if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args,
&out_args)) != AACENC_OK) {
if (!frame && err == AACENC_ENCODE_EOF)
return 0;
av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n",
aac_get_error(err));
return AVERROR(EINVAL);
}

if (!out_args.numOutBytes)
return 0;

/* Get the next frame pts & duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);

avpkt->size = out_args.numOutBytes;
*got_packet_ptr = 1;
return 0;
}

static const AVProfile profiles[] = {
{ FF_PROFILE_AAC_LOW, "LC" },
{ FF_PROFILE_AAC_HE, "HE-AAC" },
{ FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
{ FF_PROFILE_AAC_LD, "LD" },
{ FF_PROFILE_AAC_ELD, "ELD" },
{ FF_PROFILE_UNKNOWN },
};

static const AVCodecDefault aac_encode_defaults[] = {
{ "b", "0" },
{ NULL }
};

static const uint64_t aac_channel_layout[] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
0,
};

AVCodec ff_libfdk_aac_encoder = {
.name = "libfdk_aac",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_encode_init,
.encode2 = aac_encode_frame,
.close = aac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"),
.priv_class = &aac_enc_class,
.defaults = aac_encode_defaults,
.profiles = profiles,
.channel_layouts = aac_channel_layout,
};

+ 4
- 0
libavcodec/options_table.h View File

@@ -324,6 +324,10 @@ static const AVOption options[]={
{"aac_low", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_LOW }, INT_MIN, INT_MAX, A|E, "profile"},
{"aac_ssr", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_SSR }, INT_MIN, INT_MAX, A|E, "profile"},
{"aac_ltp", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_LTP }, INT_MIN, INT_MAX, A|E, "profile"},
{"aac_he", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_HE }, INT_MIN, INT_MAX, A|E, "profile"},
{"aac_he_v2", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_HE_V2 }, INT_MIN, INT_MAX, A|E, "profile"},
{"aac_ld", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_LD }, INT_MIN, INT_MAX, A|E, "profile"},
{"aac_eld", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_ELD }, INT_MIN, INT_MAX, A|E, "profile"},
{"dts", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_DTS }, INT_MIN, INT_MAX, A|E, "profile"},
{"dts_es", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_DTS_ES }, INT_MIN, INT_MAX, A|E, "profile"},
{"dts_96_24", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_DTS_96_24 }, INT_MIN, INT_MAX, A|E, "profile"},


+ 1
- 1
libavcodec/version.h View File

@@ -27,7 +27,7 @@
*/

#define LIBAVCODEC_VERSION_MAJOR 54
#define LIBAVCODEC_VERSION_MINOR 34
#define LIBAVCODEC_VERSION_MINOR 35
#define LIBAVCODEC_VERSION_MICRO 100

#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \


+ 3
- 0
libavformat/Makefile View File

@@ -132,6 +132,9 @@ OBJS-$(CONFIG_IMAGE2PIPE_DEMUXER) += img2dec.o img2.o
OBJS-$(CONFIG_IMAGE2PIPE_MUXER) += img2enc.o img2.o
OBJS-$(CONFIG_INGENIENT_DEMUXER) += ingenientdec.o rawdec.o
OBJS-$(CONFIG_IPMOVIE_DEMUXER) += ipmovie.o
OBJS-$(CONFIG_ISMV_MUXER) += movenc.o isom.o avc.o \
movenchint.o rtpenc_chain.o \
mov_chan.o
OBJS-$(CONFIG_ISS_DEMUXER) += iss.o
OBJS-$(CONFIG_IV8_DEMUXER) += iv8.o
OBJS-$(CONFIG_IVF_DEMUXER) += ivfdec.o


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