On 10 April 2016 at 16:38, Kieran Kunhya <kie...@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
>
> -@anchor{libfaac}
> -@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
> -@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
> -@code{--enable-libfaac --enable-nonfree}.
> -
> -@subsection Options
> -
> -The following shared FFmpeg codec options are recognized.
> -
> -The following options are supported by the libfaac wrapper. The
> -@command{faac}-equivalent of the options are listed in parentheses.
> -
> -@table @option
> -@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.
> -
> -@item ar (@emph{-R})
> -Set audio sampling rate (in Hz).
> -
> -@item ac (@emph{-c})
> -Set the number of audio channels.
> -
> -@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.
> -
> -@item profile
> -Set audio profile.
> -
> -The following profiles are recognized:
> -@table @samp
> -@item aac_main
> -Main AAC (Main)
> -
> -@item aac_low
> -Low Complexity AAC (LC)
> -
> -@item aac_ssr
> -Scalable Sample Rate (SSR)
> -
> -@item aac_ltp
> -Long Term Prediction (LTP)
> -@end table
> -
> -If not specified it is set to @samp{aac_low}.
> -
> -@item flags +qscale
> -Set constant quality VBR (Variable Bit Rate) mode.
> -
> -@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.
> -
> -@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}.
> -@end table
> -
> -@subsection Examples
> -
> -@itemize
> -@item
> -Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an
> M4A (MP4)
> -container:
> -@example
> -ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
> -@end example
> -
> -@item
> -Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
> -LTP AAC profile:
> -@example
> -ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
> -@end example
> -@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 <gba...@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@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?
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