encoders.texi

@chapter Encoders
@c man begin ENCODERS

Encoders are configured elements in FFmpeg which allow the encoding of
multimedia streams.

When you configure your FFmpeg build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available encoders using the configure option @code{--list-encoders}.

You can disable all the encoders with the configure option
@code{--disable-encoders} and selectively enable / disable single encoders
with the options @code{--enable-encoder=@var{ENCODER}} /
@code{--disable-encoder=@var{ENCODER}}.

The option @code{-encoders} of the ff* tools will display the list of
enabled encoders.

@c man end ENCODERS

@chapter Audio Encoders
@c man begin AUDIO ENCODERS

A description of some of the currently available audio encoders
follows.

@anchor{aacenc}
@section aac

Advanced Audio Coding (AAC) encoder.

This encoder is an experimental FFmpeg-native AAC encoder. Currently only the
low complexity (AAC-LC) profile is supported. To use this encoder, you must set
@option{strict} option to @samp{experimental} or lower.

As this encoder is experimental, unexpected behavior may exist from time to
time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
that it has a worse quality reported by some users.

@c todo @ref{libaacplus}
See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.

@subsection Options

@table @option
@item b
Set bit rate in bits/s. Setting this automatically activates constant bit rate
(CBR) mode.

@item q
Set quality for variable bit rate (VBR) mode. This option is valid only using
the @command{ffmpeg} command-line tool. For library interface users, use
@option{global_quality}.

@item stereo_mode
Set stereo encoding mode. Possible values:

@table @samp
@item auto
Automatically selected by the encoder.

@item ms_off
Disable middle/side encoding. This is the default.

@item ms_force
Force middle/side encoding.
@end table

@item aac_coder
Set AAC encoder coding method. Possible values:

@table @samp
@item faac
FAAC-inspired method.

This method is a simplified reimplementation of the method used in FAAC, which
sets thresholds proportional to the band energies, and then decreases all the
thresholds with quantizer steps to find the appropriate quantization with
distortion below threshold band by band.

The quality of this method is comparable to the two loop searching method
described below, but somewhat a little better and slower.

@item anmr
Average noise to mask ratio (ANMR) trellis-based solution.

This has a theoretic best quality out of all the coding methods, but at the
cost of the slowest speed.

@item twoloop
Two loop searching (TLS) method.

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.

This method produces similar quality with the FAAC method and is the default.

@item fast
Constant quantizer method.

This method sets a constant quantizer for all bands. This is the fastest of all
the methods, yet produces the worst quality.

@end table

@end table

@section ac3 and ac3_fixed

AC-3 audio encoders.

These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).

The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed}
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option @code{-acodec ac3_fixed} in order to use it.

@subsection AC-3 Metadata

The AC-3 metadata options are used to set parameters that describe the audio,
but in most cases do not affect the audio encoding itself. Some of the options
do directly affect or influence the decoding and playback of the resulting
bitstream, while others are just for informational purposes. A few of the
options will add bits to the output stream that could otherwise be used for
audio data, and will thus affect the quality of the output. Those will be
indicated accordingly with a note in the option list below.

These parameters are described in detail in several publicly-available
documents.
@itemize
@item @uref{http://www.atsc.org/cms/standards/a_52-2010.pdf,A/52:2010 - Digital Audio Compression (AC-3) (E-AC-3) Standard}
@item @uref{http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf,A/54 - Guide to the Use of the ATSC Digital Television Standard}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf,Dolby Metadata Guide}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf,Dolby Digital Professional Encoding Guidelines}
@end itemize

@subsubsection Metadata Control Options

@table @option

@item -per_frame_metadata @var{boolean}
Allow Per-Frame Metadata. Specifies if the encoder should check for changing
metadata for each frame.
@table @option
@item 0
The metadata values set at initialization will be used for every frame in the
stream. (default)
@item 1
Metadata values can be changed before encoding each frame.
@end table

@end table

@subsubsection Downmix Levels

@table @option

@item -center_mixlev @var{level}
Center Mix Level. The amount of gain the decoder should apply to the center
channel when downmixing to stereo. This field will only be written to the
bitstream if a center channel is present. The value is specified as a scale
factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6dB gain
@end table

@item -surround_mixlev @var{level}
Surround Mix Level. The amount of gain the decoder should apply to the surround
channel(s) when downmixing to stereo. This field will only be written to the
bitstream if one or more surround channels are present. The value is specified
as a scale factor.  There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.500
Apply -6dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table

@end table

@subsubsection Audio Production Information
Audio Production Information is optional information describing the mixing
environment.  Either none or both of the fields are written to the bitstream.

@table @option

@item -mixing_level @var{number}
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the @code{room_type} option is not the default value, the @code{mixing_level}
option must not be -1.

@item -room_type @var{type}
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the @code{mixing_level}
option and the @code{room_type} option have the default values.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx large
Large Room
@item 2
@itemx small
Small Room
@end table

@end table

@subsubsection Other Metadata Options

@table @option

@item -copyright @var{boolean}
Copyright Indicator. Specifies whether a copyright exists for this audio.
@table @option
@item 0
@itemx off
No Copyright Exists (default)
@item 1
@itemx on
Copyright Exists
@end table

@item -dialnorm @var{value}
Dialogue Normalization. Indicates how far the average dialogue level of the
program is below digital 100% full scale (0 dBFS). This parameter determines a
level shift during audio reproduction that sets the average volume of the
dialogue to a preset level. The goal is to match volume level between program
sources. A value of -31dB will result in no volume level change, relative to
the source volume, during audio reproduction. Valid values are whole numbers in
the range -31 to -1, with -31 being the default.

@item -dsur_mode @var{mode}
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround
(Pro Logic). This field will only be written to the bitstream if the audio
stream is stereo. Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx off
Not Dolby Surround Encoded
@item 2
@itemx on
Dolby Surround Encoded
@end table

@item -original @var{boolean}
Original Bit Stream Indicator. Specifies whether this audio is from the
original source and not a copy.
@table @option
@item 0
@itemx off
Not Original Source
@item 1
@itemx on
Original Source (default)
@end table

@end table

@subsection Extended Bitstream Information
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream.  Default values are used for those that are written
but have not been specified.  If the mixing levels are written, the decoder
will use these values instead of the ones specified in the @code{center_mixlev}
and @code{surround_mixlev} options if it supports the Alternate Bit Stream
Syntax.

@subsubsection Extended Bitstream Information - Part 1

@table @option

@item -dmix_mode @var{mode}
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt
(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx ltrt
Lt/Rt Downmix Preferred
@item 2
@itemx loro
Lo/Ro Downmix Preferred
@end table

@item -ltrt_cmixlev @var{level}
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lt/Rt mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table

@item -ltrt_surmixlev @var{level}
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lt/Rt mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table

@item -loro_cmixlev @var{level}
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lo/Ro mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table

@item -loro_surmixlev @var{level}
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lo/Ro mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table

@end table

@subsubsection Extended Bitstream Information - Part 2

@table @option

@item -dsurex_mode @var{mode}
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX
(7.1 matrixed to 5.1). Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround EX processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Surround EX Off
@item 2
@itemx off
Dolby Surround EX On
@end table

@item -dheadphone_mode @var{mode}
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone
encoding (multi-channel matrixed to 2.0 for use with headphones). Using this
option does @b{NOT} mean the encoder will actually apply Dolby Headphone
processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Headphone Off
@item 2
@itemx off
Dolby Headphone On
@end table

@item -ad_conv_type @var{type}
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D
conversion.
@table @option
@item 0
@itemx standard
Standard A/D Converter (default)
@item 1
@itemx hdcd
HDCD A/D Converter
@end table

@end table

@subsection Other AC-3 Encoding Options

@table @option

@item -stereo_rematrixing @var{boolean}
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This
is an optional AC-3 feature that increases quality by selectively encoding
the left/right channels as mid/side. This option is enabled by default, and it
is highly recommended that it be left as enabled except for testing purposes.

@end table

@subsection Floating-Point-Only AC-3 Encoding Options

These options are only valid for the floating-point encoder and do not exist
for the fixed-point encoder due to the corresponding features not being
implemented in fixed-point.

@table @option

@item -channel_coupling @var{boolean}
Enables/Disables use of channel coupling, which is an optional AC-3 feature
that increases quality by combining high frequency information from multiple
channels into a single channel. The per-channel high frequency information is
sent with less accuracy in both the frequency and time domains. This allows
more bits to be used for lower frequencies while preserving enough information
to reconstruct the high frequencies. This option is enabled by default for the
floating-point encoder and should generally be left as enabled except for
testing purposes or to increase encoding speed.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@item 0
@itemx off
Disable Channel Coupling
@item 1
@itemx on
Enable Channel Coupling
@end table

@item -cpl_start_band @var{number}
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a
value higher than the bandwidth is used, it will be reduced to 1 less than the
coupling end band. If @var{auto} is used, the start band will be determined by
the encoder based on the bit rate, sample rate, and channel layout. This option
has no effect if channel coupling is disabled.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@end table

@end table

@anchor{flac}
@section flac

FLAC (Free Lossless Audio Codec) Encoder

@subsection Options

The following options are supported by FFmpeg's flac encoder.

@table @option
@item compression_level
Sets the compression level, which chooses defaults for many other options
if they are not set explicitly.

@item frame_size
Sets the size of the frames in samples per channel.

@item lpc_coeff_precision
Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the
default.

@item lpc_type
Sets the first stage LPC algorithm
@table @samp
@item none
LPC is not used

@item fixed
fixed LPC coefficients

@item levinson

@item cholesky
@end table

@item lpc_passes
Number of passes to use for Cholesky factorization during LPC analysis

@item min_partition_order
The minimum partition order

@item max_partition_order
The maximum partition order

@item prediction_order_method
@table @samp
@item estimation
@item 2level
@item 4level
@item 8level
@item search
Bruteforce search
@item log
@end table

@item ch_mode
Channel mode
@table @samp
@item auto
The mode is chosen automatically for each frame
@item indep
Chanels are independently coded
@item left_side
@item right_side
@item mid_side
@end table

@item exact_rice_parameters
Chooses if rice parameters are calculated exactly or approximately.
if set to 1 then they are chosen exactly, which slows the code down slightly and
improves compression slightly.

@item multi_dim_quant
Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is
applied after the first stage to finetune the coefficients. This is quite slow
and slightly improves compression.

@end table

@anchor{libfaac}
@section libfaac

libfaac AAC (Advanced Audio Coding) encoder wrapper.

Requires the presence of the libfaac headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libfaac --enable-nonfree}.

This encoder is considered to be of higher quality with respect to the
@ref{aacenc,,the native experimental FFmpeg AAC encoder}.

For more information see the libfaac project at
@url{http://www.audiocoding.com/faac.html/}.

@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

libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.

The libfdk-aac library is based on the Fraunhofer FDK AAC code from
the Android project.

Requires the presence of the libfdk-aac headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libfdk-aac}. The library is also incompatible with GPL,
so if you allow the use of GPL, you should configure with
@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.

This encoder is considered to be of higher quality with respect to
both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
@ref{libfaac}.

VBR encoding, enabled through the @option{vbr} or @option{flags
+qscale} options, is experimental and only works with some
combinations of parameters.

Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or
higher.

For more information see the fdk-aac project at
@url{http://sourceforge.net/p/opencore-amr/fdk-aac/}.

@subsection Options

The following options are mapped on the shared FFmpeg codec options.

@table @option
@item b
Set bit rate in bits/s. If the bitrate is not explicitly specified, it
is automatically set to a suitable value depending on the selected
profile.

In case VBR mode is enabled the option is ignored.

@item ar
Set audio sampling rate (in Hz).

@item channels
Set the number of audio channels.

@item flags +qscale
Enable fixed quality, VBR (Variable Bit Rate) mode.
Note that VBR is implicitly enabled when the @option{vbr} value is
positive.

@item cutoff
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_low
Low Complexity AAC (LC)

@item aac_he
High Efficiency AAC (HE-AAC)

@item aac_he_v2
High Efficiency AAC version 2 (HE-AACv2)

@item aac_ld
Low Delay AAC (LD)

@item aac_eld
Enhanced Low Delay AAC (ELD)
@end table

If not specified it is set to @samp{aac_low}.
@end table

The following are private options of the libfdk_aac encoder.

@table @option
@item afterburner
Enable afterburner feature if set to 1, disabled if set to 0. This
improves the quality but also the required processing power.

Default value is 1.

@item eld_sbr
Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled
if set to 0.

Default value is 0.

@item signaling
Set SBR/PS signaling style.

It can assume one of the following values:
@table @samp
@item default
choose signaling implicitly (explicit hierarchical by default,
implicit if global header is disabled)

@item implicit
implicit backwards compatible signaling

@item explicit_sbr
explicit SBR, implicit PS signaling

@item explicit_hierarchical
explicit hierarchical signaling
@end table

Default value is @samp{default}.

@item latm
Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.

Default value is 0.

@item header_period
Set StreamMuxConfig and PCE repetition period (in frames) for sending
in-band configuration buffers within LATM/LOAS transport layer.

Must be a 16-bits non-negative integer.

Default value is 0.

@item vbr
Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty
good) and 5 is highest quality. A value of 0 will disable VBR, and CBR
(Constant Bit Rate) is enabled.

Currently only the @samp{aac_low} profile supports VBR encoding.

VBR modes 1-5 correspond to roughly the following average bit rates:

@table @samp
@item 1
32 kbps/channel
@item 2
40 kbps/channel
@item 3
48-56 kbps/channel
@item 4
64 kbps/channel
@item 5
about 80-96 kbps/channel
@end table

Default value is 0.
@end table

@subsection Examples

@itemize
@item
Use @command{ffmpeg} to convert an audio file to VBR AAC in an M4A (MP4)
container:
@example
ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
@end example

@item
Use @command{ffmpeg} to convert an audio file to CBR 64k kbps AAC, using the
High-Efficiency AAC profile:
@example
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
@end example
@end itemize

@anchor{libmp3lame}
@section libmp3lame

LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.

Requires the presence of the libmp3lame headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libmp3lame}.

See @ref{libshine} for a fixed-point MP3 encoder, although with a
lower quality.

@subsection Options

The following options are supported by the libmp3lame wrapper. The
@command{lame}-equivalent of the options are listed in parentheses.

@table @option
@item b (@emph{-b})
Set bitrate expressed in bits/s for CBR or ABR. LAME @code{bitrate} is
expressed in kilobits/s.

@item q (@emph{-V})
Set constant quality setting for VBR. This option is valid only
using the @command{ffmpeg} command-line tool. For library interface
users, use @option{global_quality}.

@item compression_level (@emph{-q})
Set algorithm quality. Valid arguments are integers in the 0-9 range,
with 0 meaning highest quality but slowest, and 9 meaning fastest
while producing the worst quality.

@item reservoir
Enable use of bit reservoir when set to 1. Default value is 1. LAME
has this enabled by default, but can be overridden by use
@option{--nores} option.

@item joint_stereo (@emph{-m j})
Enable the encoder to use (on a frame by frame basis) either L/R
stereo or mid/side stereo. Default value is 1.

@item abr (@emph{--abr})
Enable the encoder to use ABR when set to 1. The @command{lame}
@option{--abr} sets the target bitrate, while this options only
tells FFmpeg to use ABR still relies on @option{b} to set bitrate.

@end table

@section libopencore-amrnb

OpenCORE Adaptive Multi-Rate Narrowband encoder.

Requires the presence of the libopencore-amrnb headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopencore-amrnb --enable-version3}.

This is a mono-only encoder. Officially it only supports 8000Hz sample rate,
but you can override it by setting @option{strict} to @samp{unofficial} or
lower.

@subsection Options

@table @option

@item b
Set bitrate in bits per second. Only the following bitrates are supported,
otherwise libavcodec will round to the nearest valid bitrate.

@table @option
@item 4750
@item 5150
@item 5900
@item 6700
@item 7400
@item 7950
@item 10200
@item 12200
@end table

@item dtx
Allow discontinuous transmission (generate comfort noise) when set to 1. The
default value is 0 (disabled).

@end table

@anchor{libshine}
@section libshine

Shine Fixed-Point MP3 encoder wrapper.

Shine is a fixed-point MP3 encoder. It has a far better performance on
platforms without an FPU, e.g. armel CPUs, and some phones and tablets.
However, as it is more targeted on performance than quality, it is not on par
with LAME and other production-grade encoders quality-wise. Also, according to
the project's homepage, this encoder may not be free of bugs as the code was
written a long time ago and the project was dead for at least 5 years.

This encoder only supports stereo and mono input. This is also CBR-only.

The original project (last updated in early 2007) is at
@url{http://sourceforge.net/projects/libshine-fxp/}. We only support the
updated fork by the Savonet/Liquidsoap project at @url{https://github.com/savonet/shine}.

Requires the presence of the libshine headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libshine}.

See also @ref{libmp3lame}.

@subsection Options

The following options are supported by the libshine wrapper. The
@command{shineenc}-equivalent of the options are listed in parentheses.

@table @option
@item b (@emph{-b})
Set bitrate expressed in bits/s for CBR. @command{shineenc} @option{-b} option
is expressed in kilobits/s.

@end table

@section libtwolame

TwoLAME MP2 encoder wrapper.

Requires the presence of the libtwolame headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libtwolame}.

@subsection Options

The following options are supported by the libtwolame wrapper. The
@command{twolame}-equivalent options follow the FFmpeg ones and are in
parentheses.

@table @option
@item b (@emph{-b})
Set bitrate expressed in bits/s for CBR. @command{twolame} @option{b}
option is expressed in kilobits/s. Default value is 128k.

@item q (@emph{-V})
Set quality for experimental VBR support. Maximum value range is
from -50 to 50, useful range is from -10 to 10. The higher the
value, the better the quality. This option is valid only using the
@command{ffmpeg} command-line tool. For library interface users,
use @option{global_quality}.