This method first sets quantizers depending on band thresholds and then tries
to find an optimal combination by adding or subtracting a specific value from
-all quantizers and adjusting some individual quantizer a little.
-Will tune itself based on whether aac_is/aac_ms/aac_pns are enabled.
+all quantizers and adjusting some individual quantizer a little. Will tune
+itself based on whether @option{aac_is}, @option{aac_ms} and @option{aac_pns}
+are enabled.
This is the default choice for a coder.
@item anmr
@end table
@item aac_ms
-Sets mid/side coding mode. The default value of auto will automatically use
+Sets mid/side coding mode. The default value of "auto" will automatically use
M/S with bands which will benefit from such coding. Can be forced for all bands
using the value "enable", which is mainly useful for debugging or disabled using
"disable".
decent quality.
@item mpeg2_aac_low
-Equivalent to -profile:a aac_low -aac_pns 0. PNS was introduced with the MPEG4
-specifications.
+Equivalent to @code{-profile:a aac_low -aac_pns 0}. PNS was introduced with the
+MPEG4 specifications.
@item aac_ltp
-Long term prediction profile, is enabled by and will enable the aac_ltp option.
-Introduced in MPEG4.
+Long term prediction profile, is enabled by and will enable the @option{aac_ltp}
+option. Introduced in MPEG4.
@item aac_main
-Main-type prediction profile, is enabled by and will enable the aac_pred option.
-Introduced in MPEG2.
+Main-type prediction profile, is enabled by and will enable the @option{aac_pred}
+option. Introduced in MPEG2.
-If this option is unspecified it is set to @samp{aac_low}.
@end table
+If this option is unspecified it is set to @samp{aac_low}.
@end table
@section ac3 and ac3_fixed
@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
narrowband, mediumband, wideband, super wideband, and fullband
respectively. The default is 0 (cutoff disabled).
+@item mapping_family (@emph{mapping_family})
+Set channel mapping family to be used by the encoder. The default value of -1
+uses mapping family 0 for mono and stereo inputs, and mapping family 1
+otherwise. The default also disables the surround masking and LFE bandwidth
+optimzations in libopus, and requires that the input contains 8 channels or
+fewer.
+
+Other values include 0 for mono and stereo, 1 for surround sound with masking
+and LFE bandwidth optimizations, and 255 for independent streams with an
+unspecified channel layout.
+
@end table
@section libvorbis
@samp{fixed}.
@item slice_mode
-Set slice mode. Can assume one of the follwing possible values:
+Set slice mode. Can assume one of the following possible values:
@table @samp
@item fixed
projects / ffmpeg / blobdiff