[FFmpeg-devel] [PATCH] avcodec: Remove libfaac, the internal AAC encoder is better

Robert Kr├╝ger krueger at lesspain.software
Mon Aug 15 13:51:34 EEST 2016


On Sun, Aug 14, 2016 at 5:05 PM, Rostislav Pehlivanov <atomnuker at gmail.com>
wrote:

> On 10 April 2016 at 16:38, Kieran Kunhya <kieran at kunhya.com> wrote:
>
> > ---
> >  Changelog              |   1 +
> >  configure              |   6 --
> >  doc/encoders.texi      | 105 ---------------------
> >  doc/ffserver.conf      |   2 +-
> >  doc/general.texi       |   2 +-
> >  doc/muxers.texi        |   4 +-
> >  doc/platform.texi      |   2 +-
> >  libavcodec/Makefile    |   1 -
> >  libavcodec/allcodecs.c |   1 -
> >  libavcodec/libfaac.c   | 248 ------------------------------
> > -------------------
> >  libavcodec/version.h   |   2 +-
> >  11 files changed, 7 insertions(+), 367 deletions(-)
> >  delete mode 100644 libavcodec/libfaac.c
> >
> > diff --git a/Changelog b/Changelog
> > index b4a4dd7..7bcb0c4 100644
> > --- a/Changelog
> > +++ b/Changelog
> > @@ -22,6 +22,7 @@ version <next>:
> >  - musx demuxer
> >  - aix demuxer
> >  - remap filter
> > +- libfaac removed
> >
> >  version 3.0:
> >  - Common Encryption (CENC) MP4 encoding and decoding support
> > diff --git a/configure b/configure
> > index 94a66d8..32d710d 100755
> > --- a/configure
> > +++ b/configure
> > @@ -219,7 +219,6 @@ External library support:
> >    --enable-libcdio         enable audio CD grabbing with libcdio [no]
> >    --enable-libdc1394       enable IIDC-1394 grabbing using libdc1394
> >                             and libraw1394 [no]
> > -  --enable-libfaac         enable AAC encoding via libfaac [no]
> >    --enable-libfdk-aac      enable AAC de/encoding via libfdk-aac [no]
> >    --enable-libflite        enable flite (voice synthesis) support via
> > libflite [no]
> >    --enable-libfreetype     enable libfreetype, needed for drawtext
> filter
> > [no]
> > @@ -1467,7 +1466,6 @@ EXTERNAL_LIBRARY_LIST="
> >      libcdio
> >      libcelt
> >      libdc1394
> > -    libfaac
> >      libfdk_aac
> >      libflite
> >      libfontconfig
> > @@ -2673,8 +2671,6 @@ pcm_mulaw_at_encoder_select="audio_frame_queue"
> >  chromaprint_muxer_deps="chromaprint"
> >  h264_videotoolbox_encoder_deps="videotoolbox_encoder pthreads"
> >  libcelt_decoder_deps="libcelt"
> > -libfaac_encoder_deps="libfaac"
> > -libfaac_encoder_select="audio_frame_queue"
> >  libfdk_aac_decoder_deps="libfdk_aac"
> >  libfdk_aac_encoder_deps="libfdk_aac"
> >  libfdk_aac_encoder_select="audio_frame_queue"
> > @@ -4946,7 +4942,6 @@ die_license_disabled gpl libxvid
> >  die_license_disabled gpl x11grab
> >
> >  die_license_disabled nonfree cuda
> > -die_license_disabled nonfree libfaac
> >  die_license_disabled nonfree nvenc
> >  enabled gpl && die_license_disabled_gpl nonfree libfdk_aac
> >  enabled gpl && die_license_disabled_gpl nonfree openssl
> > @@ -5534,7 +5529,6 @@ enabled libcelt           && require libcelt
> > celt/celt.h celt_decode -lcelt0 &&
> >                               { check_lib celt/celt.h
> > celt_decoder_create_custom -lcelt0 ||
> >                                 die "ERROR: libcelt must be installed and
> > version must be >= 0.11.0."; }
> >  enabled libcaca           && require_pkg_config caca caca.h
> > caca_create_canvas
> > -enabled libfaac           && require2 libfaac "stdint.h faac.h"
> > faacEncGetVersion -lfaac
> >  enabled libfdk_aac        && { use_pkg_config fdk-aac
> > "fdk-aac/aacenc_lib.h" aacEncOpen ||
> >                                 { require libfdk_aac fdk-aac/aacenc_lib.h
> > aacEncOpen -lfdk-aac &&
> >                                   warn "using libfdk without pkg-config";
> > } }
> > diff --git a/doc/encoders.texi b/doc/encoders.texi
> > index f38cad3..5c09136 100644
> > --- a/doc/encoders.texi
> > +++ b/doc/encoders.texi
> > @@ -611,111 +611,6 @@ and slightly improves compression.
> >
> >  @end table
> >
> > - at anchor{libfaac}
> > - at section libfaac
> > -
> > -libfaac AAC (Advanced Audio Coding) encoder wrapper.
> > -
> > -This encoder is of much lower quality and is more unstable than any
> other
> > AAC
> > -encoders, so it's highly recommended to instead use other encoders, like
> > - at ref{aacenc,,the native FFmpeg AAC encoder}.
> > -
> > -This encoder also requires the presence of the libfaac headers and
> library
> > -during configuration. You need to explicitly configure the build with
> > - at code{--enable-libfaac --enable-nonfree}.
> > -
> > - at subsection Options
> > -
> > -The following shared FFmpeg codec options are recognized.
> > -
> > -The following options are supported by the libfaac wrapper. The
> > - at command{faac}-equivalent of the options are listed in parentheses.
> > -
> > - at table @option
> > - at item b (@emph{-b})
> > -Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
> > -is not explicitly specified, it is automatically set to a suitable
> > -value depending on the selected profile. @command{faac} bitrate is
> > -expressed in kilobits/s.
> > -
> > -Note that libfaac does not support CBR (Constant Bit Rate) but only
> > -ABR (Average Bit Rate).
> > -
> > -If VBR mode is enabled this option is ignored.
> > -
> > - at item ar (@emph{-R})
> > -Set audio sampling rate (in Hz).
> > -
> > - at item ac (@emph{-c})
> > -Set the number of audio channels.
> > -
> > - at item cutoff (@emph{-C})
> > -Set cutoff frequency. If not specified (or explicitly set to 0) it
> > -will use a value automatically computed by the library. Default value
> > -is 0.
> > -
> > - at item profile
> > -Set audio profile.
> > -
> > -The following profiles are recognized:
> > - at table @samp
> > - at item aac_main
> > -Main AAC (Main)
> > -
> > - at item aac_low
> > -Low Complexity AAC (LC)
> > -
> > - at item aac_ssr
> > -Scalable Sample Rate (SSR)
> > -
> > - at item aac_ltp
> > -Long Term Prediction (LTP)
> > - at end table
> > -
> > -If not specified it is set to @samp{aac_low}.
> > -
> > - at item flags +qscale
> > -Set constant quality VBR (Variable Bit Rate) mode.
> > -
> > - at item global_quality
> > -Set quality in VBR mode as an integer number of lambda units.
> > -
> > -Only relevant when VBR mode is enabled with @code{flags +qscale}.  The
> > -value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
> > -and used to set the quality value used by libfaac. A reasonable range
> > -for the option value in QP units is [10-500], the higher the value the
> > -higher the quality.
> > -
> > - at item q (@emph{-q})
> > -Enable VBR mode when set to a non-negative value, and set constant
> > -quality value as a double floating point value in QP units.
> > -
> > -The value sets the quality value used by libfaac. A reasonable range
> > -for the option value is [10-500], the higher the value the higher the
> > -quality.
> > -
> > -This option is valid only using the @command{ffmpeg} command-line
> > -tool. For library interface users, use @option{global_quality}.
> > - at end table
> > -
> > - at subsection Examples
> > -
> > - at itemize
> > - at item
> > -Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an
> > M4A (MP4)
> > -container:
> > - at example
> > -ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
> > - at end example
> > -
> > - at item
> > -Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
> > -LTP AAC profile:
> > - at example
> > -ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
> > - at end example
> > - at end itemize
> > -
> >  @anchor{libfdk-aac-enc}
> >  @section libfdk_aac
> >
> > diff --git a/doc/ffserver.conf b/doc/ffserver.conf
> > index 7a30fb6..e3f99bb 100644
> > --- a/doc/ffserver.conf
> > +++ b/doc/ffserver.conf
> > @@ -317,7 +317,7 @@ StartSendOnKey
> >  #AVPresetVideo baseline
> >  #AVOptionVideo flags +global_header
> >  #
> > -#AudioCodec libfaac
> > +#AudioCodec aac
> >  #AudioBitRate 32
> >  #AudioChannels 2
> >  #AudioSampleRate 22050
> > diff --git a/doc/general.texi b/doc/general.texi
> > index 59ea4f4..fbcccdc 100644
> > --- a/doc/general.texi
> > +++ b/doc/general.texi
> > @@ -875,7 +875,7 @@ following image formats are supported:
> >  @item 8SVX exponential       @tab     @tab  X
> >  @item 8SVX fibonacci         @tab     @tab  X
> >  @item AAC                    @tab EX  @tab  X
> > -    @tab encoding supported through internal encoder and external
> > libraries libfaac and libfdk-aac
> > +    @tab encoding supported through internal encoder and external
> library
> > libfdk-aac
> >  @item AAC+                   @tab  E  @tab  IX
> >      @tab encoding supported through external library libfdk-aac
> >  @item AC-3                   @tab IX  @tab  IX
> > diff --git a/doc/muxers.texi b/doc/muxers.texi
> > index 2aafbad..839909d 100644
> > --- a/doc/muxers.texi
> > +++ b/doc/muxers.texi
> > @@ -1290,9 +1290,9 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment
> > -segment_list out.csv -segment_fr
> >
> >  @item
> >  Convert the @file{in.mkv} to TS segments using the @code{libx264}
> > -and @code{libfaac} encoders:
> > +and internal AAC encoders:
> >  @example
> > -ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment
> > -segment_list out.list out%03d.ts
> > +ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment
> > -segment_list out.list out%03d.ts
> >  @end example
> >
> >  @item
> > diff --git a/doc/platform.texi b/doc/platform.texi
> > index f7ee456..65f3cb5 100644
> > --- a/doc/platform.texi
> > +++ b/doc/platform.texi
> > @@ -314,7 +314,7 @@ These library packages are only available from
> >  @uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
> >
> >  @example
> > -yasm, libSDL-devel, libfaac-devel, libgsm-devel, libmp3lame-devel,
> > +yasm, libSDL-devel, libgsm-devel, libmp3lame-devel,
> >  libschroedinger1.0-devel, speex-devel, libtheora-devel,
> libxvidcore-devel
> >  @end example
> >
> > diff --git a/libavcodec/Makefile b/libavcodec/Makefile
> > index d4faf26..9b447fe 100644
> > --- a/libavcodec/Makefile
> > +++ b/libavcodec/Makefile
> > @@ -825,7 +825,6 @@ OBJS-$(CONFIG_ILBC_AT_ENCODER)            +=
> > audiotoolboxenc.o
> >  OBJS-$(CONFIG_PCM_ALAW_AT_ENCODER)        += audiotoolboxenc.o
> >  OBJS-$(CONFIG_PCM_MULAW_AT_ENCODER)       += audiotoolboxenc.o
> >  OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
> > -OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
> >  OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
> >  OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
> >  OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsmdec.o
> > diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
> > index 44e7c3f..e4c8d37 100644
> > --- a/libavcodec/allcodecs.c
> > +++ b/libavcodec/allcodecs.c
> > @@ -581,7 +581,6 @@ void avcodec_register_all(void)
> >      REGISTER_DECODER(QDMC_AT,           qdmc_at);
> >      REGISTER_DECODER(QDM2_AT,           qdm2_at);
> >      REGISTER_DECODER(LIBCELT,           libcelt);
> > -    REGISTER_ENCODER(LIBFAAC,           libfaac);
> >      REGISTER_ENCDEC (LIBFDK_AAC,        libfdk_aac);
> >      REGISTER_ENCDEC (LIBGSM,            libgsm);
> >      REGISTER_ENCDEC (LIBGSM_MS,         libgsm_ms);
> > diff --git a/libavcodec/libfaac.c b/libavcodec/libfaac.c
> > deleted file mode 100644
> > index 98b3ba8..0000000
> > --- a/libavcodec/libfaac.c
> > +++ /dev/null
> > @@ -1,248 +0,0 @@
> > -/*
> > - * Interface to libfaac for aac encoding
> > - * Copyright (c) 2002 Gildas Bazin <gbazin at netcourrier.com>
> > - *
> > - * 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
> > - * Interface to libfaac for aac encoding.
> > - */
> > -
> > -#include <faac.h>
> > -
> > -#include "libavutil/channel_layout.h"
> > -#include "libavutil/common.h"
> > -#include "avcodec.h"
> > -#include "audio_frame_queue.h"
> > -#include "internal.h"
> > -
> > -
> > -/* libfaac has an encoder delay of 1024 samples */
> > -#define FAAC_DELAY_SAMPLES 1024
> > -
> > -typedef struct FaacAudioContext {
> > -    faacEncHandle faac_handle;
> > -    AudioFrameQueue afq;
> > -} FaacAudioContext;
> > -
> > -static av_cold int Faac_encode_close(AVCodecContext *avctx)
> > -{
> > -    FaacAudioContext *s = avctx->priv_data;
> > -
> > -    av_freep(&avctx->extradata);
> > -    ff_af_queue_close(&s->afq);
> > -
> > -    if (s->faac_handle)
> > -        faacEncClose(s->faac_handle);
> > -
> > -    return 0;
> > -}
> > -
> > -static const int channel_maps[][6] = {
> > -    { 2, 0, 1 },          //< C L R
> > -    { 2, 0, 1, 3 },       //< C L R Cs
> > -    { 2, 0, 1, 3, 4 },    //< C L R Ls Rs
> > -    { 2, 0, 1, 4, 5, 3 }, //< C L R Ls Rs LFE
> > -};
> > -
> > -static av_cold int Faac_encode_init(AVCodecContext *avctx)
> > -{
> > -    FaacAudioContext *s = avctx->priv_data;
> > -    faacEncConfigurationPtr faac_cfg;
> > -    unsigned long samples_input, max_bytes_output;
> > -    int ret;
> > -
> > -    /* number of channels */
> > -    if (avctx->channels < 1 || avctx->channels > 6) {
> > -        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not
> > allowed\n", avctx->channels);
> > -        ret = AVERROR(EINVAL);
> > -        goto error;
> > -    }
> > -
> > -    s->faac_handle = faacEncOpen(avctx->sample_rate,
> > -                                 avctx->channels,
> > -                                 &samples_input, &max_bytes_output);
> > -    if (!s->faac_handle) {
> > -        av_log(avctx, AV_LOG_ERROR, "error in faacEncOpen()\n");
> > -        ret = AVERROR_UNKNOWN;
> > -        goto error;
> > -    }
> > -
> > -    /* check faac version */
> > -    faac_cfg = faacEncGetCurrentConfiguration(s->faac_handle);
> > -    if (faac_cfg->version != FAAC_CFG_VERSION) {
> > -        av_log(avctx, AV_LOG_ERROR, "wrong libfaac version (compiled
> for:
> > %d, using %d)\n", FAAC_CFG_VERSION, faac_cfg->version);
> > -        ret = AVERROR(EINVAL);
> > -        goto error;
> > -    }
> > -
> > -    /* put the options in the configuration struct */
> > -    switch(avctx->profile) {
> > -        case FF_PROFILE_AAC_MAIN:
> > -            faac_cfg->aacObjectType = MAIN;
> > -            break;
> > -        case FF_PROFILE_UNKNOWN:
> > -        case FF_PROFILE_AAC_LOW:
> > -            faac_cfg->aacObjectType = LOW;
> > -            break;
> > -        case FF_PROFILE_AAC_SSR:
> > -            faac_cfg->aacObjectType = SSR;
> > -            break;
> > -        case FF_PROFILE_AAC_LTP:
> > -            faac_cfg->aacObjectType = LTP;
> > -            break;
> > -        default:
> > -            av_log(avctx, AV_LOG_ERROR, "invalid AAC profile\n");
> > -            ret = AVERROR(EINVAL);
> > -            goto error;
> > -    }
> > -    faac_cfg->mpegVersion = MPEG4;
> > -    faac_cfg->useTns = 0;
> > -    faac_cfg->allowMidside = 1;
> > -    faac_cfg->bitRate = avctx->bit_rate / avctx->channels;
> > -    faac_cfg->bandWidth = avctx->cutoff;
> > -    if(avctx->flags & AV_CODEC_FLAG_QSCALE) {
> > -        faac_cfg->bitRate = 0;
> > -        faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA;
> > -    }
> > -    faac_cfg->outputFormat = 1;
> > -    faac_cfg->inputFormat = FAAC_INPUT_16BIT;
> > -    if (avctx->channels > 2)
> > -        memcpy(faac_cfg->channel_map, channel_maps[avctx->channels-3],
> > -               avctx->channels * sizeof(int));
> > -
> > -    avctx->frame_size = samples_input / avctx->channels;
> > -
> > -    /* Set decoder specific info */
> > -    avctx->extradata_size = 0;
> > -    if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) {
> > -
> > -        unsigned char *buffer = NULL;
> > -        unsigned long decoder_specific_info_size;
> > -
> > -        if (!faacEncGetDecoderSpecificInfo(s->faac_handle, &buffer,
> > -
>  &decoder_specific_info_size)) {
> > -            avctx->extradata = av_malloc(decoder_specific_info_size +
> > AV_INPUT_BUFFER_PADDING_SIZE);
> > -            if (!avctx->extradata) {
> > -                ret = AVERROR(ENOMEM);
> > -                goto error;
> > -            }
> > -            avctx->extradata_size = decoder_specific_info_size;
> > -            memcpy(avctx->extradata, buffer, avctx->extradata_size);
> > -            faac_cfg->outputFormat = 0;
> > -        }
> > -        free(buffer);
> > -    }
> > -
> > -    if (!faacEncSetConfiguration(s->faac_handle, faac_cfg)) {
> > -        int i;
> > -        for (i = avctx->bit_rate/1000; i ; i--) {
> > -            faac_cfg->bitRate = 1000*i / avctx->channels;
> > -            if (faacEncSetConfiguration(s->faac_handle, faac_cfg))
> > -                break;
> > -        }
> > -        if (!i) {
> > -            av_log(avctx, AV_LOG_ERROR, "libfaac doesn't support this
> > output format!\n");
> > -            ret = AVERROR(EINVAL);
> > -            goto error;
> > -        } else {
> > -            avctx->bit_rate = 1000*i;
> > -            av_log(avctx, AV_LOG_WARNING, "libfaac doesn't support the
> > specified bitrate, using %dkbit/s instead\n", i);
> > -        }
> > -    }
> > -
> > -    avctx->initial_padding = FAAC_DELAY_SAMPLES;
> > -    ff_af_queue_init(avctx, &s->afq);
> > -
> > -    return 0;
> > -error:
> > -    Faac_encode_close(avctx);
> > -    return ret;
> > -}
> > -
> > -static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
> > -                             const AVFrame *frame, int *got_packet_ptr)
> > -{
> > -    FaacAudioContext *s = avctx->priv_data;
> > -    int bytes_written, ret;
> > -    int num_samples  = frame ? frame->nb_samples : 0;
> > -    void *samples    = frame ? frame->data[0]    : NULL;
> > -
> > -    if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) *
> > avctx->channels, 0)) < 0)
> > -        return ret;
> > -
> > -    bytes_written = faacEncEncode(s->faac_handle, samples,
> > -                                  num_samples * avctx->channels,
> > -                                  avpkt->data, avpkt->size);
> > -    if (bytes_written < 0) {
> > -        av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
> > -        return bytes_written;
> > -    }
> > -
> > -    /* add current frame to the queue */
> > -    if (frame) {
> > -        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
> > -            return ret;
> > -    }
> > -
> > -    if (!bytes_written)
> > -        return 0;
> > -
> > -    /* Get the next frame pts/duration */
> > -    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
> > -                       &avpkt->duration);
> > -
> > -    avpkt->size = bytes_written;
> > -    *got_packet_ptr = 1;
> > -    return 0;
> > -}
> > -
> > -static const AVProfile profiles[] = {
> > -    { FF_PROFILE_AAC_MAIN, "Main" },
> > -    { FF_PROFILE_AAC_LOW,  "LC"   },
> > -    { FF_PROFILE_AAC_SSR,  "SSR"  },
> > -    { FF_PROFILE_AAC_LTP,  "LTP"  },
> > -    { FF_PROFILE_UNKNOWN },
> > -};
> > -
> > -static const uint64_t faac_channel_layouts[] = {
> > -    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_libfaac_encoder = {
> > -    .name           = "libfaac",
> > -    .long_name      = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio
> > Coding)"),
> > -    .type           = AVMEDIA_TYPE_AUDIO,
> > -    .id             = AV_CODEC_ID_AAC,
> > -    .priv_data_size = sizeof(FaacAudioContext),
> > -    .init           = Faac_encode_init,
> > -    .encode2        = Faac_encode_frame,
> > -    .close          = Faac_encode_close,
> > -    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME |
> AV_CODEC_CAP_DELAY,
> > -    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
> > -                                                     AV_SAMPLE_FMT_NONE
> },
> > -    .profiles       = NULL_IF_CONFIG_SMALL(profiles),
> > -    .channel_layouts = faac_channel_layouts,
> > -};
> > diff --git a/libavcodec/version.h b/libavcodec/version.h
> > index 1438e2e..c0f336b 100644
> > --- a/libavcodec/version.h
> > +++ b/libavcodec/version.h
> > @@ -28,7 +28,7 @@
> >  #include "libavutil/version.h"
> >
> >  #define LIBAVCODEC_VERSION_MAJOR  57
> > -#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,
> > \
> > --
> > 1.9.1
> >
> > _______________________________________________
> > ffmpeg-devel mailing list
> > ffmpeg-devel at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> >
>
>
> Now that the AAC encoder is faster and still better than libfaac with the
> fast coder, can we push this patch?
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>

Wouldn't it be better to have a grace period for this, so ffmpeg/libavcodec
users who have faac in their workflows (probably very many out there) have
some time to evaluate and change their workflows (after all, this will
require investigation, parameter tweaking, lots of testing etc.) without
losing the ability of using ffmpeg head or having to patch manually? It is
quite a harsh step to just remove support for this without a longer
deprecation period especially for such a popular feature.


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