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>

5 files changed, 5 insertions, 110 deletions

diff --git a/doc/encoders.texi b/doc/encoders.texi
index 73ebd9cd0b..1f4044ef0f 100644
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -612,111 +612,6 @@ 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 7a30fb6c3b..e3f99bbfb3 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 a0f1122d67..f08c3cd66c 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -887,7 +887,7 @@ 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 2c937c76aa..9ec2e31a5c 100644
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -1437,9 +1437,9 @@ 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 @code{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 21b135f55d..4729d41797 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