mirror of
https://github.com/FFmpeg/FFmpeg.git
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lavc: remove libfaac wrapper
There is really no need for two aac wrappers, we already have libfdk-aac which is better. Not to mention that faac doesn't even support HEv1, or HEv2. It's also under a license which is unusable for distribution, so it would only be useful to people who will compile their own ffmpeg, only use it themselves (which at that point should just use fdk-aac). Signed-off-by: Josh de Kock <josh@itanimul.li>
This commit is contained in:
Josh de Kock
Rostislav Pehlivanov
parent
449f263f9f
commit
dc0f711459
@ -36,6 +36,7 @@ version <next>:
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- sdl2 support for ffplay
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- sdl1 output device and sdl1 support removed
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- extended mov edit list support
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- libfaac encoder removed
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version 3.1:
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@ -115,8 +115,6 @@ The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
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incompatible with the GPLv2 and v3. To the best of our knowledge, they are
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compatible with the LGPL.
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The FAAC library is incompatible with all versions of GPL and LGPL.
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The NVENC library, while its header file is licensed under the compatible MIT
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license, requires a proprietary binary blob at run time, and is deemed to be
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incompatible with the GPL. We are not certain if it is compatible with the
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6
configure
vendored
6
configure
vendored
@ -225,7 +225,6 @@ External library support:
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and libraw1394 [no]
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--enable-libebur128 enable libebur128 for EBU R128 measurement,
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needed for loudnorm filter [no]
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--enable-libfaac enable AAC encoding via libfaac [no]
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--enable-libfdk-aac enable AAC de/encoding via libfdk-aac [no]
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--enable-libflite enable flite (voice synthesis) support via libflite [no]
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--enable-libfontconfig enable libfontconfig, useful for drawtext filter [no]
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@ -1488,7 +1487,6 @@ EXTERNAL_LIBRARY_LIST="
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libcelt
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libdc1394
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libebur128
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libfaac
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libfdk_aac
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libflite
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libfontconfig
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@ -2773,8 +2771,6 @@ pcm_mulaw_at_encoder_select="audio_frame_queue"
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chromaprint_muxer_deps="chromaprint"
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h264_videotoolbox_encoder_deps="videotoolbox_encoder pthreads"
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libcelt_decoder_deps="libcelt"
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libfaac_encoder_deps="libfaac"
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libfaac_encoder_select="audio_frame_queue"
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libfdk_aac_decoder_deps="libfdk_aac"
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libfdk_aac_encoder_deps="libfdk_aac"
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libfdk_aac_encoder_select="audio_frame_queue"
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@ -5072,7 +5068,6 @@ die_license_disabled gpl x11grab
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die_license_disabled nonfree cuda
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die_license_disabled nonfree cuvid
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die_license_disabled nonfree libfaac
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die_license_disabled nonfree libnpp
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enabled gpl && die_license_disabled_gpl nonfree libfdk_aac
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enabled gpl && die_license_disabled_gpl nonfree openssl
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@ -5682,7 +5677,6 @@ enabled libcelt && require libcelt celt/celt.h celt_decode -lcelt0 &&
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die "ERROR: libcelt must be installed and version must be >= 0.11.0."; }
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enabled libcaca && require_pkg_config caca caca.h caca_create_canvas
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enabled libebur128 && require ebur128 ebur128.h ebur128_relative_threshold -lebur128
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enabled libfaac && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac
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enabled libfdk_aac && { use_pkg_config fdk-aac "fdk-aac/aacenc_lib.h" aacEncOpen ||
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{ require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac &&
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warn "using libfdk without pkg-config"; } }
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@ -612,111 +612,6 @@ and slightly improves compression.
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@end table
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@anchor{libfaac}
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@section libfaac
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libfaac AAC (Advanced Audio Coding) encoder wrapper.
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This encoder is of much lower quality and is more unstable than any other AAC
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encoders, so it's highly recommended to instead use other encoders, like
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@ref{aacenc,,the native FFmpeg AAC encoder}.
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This encoder also requires the presence of the libfaac headers and library
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during configuration. You need to explicitly configure the build with
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@code{--enable-libfaac --enable-nonfree}.
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@subsection Options
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The following shared FFmpeg codec options are recognized.
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The following options are supported by the libfaac wrapper. The
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@command{faac}-equivalent of the options are listed in parentheses.
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@table @option
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@item b (@emph{-b})
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Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
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is not explicitly specified, it is automatically set to a suitable
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value depending on the selected profile. @command{faac} bitrate is
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expressed in kilobits/s.
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Note that libfaac does not support CBR (Constant Bit Rate) but only
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ABR (Average Bit Rate).
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If VBR mode is enabled this option is ignored.
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@item ar (@emph{-R})
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Set audio sampling rate (in Hz).
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@item ac (@emph{-c})
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Set the number of audio channels.
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@item cutoff (@emph{-C})
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Set cutoff frequency. If not specified (or explicitly set to 0) it
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will use a value automatically computed by the library. Default value
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is 0.
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@item profile
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Set audio profile.
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The following profiles are recognized:
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@table @samp
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@item aac_main
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Main AAC (Main)
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@item aac_low
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Low Complexity AAC (LC)
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@item aac_ssr
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Scalable Sample Rate (SSR)
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@item aac_ltp
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Long Term Prediction (LTP)
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@end table
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If not specified it is set to @samp{aac_low}.
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@item flags +qscale
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Set constant quality VBR (Variable Bit Rate) mode.
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@item global_quality
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Set quality in VBR mode as an integer number of lambda units.
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Only relevant when VBR mode is enabled with @code{flags +qscale}. The
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value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
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and used to set the quality value used by libfaac. A reasonable range
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for the option value in QP units is [10-500], the higher the value the
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higher the quality.
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@item q (@emph{-q})
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Enable VBR mode when set to a non-negative value, and set constant
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quality value as a double floating point value in QP units.
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The value sets the quality value used by libfaac. A reasonable range
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for the option value is [10-500], the higher the value the higher the
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quality.
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This option is valid only using the @command{ffmpeg} command-line
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tool. For library interface users, use @option{global_quality}.
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@end table
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@subsection Examples
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@itemize
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@item
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Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
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container:
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@example
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ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
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@end example
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@item
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Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
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LTP AAC profile:
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@example
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ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
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@end example
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@end itemize
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@anchor{libfdk-aac-enc}
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@section libfdk_aac
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@ -317,7 +317,7 @@ StartSendOnKey
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#AVPresetVideo baseline
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#AVOptionVideo flags +global_header
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#
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#AudioCodec libfaac
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#AudioCodec aac
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#AudioBitRate 32
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#AudioChannels 2
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#AudioSampleRate 22050
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@ -887,7 +887,7 @@ following image formats are supported:
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@item 8SVX exponential @tab @tab X
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@item 8SVX fibonacci @tab @tab X
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@item AAC @tab EX @tab X
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@tab encoding supported through internal encoder and external libraries libfaac and libfdk-aac
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@tab encoding supported through internal encoder and external library libfdk-aac
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@item AAC+ @tab E @tab IX
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@tab encoding supported through external library libfdk-aac
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@item AC-3 @tab IX @tab IX
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@ -1437,9 +1437,9 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_fr
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@item
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Convert the @file{in.mkv} to TS segments using the @code{libx264}
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and @code{libfaac} encoders:
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and @code{aac} encoders:
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@example
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ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
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ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts
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@end example
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@item
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@ -314,7 +314,7 @@ These library packages are only available from
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@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
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@example
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yasm, libSDL-devel, libfaac-devel, libgsm-devel, libmp3lame-devel,
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yasm, libSDL-devel, libgsm-devel, libmp3lame-devel,
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libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
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@end example
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@ -862,7 +862,6 @@ OBJS-$(CONFIG_ILBC_AT_ENCODER) += audiotoolboxenc.o
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OBJS-$(CONFIG_PCM_ALAW_AT_ENCODER) += audiotoolboxenc.o
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OBJS-$(CONFIG_PCM_MULAW_AT_ENCODER) += audiotoolboxenc.o
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OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
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OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
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OBJS-$(CONFIG_LIBFDK_AAC_DECODER) += libfdk-aacdec.o
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OBJS-$(CONFIG_LIBFDK_AAC_ENCODER) += libfdk-aacenc.o
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OBJS-$(CONFIG_LIBGSM_DECODER) += libgsmdec.o
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@ -594,7 +594,6 @@ void avcodec_register_all(void)
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REGISTER_DECODER(QDMC_AT, qdmc_at);
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REGISTER_DECODER(QDM2_AT, qdm2_at);
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REGISTER_DECODER(LIBCELT, libcelt);
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REGISTER_ENCODER(LIBFAAC, libfaac);
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REGISTER_ENCDEC (LIBFDK_AAC, libfdk_aac);
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REGISTER_ENCDEC (LIBGSM, libgsm);
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REGISTER_ENCDEC (LIBGSM_MS, libgsm_ms);
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@ -1,248 +0,0 @@
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/*
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* Interface to libfaac for aac encoding
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* Copyright (c) 2002 Gildas Bazin <gbazin@netcourrier.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Interface to libfaac for aac encoding.
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*/
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#include <faac.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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/* libfaac has an encoder delay of 1024 samples */
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#define FAAC_DELAY_SAMPLES 1024
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typedef struct FaacAudioContext {
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faacEncHandle faac_handle;
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AudioFrameQueue afq;
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} FaacAudioContext;
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static av_cold int Faac_encode_close(AVCodecContext *avctx)
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{
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FaacAudioContext *s = avctx->priv_data;
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av_freep(&avctx->extradata);
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ff_af_queue_close(&s->afq);
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if (s->faac_handle)
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faacEncClose(s->faac_handle);
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return 0;
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}
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static const int channel_maps[][6] = {
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{ 2, 0, 1 }, //< C L R
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{ 2, 0, 1, 3 }, //< C L R Cs
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{ 2, 0, 1, 3, 4 }, //< C L R Ls Rs
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{ 2, 0, 1, 4, 5, 3 }, //< C L R Ls Rs LFE
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};
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static av_cold int Faac_encode_init(AVCodecContext *avctx)
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{
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FaacAudioContext *s = avctx->priv_data;
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faacEncConfigurationPtr faac_cfg;
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unsigned long samples_input, max_bytes_output;
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int ret;
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/* number of channels */
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if (avctx->channels < 1 || avctx->channels > 6) {
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av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
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ret = AVERROR(EINVAL);
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goto error;
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}
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s->faac_handle = faacEncOpen(avctx->sample_rate,
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avctx->channels,
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&samples_input, &max_bytes_output);
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if (!s->faac_handle) {
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av_log(avctx, AV_LOG_ERROR, "error in faacEncOpen()\n");
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ret = AVERROR_UNKNOWN;
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goto error;
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}
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/* check faac version */
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faac_cfg = faacEncGetCurrentConfiguration(s->faac_handle);
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if (faac_cfg->version != FAAC_CFG_VERSION) {
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av_log(avctx, AV_LOG_ERROR, "wrong libfaac version (compiled for: %d, using %d)\n", FAAC_CFG_VERSION, faac_cfg->version);
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ret = AVERROR(EINVAL);
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goto error;
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}
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/* put the options in the configuration struct */
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switch(avctx->profile) {
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case FF_PROFILE_AAC_MAIN:
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faac_cfg->aacObjectType = MAIN;
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break;
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case FF_PROFILE_UNKNOWN:
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case FF_PROFILE_AAC_LOW:
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faac_cfg->aacObjectType = LOW;
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break;
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case FF_PROFILE_AAC_SSR:
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faac_cfg->aacObjectType = SSR;
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break;
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case FF_PROFILE_AAC_LTP:
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faac_cfg->aacObjectType = LTP;
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break;
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default:
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av_log(avctx, AV_LOG_ERROR, "invalid AAC profile\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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faac_cfg->mpegVersion = MPEG4;
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faac_cfg->useTns = 0;
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faac_cfg->allowMidside = 1;
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faac_cfg->bitRate = avctx->bit_rate / avctx->channels;
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faac_cfg->bandWidth = avctx->cutoff;
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if(avctx->flags & AV_CODEC_FLAG_QSCALE) {
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faac_cfg->bitRate = 0;
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faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA;
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}
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faac_cfg->outputFormat = 1;
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faac_cfg->inputFormat = FAAC_INPUT_16BIT;
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if (avctx->channels > 2)
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memcpy(faac_cfg->channel_map, channel_maps[avctx->channels-3],
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avctx->channels * sizeof(int));
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avctx->frame_size = samples_input / avctx->channels;
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/* Set decoder specific info */
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avctx->extradata_size = 0;
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if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) {
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|
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unsigned char *buffer = NULL;
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unsigned long decoder_specific_info_size;
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|
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if (!faacEncGetDecoderSpecificInfo(s->faac_handle, &buffer,
|
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&decoder_specific_info_size)) {
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avctx->extradata = av_malloc(decoder_specific_info_size + AV_INPUT_BUFFER_PADDING_SIZE);
|
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if (!avctx->extradata) {
|
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ret = AVERROR(ENOMEM);
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goto error;
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}
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avctx->extradata_size = decoder_specific_info_size;
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memcpy(avctx->extradata, buffer, avctx->extradata_size);
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faac_cfg->outputFormat = 0;
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}
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free(buffer);
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}
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|
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if (!faacEncSetConfiguration(s->faac_handle, faac_cfg)) {
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int i;
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for (i = avctx->bit_rate/1000; i ; i--) {
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faac_cfg->bitRate = 1000*i / avctx->channels;
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if (faacEncSetConfiguration(s->faac_handle, faac_cfg))
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break;
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}
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if (!i) {
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av_log(avctx, AV_LOG_ERROR, "libfaac doesn't support this output format!\n");
|
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ret = AVERROR(EINVAL);
|
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goto error;
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} else {
|
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avctx->bit_rate = 1000*i;
|
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av_log(avctx, AV_LOG_WARNING, "libfaac doesn't support the specified bitrate, using %dkbit/s instead\n", i);
|
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}
|
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}
|
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|
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avctx->initial_padding = FAAC_DELAY_SAMPLES;
|
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ff_af_queue_init(avctx, &s->afq);
|
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|
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return 0;
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error:
|
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Faac_encode_close(avctx);
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return ret;
|
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}
|
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|
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static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
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const AVFrame *frame, int *got_packet_ptr)
|
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{
|
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FaacAudioContext *s = avctx->priv_data;
|
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int bytes_written, ret;
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int num_samples = frame ? frame->nb_samples : 0;
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void *samples = frame ? frame->data[0] : NULL;
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if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels, 0)) < 0)
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return ret;
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bytes_written = faacEncEncode(s->faac_handle, samples,
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num_samples * avctx->channels,
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avpkt->data, avpkt->size);
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||||
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,
|
||||
};
|
||||
Loading…
Reference in New Issue
Block a user