This document describes the codecs (decoders and encoders) provided by the libavcodec library.
libavcodec provides some generic global options, which can be set on all the encoders and decoders. In addition each codec may support so-called private options, which are specific for a given codec.
Sometimes, a global option may only affect a specific kind of codec, and may be unsensical or ignored by another, so you need to be aware of the meaning of the specified options. Also some options are meant only for decoding or encoding.
Options may be set by specifying -option value in the
FFmpeg tools, or by setting the value explicitly in the
AVCodecContext options or using the ‘libavutil/opt.h’ API
for programmatic use.
The list of supported options follow:
Set bitrate in bits/s. Default value is 200K.
Set audio bitrate (in bits/s). Default value is 128K.
Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far ratecontrol is willing to deviate from the target average bitrate value. This is not related to min/max bitrate. Lowering tolerance too much has an adverse effect on quality.
Set generic flags.
Possible values:
Use four motion vector by macroblock (mpeg4).
Use 1/4 pel motion compensation.
Use loop filter.
Use fixed qscale.
Use gmc.
Always try a mb with mv=<0,0>.
Use internal 2pass ratecontrol in first pass mode.
Use internal 2pass ratecontrol in second pass mode.
Only decode/encode grayscale.
Do not draw edges.
Set error[?] variables during encoding.
Normalize adaptive quantization.
Use interlaced DCT.
Force low delay.
Place global headers in extradata instead of every keyframe.
Use only bitexact stuff (except (I)DCT).
Apply H263 advanced intra coding / mpeg4 ac prediction.
Deprecated, use mpegvideo private options instead.
Deprecated, use mpegvideo private options instead.
Apply interlaced motion estimation.
Use closed gop.
Set motion estimation method.
Possible values:
zero motion estimation (fastest)
full motion estimation (slowest)
EPZS motion estimation (default)
esa motion estimation (alias for full)
tesa motion estimation
dia motion estimation (alias for epzs)
log motion estimation
phods motion estimation
X1 motion estimation
hex motion estimation
umh motion estimation
iter motion estimation
Set extradata size.
Set codec time base.
It is the fundamental unit of time (in seconds) in terms of which
frame timestamps are represented. For fixed-fps content, timebase
should be 1 / frame_rate and timestamp increments should be
identically 1.
Set the group of picture size. Default value is 12.
Set audio sampling rate (in Hz).
Set number of audio channels.
Set cutoff bandwidth.
Set audio frame size.
Each submitted frame except the last must contain exactly frame_size samples per channel. May be 0 when the codec has CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not restricted. It is set by some decoders to indicate constant frame size.
Set the frame number.
Set video quantizer scale compression (VBR). It is used as a constant in the ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0.
Set video quantizer scale blur (VBR).
Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2.
Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31.
Set max difference between the quantizer scale (VBR).
Set max number of B frames between non-B-frames.
Must be an integer between -1 and 16. 0 means that B-frames are disabled. If a value of -1 is used, it will choose an automatic value depending on the encoder.
Default value is 0.
Set qp factor between P and B frames.
Set ratecontrol method.
Set strategy to choose between I/P/B-frames.
Set RTP payload size in bytes.
Workaround not auto detected encoder bugs.
Possible values:
some old lavc generated msmpeg4v3 files (no autodetection)
Xvid interlacing bug (autodetected if fourcc==XVIX)
(autodetected if fourcc==UMP4)
padding bug (autodetected)
illegal vlc bug (autodetected per fourcc)
old standard qpel (autodetected per fourcc/version)
direct-qpel-blocksize bug (autodetected per fourcc/version)
edge padding bug (autodetected per fourcc/version)
Workaround various bugs in microsoft broken decoders.
trancated frames
Set single coefficient elimination threshold for luminance (negative values also consider DC coefficient).
Set single coefficient elimination threshold for chrominance (negative values also consider dc coefficient)
Specify how strictly to follow the standards.
Possible values:
strictly conform to a older more strict version of the spec or reference software
strictly conform to all the things in the spec no matter what consequences
allow unofficial extensions
allow non standardized experimental things, experimental (unfinished/work in progress/not well tested) decoders and encoders. Note: experimental decoders can pose a security risk, do not use this for decoding untrusted input.
Set QP offset between P and B frames.
Set error detection flags.
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
consider things that violate the spec and have not been seen in the wild as errors
consider all spec non compliancies as errors
consider things that a sane encoder should not do as an error
Use MPEG quantizers instead of H.263.
How to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function).
Set experimental quantizer modulation.
Set experimental quantizer modulation.
Set rate control equation. When computing the expression, besides the standard functions defined in the section ’Expression Evaluation’, the following functions are available: bits2qp(bits), qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise.
Set ratecontrol buffer size (in bits).
Currently useless.
Set QP factor between P and I frames.
Set QP offset between P and I frames.
Set initial complexity for 1-pass encoding.
Set DCT algorithm.
Possible values:
autoselect a good one (default)
fast integer
accurate integer
floating point AAN DCT
Compress bright areas stronger than medium ones.
Set temporal complexity masking.
Set spatial complexity masking.
Set inter masking.
Compress dark areas stronger than medium ones.
Select IDCT implementation.
Possible values:
floating point AAN IDCT
Set error concealment strategy.
Possible values:
iterative motion vector (MV) search (slow)
use strong deblock filter for damaged MBs
Set prediction method.
Possible values:
Set sample aspect ratio.
Print specific debug info.
Possible values:
picture info
rate control
macroblock (MB) type
per-block quantization parameter (QP)
motion vector
error recognition
memory management control operations (H.264)
visualize quantization parameter (QP), lower QP are tinted greener
visualize block types
picture buffer allocations
threading operations
Visualize motion vectors (MVs).
Possible values:
forward predicted MVs of P-frames
forward predicted MVs of B-frames
backward predicted MVs of B-frames
Set full pel me compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set sub pel me compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set macroblock compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set interlaced dct compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set diamond type & size for motion estimation.
Set amount of motion predictors from the previous frame.
Set pre motion estimation.
Set pre motion estimation compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set diamond type & size for motion estimation pre-pass.
Set sub pel motion estimation quality.
Set limit motion vectors range (1023 for DivX player).
Set intra quant bias.
Set inter quant bias.
Possible values:
variable length coder / huffman coder
arithmetic coder
raw (no encoding)
run-length coder
deflate-based coder
Set context model.
Set macroblock decision algorithm (high quality mode).
Possible values:
use mbcmp (default)
use fewest bits
use best rate distortion
Set scene change threshold.
Set min lagrange factor (VBR).
Set max lagrange factor (VBR).
Set noise reduction.
Set number of bits which should be loaded into the rc buffer before decoding starts.
Possible values:
Allow non spec compliant speedup tricks.
Deprecated, use mpegvideo private options instead.
Skip bitstream encoding.
Ignore cropping information from sps.
Place global headers at every keyframe instead of in extradata.
Frame data might be split into multiple chunks.
Show all frames before the first keyframe.
Deprecated, use mpegvideo private options instead.
Deprecated, use mpegvideo private options instead.
Possible values:
detect a good number of threads
Set motion estimation threshold.
Set macroblock threshold.
Set intra_dc_precision.
Set nsse weight.
Set number of macroblock rows at the top which are skipped.
Set number of macroblock rows at the bottom which are skipped.
Possible values:
Possible values:
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
Set frame skip threshold.
Set frame skip factor.
Set frame skip exponent. Negative values behave identical to the corresponding positive ones, except that the score is normalized. Positive values exist primarly for compatibility reasons and are not so useful.
Set frame skip compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Increase the quantizer for macroblocks close to borders.
Set min macroblock lagrange factor (VBR).
Set max macroblock lagrange factor (VBR).
Set motion estimation bitrate penalty compensation (1.0 = 256).
Make decoder discard processing depending on the frame type selected by the option value.
‘skip_loop_filter’ skips frame loop filtering, ‘skip_idct’ skips frame IDCT/dequantization, ‘skip_frame’ skips decoding.
Possible values:
Discard no frame.
Discard useless frames like 0-sized frames.
Discard all non-reference frames.
Discard all bidirectional frames.
Discard all frames excepts keyframes.
Discard all frames.
Default value is ‘default’.
Refine the two motion vectors used in bidirectional macroblocks.
Downscale frames for dynamic B-frame decision.
Set minimum interval between IDR-frames.
Set reference frames to consider for motion compensation.
Set chroma qp offset from luma.
Set rate-distortion optimal quantization.
Set value multiplied by qscale for each frame and added to scene_change_score.
Adjust sensitivity of b_frame_strategy 1.
Set GOP timecode frame start number, in non drop frame format.
Set desired number of audio channels.
Possible values:
Possible values:
Set the log level offset.
Number of slices, used in parallelized encoding.
Select multithreading type.
Possible values:
Set audio service type.
Possible values:
Main Audio Service
Effects
Visually Impaired
Hearing Impaired
Dialogue
Commentary
Emergency
Voice Over
Karaoke
Set sample format audio decoders should prefer. Default value is
none.
Set the input subtitles character encoding.
Set/override the field order of the video. Possible values:
Progressive video
Interlaced video, top field coded and displayed first
Interlaced video, bottom field coded and displayed first
Interlaced video, top coded first, bottom displayed first
Interlaced video, bottom coded first, top displayed first
Set to 1 to disable processing alpha (transparency). This works like the ‘gray’ flag in the ‘flags’ option which skips chroma information instead of alpha. Default is 0.
Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding --enable-lib option. You can list all
available decoders using the configure option --list-decoders.
You can disable all the decoders with the configure option
--disable-decoders and selectively enable / disable single decoders
with the options --enable-decoder=DECODER /
--disable-decoder=DECODER.
The option -decoders of the ff* tools will display the list of
enabled decoders.
A description of some of the currently available video decoders follows.
Raw video decoder.
This decoder decodes rawvideo streams.
Specify the assumed field type of the input video.
the video is assumed to be progressive (default)
bottom-field-first is assumed
top-field-first is assumed
A description of some of the currently available audio decoders follows.
AC-3 audio decoder.
This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet).
Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This factor is applied exponentially. There are 3 notable scale factor ranges:
DRC disabled. Produces full range audio.
DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression.
DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds are enhanced.
Internal wave synthetizer.
This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented.
libcelt decoder wrapper.
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
Requires the presence of the libcelt headers and library during configuration.
You need to explicitly configure the build with --enable-libcelt.
libgsm decoder wrapper.
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
the presence of the libgsm headers and library during configuration. You need
to explicitly configure the build with --enable-libgsm.
This decoder supports both the ordinary GSM and the Microsoft variant.
libilbc decoder wrapper.
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
audio codec. Requires the presence of the libilbc headers and library during
configuration. You need to explicitly configure the build with
--enable-libilbc.
The following option is supported by the libilbc wrapper.
Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).
libopencore-amrnb decoder wrapper.
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
Narrowband audio codec. Using it requires the presence of the
libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with --enable-libopencore-amrnb.
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library.
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
Wideband audio codec. Using it requires the presence of the
libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with --enable-libopencore-amrwb.
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library.
libopus decoder wrapper.
libopus allows libavcodec to decode the Opus Interactive Audio Codec.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
--enable-libopus.
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can also be found in VobSub file pairs and in some Matroska files.
Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when reading from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example 0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b.
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
subtitles. Requires the presence of the libzvbi headers and library during
configuration. You need to explicitly configure the build with
--enable-libzvbi.
List of teletext page numbers to decode. You may use the special * string to match all pages. Pages that do not match the specified list are dropped. Default value is *.
Discards the top teletext line. Default value is 1.
Specifies the format of the decoded subtitles. The teletext decoder is capable of decoding the teletext pages to bitmaps or to simple text, you should use "bitmap" for teletext pages, because certain graphics and colors cannot be expressed in simple text. You might use "text" for teletext based subtitles if your application can handle simple text based subtitles. Default value is bitmap.
X offset of generated bitmaps, default is 0.
Y offset of generated bitmaps, default is 0.
Chops leading and trailing spaces and removes empty lines from the generated text. This option is useful for teletext based subtitles where empty spaces may be present at the start or at the end of the lines or empty lines may be present between the subtitle lines because of double-sized teletext charactes. Default value is 1.
Sets the display duration of the decoded teletext pages or subtitles in miliseconds. Default value is 30000 which is 30 seconds.
Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque (black) background.
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 --enable-lib option. You can list all
available encoders using the configure option --list-encoders.
You can disable all the encoders with the configure option
--disable-encoders and selectively enable / disable single encoders
with the options --enable-encoder=ENCODER /
--disable-encoder=ENCODER.
The option -encoders of the ff* tools will display the list of
enabled encoders.
A description of some of the currently available audio encoders follows.
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 ‘strict’ option to ‘experimental’ or lower.
As this encoder is experimental, unexpected behavior may exist from time to time. For a more stable AAC encoder, see libvo-aacenc. However, be warned that it has a worse quality reported by some users.
See also libfdk_aac and libfaac.
Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode.
Set quality for variable bit rate (VBR) mode. This option is valid only using
the ffmpeg command-line tool. For library interface users, use
‘global_quality’.
Set stereo encoding mode. Possible values:
Automatically selected by the encoder.
Disable middle/side encoding. This is the default.
Force middle/side encoding.
Set AAC encoder coding method. Possible values:
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 descibed below, but somewhat a little better and slower.
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.
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.
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.
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 ac3 encoder uses floating-point math, while the 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 ac3_fixed encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option -acodec ac3_fixed in order to use it.
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.
Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
The metadata values set at initialization will be used for every frame in the stream. (default)
Metadata values can be changed before encoding each frame.
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:
Apply -3dB gain
Apply -4.5dB gain (default)
Apply -6dB gain
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:
Apply -3dB gain
Apply -6dB gain (default)
Silence Surround Channel(s)
Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream.
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 room_type option is not the default value, the mixing_level
option must not be -1.
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 mixing_level
option and the room_type option have the default values.
Not Indicated (default)
Large Room
Small Room
Copyright Indicator. Specifies whether a copyright exists for this audio.
No Copyright Exists (default)
Copyright Exists
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.
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 NOT mean the encoder will actually apply Dolby Surround processing.
Not Indicated (default)
Not Dolby Surround Encoded
Dolby Surround Encoded
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.
Not Original Source
Original Source (default)
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 center_mixlev
and surround_mixlev options if it supports the Alternate Bit Stream
Syntax.
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.
Not Indicated (default)
Lt/Rt Downmix Preferred
Lo/Ro Downmix Preferred
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.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
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.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
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.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
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.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.
Not Indicated (default)
Dolby Surround EX Off
Dolby Surround EX On
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 NOT mean the encoder will actually apply Dolby Headphone processing.
Not Indicated (default)
Dolby Headphone Off
Dolby Headphone On
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.
Standard A/D Converter (default)
HDCD A/D Converter
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.
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.
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.
Selected by Encoder (default)
Disable Channel Coupling
Enable Channel Coupling
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 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.
Selected by Encoder (default)
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
--enable-libfaac --enable-nonfree.
This encoder is considered to be of higher quality with respect to the the native experimental FFmpeg AAC encoder.
For more information see the libfaac project at http://www.audiocoding.com/faac.html/.
The following shared FFmpeg codec options are recognized.
The following options are supported by the libfaac wrapper. The
faac-equivalent of the options are listed in parentheses.
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. 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.
Set audio sampling rate (in Hz).
Set the number of audio channels.
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.
Set audio profile.
The following profiles are recognized:
Main AAC (Main)
Low Complexity AAC (LC)
Scalable Sample Rate (SSR)
Long Term Prediction (LTP)
If not specified it is set to ‘aac_low’.
Set constant quality VBR (Variable Bit Rate) mode.
Set quality in VBR mode as an integer number of lambda units.
Only relevant when VBR mode is enabled with flags +qscale. The
value is converted to QP units by dividing it by 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.
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 ffmpeg command-line
tool. For library interface users, use ‘global_quality’.
ffmpeg to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
container:
ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a |
ffmpeg to convert an audio file to VBR AAC, using the
LTP AAC profile:
ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a |
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
--enable-libfdk-aac. The library is also incompatible with GPL,
so if you allow the use of GPL, you should configure with
--enable-gpl --enable-nonfree --enable-libfdk-aac.
This encoder is considered to be of higher quality with respect to both the native experimental FFmpeg AAC encoder and libfaac.
VBR encoding, enabled through the ‘vbr’ or ‘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 http://sourceforge.net/p/opencore-amr/fdk-aac/.
The following options are mapped on the shared FFmpeg codec options.
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.
Set audio sampling rate (in Hz).
Set the number of audio channels.
Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the ‘vbr’ value is positive.
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.
Set audio profile.
The following profiles are recognized:
Low Complexity AAC (LC)
High Efficiency AAC (HE-AAC)
High Efficiency AAC version 2 (HE-AACv2)
Low Delay AAC (LD)
Enhanced Low Delay AAC (ELD)
If not specified it is set to ‘aac_low’.
The following are private options of the libfdk_aac encoder.
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.
Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0.
Default value is 0.
Set SBR/PS signaling style.
It can assume one of the following values:
choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)
implicit backwards compatible signaling
explicit SBR, implicit PS signaling
explicit hierarchical signaling
Default value is ‘default’.
Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
Default value is 0.
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.
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 ‘aac_low’ profile supports VBR encoding.
VBR modes 1-5 correspond to roughly the following average bit rates:
32 kbps/channel
40 kbps/channel
48-56 kbps/channel
64 kbps/channel
about 80-96 kbps/channel
Default value is 0.
ffmpeg to convert an audio file to VBR AAC in an M4A (MP4)
container:
ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a |
ffmpeg to convert an audio file to CBR 64k kbps AAC, using the
High-Efficiency AAC profile:
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a |
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
--enable-libmp3lame.
See libshine for a fixed-point MP3 encoder, although with a lower quality.
The following options are supported by the libmp3lame wrapper. The
lame-equivalent of the options are listed in parentheses.
Set bitrate expressed in bits/s for CBR or ABR. LAME bitrate is
expressed in kilobits/s.
Set constant quality setting for VBR. This option is valid only
using the ffmpeg command-line tool. For library interface
users, use ‘global_quality’.
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.
Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled by default, but can be overriden by use ‘--nores’ option.
Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default value is 1.
Enable the encoder to use ABR when set to 1. The lame
‘--abr’ sets the target bitrate, while this options only
tells FFmpeg to use ABR still relies on ‘b’ to set bitrate.
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
--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 ‘strict’ to ‘unofficial’ or lower.
Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
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 http://sourceforge.net/projects/libshine-fxp/. We only support the updated fork by the Savonet/Liquidsoap project at https://github.com/savonet/shine.
Requires the presence of the libshine headers and library during
configuration. You need to explicitly configure the build with
--enable-libshine.
See also libmp3lame.
The following options are supported by the libshine wrapper. The
shineenc-equivalent of the options are listed in parentheses.
Set bitrate expressed in bits/s for CBR. shineenc ‘-b’ option
is expressed in kilobits/s.
TwoLAME MP2 encoder wrapper.
Requires the presence of the libtwolame headers and library during
configuration. You need to explicitly configure the build with
--enable-libtwolame.
The following options are supported by the libtwolame wrapper. The
twolame-equivalent options follow the FFmpeg ones and are in
parentheses.
Set bitrate expressed in bits/s for CBR. twolame ‘b’
option is expressed in kilobits/s. Default value is 128k.
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
ffmpeg command-line tool. For library interface users,
use ‘global_quality’.
Set the mode of the resulting audio. Possible values:
Choose mode automatically based on the input. This is the default.
Stereo
Joint stereo
Dual channel
Mono
Set psychoacoustic model to use in encoding. The argument must be an integer between -1 and 4, inclusive. The higher the value, the better the quality. The default value is 3.
Enable energy levels extensions when set to 1. The default value is 0 (disabled).
Enable CRC error protection when set to 1. The default value is 0 (disabled).
Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).
Set MPEG audio original flag when set to 1. The default value is 0 (disabled).
VisualOn AAC encoder.
Requires the presence of the libvo-aacenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvo-aacenc --enable-version3.
This encoder is considered to be worse than the native experimental FFmpeg AAC encoder, according to multiple sources.
The VisualOn AAC encoder only support encoding AAC-LC and up to 2 channels. It is also CBR-only.
Set bit rate in bits/s.
VisualOn Adaptive Multi-Rate Wideband encoder.
Requires the presence of the libvo-amrwbenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvo-amrwbenc --enable-version3.
This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can override it by setting ‘strict’ to ‘unofficial’ or lower.
Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
libopus Opus Interactive Audio Codec encoder wrapper.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
--enable-libopus.
Most libopus options are modeled after the opusenc utility from
opus-tools. The following is an option mapping chart describing options
supported by the libopus wrapper, and their opusenc-equivalent
in parentheses.
Set the bit rate in bits/s. FFmpeg’s ‘b’ option is
expressed in bits/s, while opusenc’s ‘bitrate’ in
kilobits/s.
Set VBR mode. The FFmpeg ‘vbr’ option has the following
valid arguments, with the their opusenc equivalent options
in parentheses:
Use constant bit rate encoding.
Use variable bit rate encoding (the default).
Use constrained variable bit rate encoding.
Set encoding algorithm complexity. Valid options are integers in the 0-10 range. 0 gives the fastest encodes but lower quality, while 10 gives the highest quality but slowest encoding. The default is 10.
Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller frame sizes achieve lower latency but less quality at a given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is 20ms.
Set expected packet loss percentage. The default is 0.
Set intended application type. Valid options are listed below:
Favor improved speech intelligibility.
Favor faithfulness to the input (the default).
Restrict to only the lowest delay modes.
Set cutoff bandwidth in Hz. The argument must be exactly one of the following: 4000, 6000, 8000, 12000, or 20000, corresponding to narrowband, mediumband, wideband, super wideband, and fullband respectively. The default is 0 (cutoff disabled).
libvorbis encoder wrapper.
Requires the presence of the libvorbisenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvorbis.
The following options are supported by the libvorbis wrapper. The
oggenc-equivalent of the options are listed in parentheses.
To get a more accurate and extensive documentation of the libvorbis
options, consult the libvorbisenc’s and oggenc’s documentations.
See http://xiph.org/vorbis/,
http://wiki.xiph.org/Vorbis-tools, and oggenc(1).
Set bitrate expressed in bits/s for ABR. oggenc ‘-b’ is
expressed in kilobits/s.
Set constant quality setting for VBR. The value should be a float number in the range of -1.0 to 10.0. The higher the value, the better the quality. The default value is ‘3.0’.
This option is valid only using the ffmpeg command-line tool.
For library interface users, use ‘global_quality’.
Set cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc’s
related option is expressed in kHz. The default value is ‘0’ (cutoff
disabled).
Set minimum bitrate expressed in bits/s. oggenc ‘-m’ is
expressed in kilobits/s.
Set maximum bitrate expressed in bits/s. oggenc ‘-M’ is
expressed in kilobits/s. This only has effect on ABR mode.
Set noise floor bias for impulse blocks. The value is a float number from -15.0 to 0.0. A negative bias instructs the encoder to pay special attention to the crispness of transients in the encoded audio. The tradeoff for better transient response is a higher bitrate.
A wrapper providing WavPack encoding through libwavpack.
Only lossless mode using 32-bit integer samples is supported currently.
Requires the presence of the libwavpack headers and library during
configuration. You need to explicitly configure the build with
--enable-libwavpack.
Note that a libavcodec-native encoder for the WavPack codec exists so users can encode audios with this codec without using this encoder. See wavpackenc.
wavpack command line utility’s corresponding options are listed in
parentheses, if any.
Default is 32768.
Set speed vs. compression tradeoff. Acceptable arguments are listed below:
Fast mode.
Normal (default) settings.
High quality.
Very high quality.
Same as ‘3’, but with extra processing enabled.
‘4’ is the same as ‘-x2’ and ‘8’ is the same as ‘-x6’.
WavPack lossless audio encoder.
This is a libavcodec-native WavPack encoder. There is also an encoder based on libwavpack, but there is virtually no reason to use that encoder.
See also libwavpack.
The equivalent options for wavpack command line utility are listed in
parentheses.
The following shared options are effective for this encoder. Only special notes about this particular encoder will be documented here. For the general meaning of the options, see the Codec Options chapter.
For this encoder, the range for this option is between 128 and 131072. Default is automatically decided based on sample rate and number of channel.
For the complete formula of calculating default, see ‘libavcodec/wavpackenc.c’.
This option’s syntax is consistent with libwavpack’s.
Set whether to enable joint stereo. Valid values are:
Force mid/side audio encoding.
Force left/right audio encoding.
Let the encoder decide automatically.
Set whether to enable optimization for mono. This option is only effective for non-mono streams. Available values:
enabled
disabled
A description of some of the currently available video encoders follows.
libtheora Theora encoder wrapper.
Requires the presence of the libtheora headers and library during
configuration. You need to explicitly configure the build with
--enable-libtheora.
For more informations about the libtheora project see http://www.theora.org/.
The following global options are mapped to internal libtheora options which affect the quality and the bitrate of the encoded stream.
Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In case VBR (Variable Bit Rate) mode is enabled this option is ignored.
Used to enable constant quality mode (VBR) encoding through the
‘qscale’ flag, and to enable the pass1 and pass2
modes.
Set the GOP size.
Set the global quality as an integer in lambda units.
Only relevant when VBR mode is enabled with flags +qscale. The
value is converted to QP units by dividing it by FF_QP2LAMBDA,
clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
value in the native libtheora range [0-63]. A higher value corresponds
to a higher quality.
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 is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63].
This option is valid only using the ffmpeg command-line
tool. For library interface users, use ‘global_quality’.
ffmpeg:
ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg |
ffmpeg to convert a CBR 1000 kbps Theora video stream:
ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg |
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with --enable-libvpx.
Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
g_threads
g_profile
rc_target_bitrate
kf_max_dist
kf_min_dist
rc_min_quantizer
rc_max_quantizer
rc_buf_sz
(bufsize * 1000 / vb)
rc_buf_optimal_sz
(bufsize * 1000 / vb * 5 / 6)
rc_buf_initial_sz
(rc_init_occupancy * 1000 / vb)
rc_undershoot_pct
rc_dropframe_thresh
rc_2pass_vbr_bias_pct
rc_2pass_vbr_maxsection_pct
(maxrate * 100 / vb)
rc_2pass_vbr_minsection_pct
(minrate * 100 / vb)
VPX_CBR
(minrate == maxrate == vb)
VPX_CQ, VP8E_SET_CQ_LEVEL
VPX_DL_BEST_QUALITY
VPX_DL_GOOD_QUALITY
VPX_DL_REALTIME
VP8E_SET_CPUUSED
VP8E_SET_NOISE_SENSITIVITY
VP8E_SET_STATIC_THRESHOLD
VP8E_SET_TOKEN_PARTITIONS
VP8E_SET_MAX_INTRA_BITRATE_PCT
VPX_EFLAG_FORCE_KF
VP8E_SET_ENABLEAUTOALTREF
VP8E_SET_ARNR_MAXFRAMES
VP8E_SET_ARNR_TYPE
VP8E_SET_ARNR_STRENGTH
g_lag_in_frames
g_error_resilient
For more information about libvpx see: http://www.webmproject.org/
libwebp WebP Image encoder wrapper
libwebp is Google’s official encoder for WebP images. It can encode in either lossy or lossless mode. Lossy images are essentially a wrapper around a VP8 frame. Lossless images are a separate codec developed by Google.
Currently, libwebp only supports YUV420 for lossy and RGB for lossless due to limitations of the format and libwebp. Alpha is supported for either mode. Because of API limitations, if RGB is passed in when encoding lossy or YUV is passed in for encoding lossless, the pixel format will automatically be converted using functions from libwebp. This is not ideal and is done only for convenience.
Enables/Disables use of lossless mode. Default is 0.
For lossy, this is a quality/speed tradeoff. Higher values give better quality for a given size at the cost of increased encoding time. For lossless, this is a size/speed tradeoff. Higher values give smaller size at the cost of increased encoding time. More specifically, it controls the number of extra algorithms and compression tools used, and varies the combination of these tools. This maps to the method option in libwebp. The valid range is 0 to 6. Default is 4.
For lossy encoding, this controls image quality, 0 to 100. For lossless encoding, this controls the effort and time spent at compressing more. The default value is 75. Note that for usage via libavcodec, this option is called global_quality and must be multiplied by FF_QP2LAMBDA.
Configuration preset. This does some automatic settings based on the general type of the image.
Do not use a preset.
Use the encoder default.
Digital picture, like portrait, inner shot
Outdoor photograph, with natural lighting
Hand or line drawing, with high-contrast details
Small-sized colorful images
Text-like
x264 H.264/MPEG-4 AVC encoder wrapper.
This encoder requires the presence of the libx264 headers and library
during configuration. You need to explicitly configure the build with
--enable-libx264.
libx264 supports an impressive number of features, including 8x8 and 4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, interlacing (MBAFF), lossless mode, psy optimizations for detail retention (adaptive quantization, psy-RD, psy-trellis).
Many libx264 encoder options are mapped to FFmpeg global codec
options, while unique encoder options are provided through private
options. Additionally the ‘x264opts’ and ‘x264-params’
private options allows to pass a list of key=value tuples as accepted
by the libx264 x264_param_parse function.
The x264 project website is at http://www.videolan.org/developers/x264.html.
The libx264rgb encoder is the same as libx264, except it accepts packed RGB pixel formats as input instead of YUV.
x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264’s configure time. FFmpeg only supports one bit depth in one particular build. In other words, it is not possible to build one FFmpeg with multiple versions of x264 with different bit depths.
The following options are supported by the libx264 wrapper. The
x264-equivalent options or values are listed in parentheses
for easy migration.
To reduce the duplication of documentation, only the private options and some others requiring special attention are documented here. For the documentation of the undocumented generic options, see the Codec Options chapter.
To get a more accurate and extensive documentation of the libx264
options, invoke the command x264 --full-help or consult
the libx264 documentation.
Set bitrate in bits/s. Note that FFmpeg’s ‘b’ option is
expressed in bits/s, while x264’s ‘bitrate’ is in
kilobits/s.
Set motion estimation method. Possible values in the decreasing order of speed:
Diamond search with radius 1 (fastest). ‘epzs’ is an alias for ‘dia’.
Hexagonal search with radius 2.
Uneven multi-hexagon search.
Exhaustive search.
Hadamard exhaustive search (slowest).
Set entropy encoder. Possible values:
Enable CABAC.
Enable CAVLC and disable CABAC. It generates the same effect as
x264’s ‘--no-cabac’ option.
Set full pixel motion estimation comparation algorithm. Possible values:
Enable chroma in motion estimation.
Ignore chroma in motion estimation. It generates the same effect as
x264’s ‘--no-chroma-me’ option.
Set multithreading technique. Possible values:
Slice-based multithreading. It generates the same effect as
x264’s ‘--sliced-threads’ option.
Frame-based multithreading.
Set encoding flags. It can be used to disable closed GOP and enable
open GOP by setting it to -cgop. The result is similar to
the behavior of x264’s ‘--open-gop’ option.
Set the encoding preset.
Set tuning of the encoding params.
Set profile restrictions.
Enable fast settings when encoding first pass, when set to 1. When set
to 0, it has the same effect of x264’s
‘--slow-firstpass’ option.
Set the quality for constant quality mode.
In CRF mode, prevents VBV from lowering quality beyond this point.
Set constant quantization rate control method parameter.
Set AQ method. Possible values:
Disabled.
Variance AQ (complexity mask).
Auto-variance AQ (experimental).
Set AQ strength, reduce blocking and blurring in flat and textured areas.
Use psychovisual optimizations when set to 1. When set to 0, it has the
same effect as x264’s ‘--no-psy’ option.
Set strength of psychovisual optimization, in psy-rd:psy-trellis format.
Set number of frames to look ahead for frametype and ratecontrol.
Enable weighted prediction for B-frames when set to 1. When set to 0,
it has the same effect as x264’s ‘--no-weightb’ option.
Set weighted prediction method for P-frames. Possible values:
Disabled
Enable only weighted refs
Enable both weighted refs and duplicates
Enable calculation and printing SSIM stats after the encoding.
Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.
Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting "bluray-compat=1 force-cfr=1".
Set the influence on how often B-frames are used.
Set method for keeping of some B-frames as references. Possible values:
Disabled.
Strictly hierarchical pyramid.
Non-strict (not Blu-ray compatible).
Enable the use of one reference per partition, as opposed to one
reference per macroblock when set to 1. When set to 0, it has the
same effect as x264’s ‘--no-mixed-refs’ option.
Enable adaptive spatial transform (high profile 8x8 transform)
when set to 1. When set to 0, it has the same effect as
x264’s ‘--no-8x8dct’ option.
Enable early SKIP detection on P-frames when set to 1. When set
to 0, it has the same effect as x264’s
‘--no-fast-pskip’ option.
Enable use of access unit delimiters when set to 1.
Enable use macroblock tree ratecontrol when set to 1. When set
to 0, it has the same effect as x264’s
‘--no-mbtree’ option.
Set loop filter parameters, in alpha:beta form.
Set fluctuations reduction in QP (before curve compression).
Set partitions to consider as a comma-separated list of. Possible values in the list:
8x8 P-frame partition.
4x4 P-frame partition.
4x4 B-frame partition.
8x8 I-frame partition.
4x4 I-frame partition. (Enabling ‘p4x4’ requires ‘p8x8’ to be enabled. Enabling ‘i8x8’ requires adaptive spatial transform (‘8x8dct’ option) to be enabled.)
Do not consider any partitions.
Consider every partition.
Set direct MV prediction mode. Possible values:
Disable MV prediction.
Enable spatial predicting.
Enable temporal predicting.
Automatically decided.
Set the limit of the size of each slice in bytes. If not specified but RTP payload size (‘ps’) is specified, that is used.
Set the file name for multi-pass stats.
Set signal HRD information (requires ‘vbv-bufsize’ to be set). Possible values:
Disable HRD information signaling.
Variable bit rate.
Constant bit rate (not allowed in MP4 container).
Set any x264 option, see x264 --fullhelp for a list.
Argument is a list of key=value couples separated by ":". In filter and psy-rd options that use ":" as a separator themselves, use "," instead. They accept it as well since long ago but this is kept undocumented for some reason.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv |
Override the x264 configuration using a :-separated list of key=value parameters.
This option is functionally the same as the ‘x264opts’, but is duplicated for compability with the Libav fork.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\ cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\ no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT |
Encoding ffpresets for common usages are provided so they can be used with the general presets system (e.g. passing the ‘pre’ option).
Xvid MPEG-4 Part 2 encoder wrapper.
This encoder requires the presence of the libxvidcore headers and library
during configuration. You need to explicitly configure the build with
--enable-libxvid --enable-gpl.
The native mpeg4 encoder supports the MPEG-4 Part 2 format, so
users can encode to this format without this library.
The following options are supported by the libxvid wrapper. Some of the following options are listed but are not documented, and correspond to shared codec options. See the Codec Options chapter for their documentation. The other shared options which are not listed have no effect for the libxvid encoder.
Set specific encoding flags. Possible values:
Use four motion vector by macroblock.
Enable high quality AC prediction.
Only encode grayscale.
Enable the use of global motion compensation (GMC).
Enable quarter-pixel motion compensation.
Enable closed GOP.
Place global headers in extradata instead of every keyframe.
Set motion estimation method. Possible values in decreasing order of speed and increasing order of quality:
Use no motion estimation (default).
Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks. ‘x1’ and ‘log’ are aliases for ‘phods’.
Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion estimation on chroma planes.
Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.
Set macroblock decision algorithm. Possible values in the increasing order of quality:
Use macroblock comparing function algorithm (default).
Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.
Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel refinement for 8x8 blocks, and rate distortion-based search using square pattern.
Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).
Enable variance adaptive quantization when set to 1. Default is 0 (disabled).
When combined with ‘lumi_aq’, the resulting quality will not be better than any of the two specified individually. In other words, the resulting quality will be the worse one of the two effects.
Set structural similarity (SSIM) displaying method. Possible values:
Disable displaying of SSIM information.
Output average SSIM at the end of encoding to stdout. The format of showing the average SSIM is:
Average SSIM: %f |
For users who are not familiar with C, %f means a float number, or a decimal (e.g. 0.939232).
Output both per-frame SSIM data during encoding and average SSIM at the end of encoding to stdout. The format of per-frame information is:
SSIM: avg: %1.3f min: %1.3f max: %1.3f |
For users who are not familiar with C, %1.3f means a float number rounded to 3 digits after the dot (e.g. 0.932).
Set SSIM accuracy. Valid options are integers within the range of 0-4, while 0 gives the most accurate result and 4 computes the fastest.
PNG image encoder.
Set physical density of pixels, in dots per inch, unset by default
Set physical density of pixels, in dots per meter, unset by default
Apple ProRes encoder.
FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
The used encoder can be choosen with the -vcodec option.
Select the ProRes profile to encode
Select quantization matrix.
If set to auto, the matrix matching the profile will be picked. If not set, the matrix providing the highest quality, default, will be picked.
How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits per macroblock, the maximum is 8000.
Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all situations.
Override the 4-byte vendor ID. A custom vendor ID like apl0 would claim the stream was produced by the Apple encoder.
Specify number of bits for alpha component. Possible values are 0, 8 and 16. Use 0 to disable alpha plane coding.
In the default mode of operation the encoder has to honor frame constraints (i.e. not produc frames with size bigger than requested) while still making output picture as good as possible. A frame containing a lot of small details is harder to compress and the encoder would spend more time searching for appropriate quantizers for each slice.
Setting a higher ‘bits_per_mb’ limit will improve the speed.
For the fastest encoding speed set the ‘qscale’ parameter (4 is the recommended value) and do not set a size constraint.
ffmpeg, ffplay, ffprobe, ffserver, libavcodec
The FFmpeg developers.
For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
git log in the FFmpeg source directory, or browsing the
online repository at http://source.ffmpeg.org.
Maintainers for the specific components are listed in the file ‘MAINTAINERS’ in the source code tree.