Index: configure =================================================================== --- configure (revision 23548) +++ configure (working copy) @@ -164,6 +164,7 @@ External library support: --enable-avisynth enable reading of AVISynth script files [no] --enable-bzlib enable bzlib [autodetect] + --enable-libaacplus enable AAC+ support via libaacplus [no] --enable-libopencore-amrnb enable AMR-NB de/encoding via libopencore-amrnb [no] --enable-libopencore-amrwb enable AMR-WB decoding via libopencore-amrwb [no] --enable-libdc1394 enable IIDC-1394 grabbing using libdc1394 @@ -915,6 +916,7 @@ libdc1394 libdirac libfaac + libaacplus libfaad libfaadbin libgsm @@ -1335,6 +1337,7 @@ libdirac_decoder_deps="libdirac !libschroedinger" libdirac_encoder_deps="libdirac" libfaac_encoder_deps="libfaac" +libaacplus_encoder_deps="libaacplus" libfaad_decoder_deps="libfaad" libfaadbin_extralibs='$ldl' libgsm_decoder_deps="libgsm" @@ -2610,6 +2613,8 @@ require libdirac libdirac_decoder/dirac_parser.h dirac_decoder_init $(pkg-config --libs dirac) && require libdirac libdirac_encoder/dirac_encoder.h dirac_encoder_init $(pkg-config --libs dirac) enabled libfaac && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac +enabled libaacplus && add_cflags $(pkg-config --cflags aacplus) && + require libaacplus libaacplus/aacenc.h AacInitDefaultConfig $(pkg-config --libs aacplus) enabled libfaad && require2 libfaad faad.h faacDecOpen -lfaad enabled libgsm && require libgsm gsm/gsm.h gsm_create -lgsm enabled libmp3lame && require libmp3lame lame/lame.h lame_init -lmp3lame -lm @@ -2877,6 +2882,7 @@ echo "libdc1394 support ${libdc1394-no}" echo "libdirac enabled ${libdirac-no}" echo "libfaac enabled ${libfaac-no}" +echo "libaacplus enabled ${libaacplus-no}" echo "libfaad enabled ${libfaad-no}" echo "libfaad dlopened ${libfaadbin-no}" echo "libgsm enabled ${libgsm-no}" Index: libavcodec/libaacplus.c =================================================================== --- libavcodec/libaacplus.c (revision 0) +++ libavcodec/libaacplus.c (revision 0) @@ -0,0 +1,265 @@ +/* + * Interface to libaacplus for aac+ encoding + * Copyright (c) 2002 The FFmpeg Team + * + * 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 libavcodec/libaacplus.c + * Interface to libaacplus for aac+ encoding. + */ + +#include "avcodec.h" +#include "put_bits.h" +#include "mpeg4audio.h" +#include +#include +#include +#include +#include +#include +#include +#include + +#define CORE_DELAY (1600) +/* ((1600 (core codec)*2 (multi rate) + 6*64 (sbr dec delay) - 2048 (sbr enc delay) + magic */ +#define INPUT_DELAY ((CORE_DELAY)*2 +6*64-2048+1) +/* the additional max resampler filter delay (source fs) */ +#define MAX_DS_FILTER_DELAY 16 +/* (96-64) makes AAC still some 64 core samples too early wrt SBR ... maybe -32 would be even more correct, + * but 1024-32 would need additional SBR bitstream delay by one frame */ +#define CORE_INPUT_OFFSET_PS (0) + +typedef struct AacpAudioContext { + struct AAC_ENCODER *aacEnc; + HANDLE_SBR_ENCODER hEnvEnc; + + AACENC_CONFIG config; + sbrConfiguration sbrConfig; + + IIR21_RESAMPLER IIR21_reSampler[MAX_CHANNELS]; + float inputBuffer[(AACENC_BLOCKSIZE*2 + MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS]; + + int nChannelsAAC, nChannelsSBR; + unsigned int sampleRateAAC; + int samplerate_index; + + unsigned int numAncDataBytes; + int useParametricStereo; + int coreWriteOffset; + int envReadOffset; + int writeOffset; + + unsigned char ancDataBytes[MAX_PAYLOAD_SIZE]; + unsigned char adtsDataBytes[ADTS_HEADER_SIZE]; + int adtsOffset; +} AacpAudioContext; + +/** + * Make AAC audio config object. + * @see 1.6.2.1 "Syntax - AudioSpecificConfig" + */ +static void put_audio_specific_config(AVCodecContext *avctx) +{ + PutBitContext pb; + AacpAudioContext *s = avctx->priv_data; + + init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); + put_bits(&pb, 5, FF_PROFILE_AAC_LOW+1); /* we have HE-AAC, but it's not compatible with mp3 container */ + put_bits(&pb, 4, s->samplerate_index); /* downsampled sample rate index */ + put_bits(&pb, 4, s->nChannelsAAC); + + put_bits(&pb, 1, 0); //frame length - 1024 samples + put_bits(&pb, 1, 0); //depend on core coder + put_bits(&pb, 1, 0); //is not extension + flush_put_bits(&pb); +} + + +static av_cold int Aacp_encode_init(AVCodecContext *avctx) +{ + AacpAudioContext *s = avctx->priv_data; + int i; + + s->coreWriteOffset = 0; + s->envReadOffset = 0; + s->writeOffset=INPUT_DELAY*MAX_CHANNELS; + + s->useParametricStereo = 0; + s->numAncDataBytes=0; + s->adtsOffset=ADTS_HEADER_SIZE; + + + /* number of channels */ + if (avctx->channels < 1 || avctx->channels > 2) { + av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels); + return -1; + } + + AacInitDefaultConfig(&s->config); + s->nChannelsAAC = s->nChannelsSBR = avctx->channels; + s->sampleRateAAC = avctx->sample_rate; + + /* ps or sbr? */ + if ( (avctx->channels == 2) && (avctx->bit_rate >= 16000) && (avctx->bit_rate < 44001) ) { + s->useParametricStereo = 1; + s->nChannelsAAC = 1; + + s->envReadOffset = (MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS; + s->coreWriteOffset = CORE_INPUT_OFFSET_PS; + s->writeOffset = s->envReadOffset; + } else { + /* set up 2:1 downsampling */ + InitIIR21_Resampler(&(s->IIR21_reSampler[0])); + if (avctx->channels == 2) InitIIR21_Resampler(&(s->IIR21_reSampler[1])); + if (s->IIR21_reSampler[0].delay > MAX_DS_FILTER_DELAY) { + av_log(avctx, AV_LOG_ERROR, "Initialisation of AAC+ encoder failed (IIR21_reSampler size too big)!\n"); + AacEncClose(s->aacEnc); + return -1; + } + s->writeOffset += s->IIR21_reSampler[0].delay * avctx->channels; //MAX_CHANNELS; + } + + + s->config.bitRate = avctx->bit_rate; + s->config.nChannelsIn=avctx->channels; + s->config.nChannelsOut=s->nChannelsAAC; + s->config.bandWidth=avctx->cutoff; + + /* set up SBR configuration */ + if(!IsSbrSettingAvail (avctx->bit_rate, s->nChannelsAAC, s->sampleRateAAC, &s->sampleRateAAC)) { + av_log(avctx, AV_LOG_ERROR, "No valid AAC+ SBR configuration found: br=%d, nChAAC=%d, srAAC=%d\n", + avctx->bit_rate, s->nChannelsAAC, s->sampleRateAAC ); + return -1; + } + + InitializeSbrDefaults (&s->sbrConfig); + s->sbrConfig.usePs = s->useParametricStereo; + + AdjustSbrSettings(&s->sbrConfig, avctx->bit_rate, s->nChannelsAAC, s->sampleRateAAC, AACENC_TRANS_FAC, 24000); + EnvOpen(&s->hEnvEnc, s->inputBuffer + s->coreWriteOffset, &s->sbrConfig, &s->config.bandWidth); + + /* set up AAC encoder, now that samling rate is known */ + s->config.sampleRate = s->sampleRateAAC; + if (AacEncOpen(&s->aacEnc, s->config) != 0){ + av_log(avctx, AV_LOG_ERROR, "Initialisation of AAC+ encoder failed!\n"); + AacEncClose(s->aacEnc); + return -1; + } + + init_plans(); + + /* set up input samples block size feed */ + avctx->frame_size = AACENC_BLOCKSIZE * 2; + avctx->coded_frame= avcodec_alloc_frame(); + avctx->coded_frame->key_frame= 1; + + avctx->extradata_size = 0; + /* Set decoder specific info */ + if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) { + + for(i = 0; i < 16; i++) + if(s->sampleRateAAC == ff_mpeg4audio_sample_rates[i]) + break; + if(i == 16){ + av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); + return -1; + } + s->samplerate_index=i; + + /* extradata init */ + avctx->extradata = av_malloc(2); + avctx->extradata_size = 2; + put_audio_specific_config(avctx); + s->adtsOffset=0; + } else { + /* ADTS header needed */ + adts_hdr(s->adtsDataBytes, &s->config); + } + + return 0; +} + + +static int Aacp_encode_frame(AVCodecContext *avctx, + unsigned char *frame, int buf_size, void *data) +{ + AacpAudioContext *s = avctx->priv_data; + short *TimeDataPcm = data; + int i, ch, outSamples, bytes_written; + + for (i=0; ichannels; i++) + s->inputBuffer[i+s->writeOffset] = (float) TimeDataPcm[i]; + + /* encode one SBR frame */ + EnvEncodeFrame( s->hEnvEnc, s->inputBuffer + s->envReadOffset, + s->inputBuffer + s->coreWriteOffset, avctx->channels, + &s->numAncDataBytes, s->ancDataBytes); + + /* 2:1 downsampling for AAC core */ + if (!s->useParametricStereo) { + for( ch=0; ch < avctx->channels; ch++ ) + IIR21_Downsample( &(s->IIR21_reSampler[ch] ), s->inputBuffer + s->writeOffset+ch, + AACENC_BLOCKSIZE * 2, avctx->channels, s->inputBuffer + ch, &outSamples, avctx->channels); + } + + /* encode one AAC frame */ + AacEncEncode( s->aacEnc, s->inputBuffer, + s->useParametricStereo ? 1 : avctx->channels, /* stride (step) */ s->ancDataBytes, + &s->numAncDataBytes, (unsigned *) (frame + s->adtsOffset), &bytes_written); + + /* write ADTS header (if needed) */ + if(s->adtsOffset){ + memcpy(frame, s->adtsDataBytes, s->adtsOffset); + adts_hdr_up(frame, bytes_written); + bytes_written += s->adtsOffset; + } + + if (s->useParametricStereo){ + memcpy( s->inputBuffer,s->inputBuffer+AACENC_BLOCKSIZE,CORE_INPUT_OFFSET_PS*sizeof(float)); + } else { + memmove( s->inputBuffer,s->inputBuffer+AACENC_BLOCKSIZE*2*avctx->channels,s->writeOffset*sizeof(float)); + } + + return bytes_written; +} + +static av_cold int Aacp_encode_close(AVCodecContext *avctx) +{ + AacpAudioContext *s = avctx->priv_data; + + av_freep(&avctx->coded_frame); + av_freep(&avctx->extradata); + + AacEncClose(s->aacEnc); + destroy_plans(); + EnvClose(s->hEnvEnc); + return 0; +} + +AVCodec libaacplus_encoder = { + "libaacplus", + CODEC_TYPE_AUDIO, + CODEC_ID_AAC, + sizeof(AacpAudioContext), + Aacp_encode_init, + Aacp_encode_frame, + Aacp_encode_close, + .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, + .long_name = NULL_IF_CONFIG_SMALL("libaacplus AAC+ (he-aac+, he-aac+ v2)"), +}; Index: libavcodec/allcodecs.c =================================================================== --- libavcodec/allcodecs.c (revision 23548) +++ libavcodec/allcodecs.c (working copy) @@ -339,6 +339,7 @@ /* external libraries */ REGISTER_ENCDEC (LIBDIRAC, libdirac); REGISTER_ENCODER (LIBFAAC, libfaac); + REGISTER_ENCODER (LIBAACPLUS, libaacplus); REGISTER_DECODER (LIBFAAD, libfaad); REGISTER_ENCDEC (LIBGSM, libgsm); REGISTER_ENCDEC (LIBGSM_MS, libgsm_ms); Index: libavcodec/Makefile =================================================================== --- libavcodec/Makefile (revision 23548) +++ libavcodec/Makefile (working copy) @@ -516,6 +516,7 @@ flacdec.o flacdata.o flac.o # external codec libraries +OBJS-$(CONFIG_LIBAACPLUS_ENCODER) += libaacplus.o OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o OBJS-$(CONFIG_LIBDIRAC_ENCODER) += libdiracenc.o libdirac_libschro.o OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o Index: doc/general.texi =================================================================== --- doc/general.texi (revision 23548) +++ doc/general.texi (working copy) @@ -516,6 +516,8 @@ @item 8SVX audio @tab @tab X @item AAC @tab E @tab X @tab encoding supported through external library libfaac +@item AACP @tab E @tab X + @tab encoding supported through external library libaacplus @item AC-3 @tab IX @tab X @item ADPCM 4X Movie @tab @tab X @item ADPCM CDROM XA @tab @tab X @@ -1030,7 +1032,7 @@ (@url{http://sourceware.org/cygwinports/}) : @example -yasm, libSDL-devel, libdirac-devel, libfaac-devel, libfaad-devel, libgsm-devel, +yasm, libSDL-devel, libdirac-devel, libfaac-devel, libaacplus-devel, libfaad-devel, libgsm-devel, libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel @end example