ffmpeg Documentation

Table of Contents

1 Synopsis

ffmpeg [global_options] {[input_file_options] -i input_url} ... {[output_file_options] output_url} ...

2 Description

ffmpeg is a universal media converter. It can read a wide variety of inputs - including live grabbing/recording devices - filter, and transcode them into a plethora of output formats.

ffmpeg reads from an arbitrary number of inputs (which can be regular files, pipes, network streams, grabbing devices, etc.), specified by the -i option, and writes to an arbitrary number of outputs, which are specified by a plain output url. Anything found on the command line which cannot be interpreted as an option is considered to be an output url.

Each input or output can, in principle, contain any number of elementary streams of different types (video/audio/subtitle/attachment/data), though the allowed stream counts and/or types may be limited by the container format. Selecting which streams from which inputs will go into which output is either done automatically or with the -map option (see the Stream selection chapter).

To refer to inputs/outputs in options, you must use their indices (0-based). E.g. the first input is 0, the second is 1, etc. Similarly, streams within an input/output are referred to by their indices. E.g. 2:3 refers to the fourth stream in the third input or output. Also see the Stream specifiers chapter.

As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same option on the command line multiple times. Each occurrence is then applied to the next input or output file. Exceptions from this rule are the global options (e.g. verbosity level), which should be specified first.

Do not mix input and output files – first specify all input files, then all output files. Also do not mix options which belong to different files. All options apply ONLY to the next input or output file and are reset between files.

Some simple examples follow.

  • Convert an input media file to a different format, by re-encoding media streams:
    ffmpeg -i input.avi output.mp4
    
  • Set the video bitrate of the output file to 64 kbit/s:
    ffmpeg -i input.avi -b:v 64k -bufsize 64k output.mp4
    
  • Force the frame rate of the output file to 24 fps:
    ffmpeg -i input.avi -r 24 output.mp4
    
  • Force the frame rate of the input file (valid for raw formats only) to 1 fps and the frame rate of the output file to 24 fps:
    ffmpeg -r 1 -i input.m2v -r 24 output.mp4
    

The format option may be needed for raw input files.

3 Detailed description

ffmpeg builds a transcoding pipeline out of the components listed below. The program’s operation then consists of input data chunks flowing from the sources down the pipes towards the sinks, while being transformed by the components they encounter along the way.

The following kinds of components are available:

  • Demuxers (short for "demultiplexers") read an input source in order to extract
    • global properties such as metadata or chapters;
    • list of input elementary streams and their properties

    One demuxer instance is created for each -i option, and sends encoded packets to decoders or muxers.

    In other literature, demuxers are sometimes called splitters, because their main function is splitting a file into elementary streams (though some files only contain one elementary stream).

    A schematic representation of a demuxer looks like this:

    ┌──────────┬───────────────────────┐
    │ demuxer  │                       │ packets for stream 0
    ╞══════════╡ elementary stream 0   ├──────────────────────⮞
    │          │                       │
    │  global  ├───────────────────────┤
    │properties│                       │ packets for stream 1
    │   and    │ elementary stream 1   ├──────────────────────⮞
    │ metadata │                       │
    │          ├───────────────────────┤
    │          │                       │
    │          │     ...........       │
    │          │                       │
    │          ├───────────────────────┤
    │          │                       │ packets for stream N
    │          │ elementary stream N   ├──────────────────────⮞
    │          │                       │
    └──────────┴───────────────────────┘
         ⯅
         │
         │ read from file, network stream,
         │     grabbing device, etc.
         │
    
  • Decoders receive encoded (compressed) packets for an audio, video, or subtitle elementary stream, and decode them into raw frames (arrays of pixels for video, PCM for audio). A decoder is typically associated with (and receives its input from) an elementary stream in a demuxer, but sometimes may also exist on its own (see Loopback decoders).

    A schematic representation of a decoder looks like this:

              ┌─────────┐
     packets  │         │ raw frames
    ─────────⮞│ decoder ├────────────⮞
              │         │
              └─────────┘
    
  • Filtergraphs process and transform raw audio or video frames. A filtergraph consists of one or more individual filters linked into a graph. Filtergraphs come in two flavors - simple and complex, configured with the -filter and -filter_complex options, respectively.

    A simple filtergraph is associated with an output elementary stream; it receives the input to be filtered from a decoder and sends filtered output to that output stream’s encoder.

    A simple video filtergraph that performs deinterlacing (using the yadif deinterlacer) followed by resizing (using the scale filter) can look like this:

                 ┌────────────────────────┐
                 │  simple filtergraph    │
     frames from ╞════════════════════════╡ frames for
     a decoder   │  ┌───────┐  ┌───────┐  │ an encoder
    ────────────⮞├─⮞│ yadif ├─⮞│ scale ├─⮞│────────────⮞
                 │  └───────┘  └───────┘  │
                 └────────────────────────┘
    

    A complex filtergraph is standalone and not associated with any specific stream. It may have multiple (or zero) inputs, potentially of different types (audio or video), each of which receiving data either from a decoder or another complex filtergraph’s output. It also has one or more outputs that feed either an encoder or another complex filtergraph’s input.

    The following example diagram represents a complex filtergraph with 3 inputs and 2 outputs (all video):

              ┌─────────────────────────────────────────────────┐
              │               complex filtergraph               │
              ╞═════════════════════════════════════════════════╡
     frames   ├───────┐  ┌─────────┐      ┌─────────┐  ┌────────┤ frames
    ─────────⮞│input 0├─⮞│ overlay ├─────⮞│ overlay ├─⮞│output 0├────────⮞
              ├───────┘  │         │      │         │  └────────┤
     frames   ├───────┐╭⮞│         │    ╭⮞│         │           │
    ─────────⮞│input 1├╯ └─────────┘    │ └─────────┘           │
              ├───────┘                 │                       │
     frames   ├───────┐ ┌─────┐ ┌─────┬─╯              ┌────────┤ frames
    ─────────⮞│input 2├⮞│scale├⮞│split├───────────────⮞│output 1├────────⮞
              ├───────┘ └─────┘ └─────┘                └────────┤
              └─────────────────────────────────────────────────┘
    

    Frames from second input are overlaid over those from the first. Frames from the third input are rescaled, then the duplicated into two identical streams. One of them is overlaid over the combined first two inputs, with the result exposed as the filtergraph’s first output. The other duplicate ends up being the filtergraph’s second output.

  • Encoders receive raw audio, video, or subtitle frames and encode them into encoded packets. The encoding (compression) process is typically lossy - it degrades stream quality to make the output smaller; some encoders are lossless, but at the cost of much higher output size. A video or audio encoder receives its input from some filtergraph’s output, subtitle encoders receive input from a decoder (since subtitle filtering is not supported yet). Every encoder is associated with some muxer’s output elementary stream and sends its output to that muxer.

    A schematic representation of an encoder looks like this:

                 ┌─────────┐
     raw frames  │         │ packets
    ────────────⮞│ encoder ├─────────⮞
                 │         │
                 └─────────┘
    
  • Muxers (short for "multiplexers") receive encoded packets for their elementary streams from encoders (the transcoding path) or directly from demuxers (the streamcopy path), interleave them (when there is more than one elementary stream), and write the resulting bytes into the output file (or pipe, network stream, etc.).

    A schematic representation of a muxer looks like this:

                           ┌──────────────────────┬───────────┐
     packets for stream 0  │                      │   muxer   │
    ──────────────────────⮞│  elementary stream 0 ╞═══════════╡
                           │                      │           │
                           ├──────────────────────┤  global   │
     packets for stream 1  │                      │properties │
    ──────────────────────⮞│  elementary stream 1 │   and     │
                           │                      │ metadata  │
                           ├──────────────────────┤           │
                           │                      │           │
                           │     ...........      │           │
                           │                      │           │
                           ├──────────────────────┤           │
     packets for stream N  │                      │           │
    ──────────────────────⮞│  elementary stream N │           │
                           │                      │           │
                           └──────────────────────┴─────┬─────┘
                                                        │
                         write to file, network stream, │
                             grabbing device, etc.      │
                                                        │
                                                        ▼
    

3.1 Streamcopy

The simplest pipeline in ffmpeg is single-stream streamcopy, that is copying one input elementary stream’s packets without decoding, filtering, or encoding them. As an example, consider an input file called INPUT.mkv with 3 elementary streams, from which we take the second and write it to file OUTPUT.mp4. A schematic representation of such a pipeline looks like this:

┌──────────┬─────────────────────┐
│ demuxer  │                     │ unused
╞══════════╡ elementary stream 0 ├────────╳
│          │                     │
│INPUT.mkv ├─────────────────────┤          ┌──────────────────────┬───────────┐
│          │                     │ packets  │                      │   muxer   │
│          │ elementary stream 1 ├─────────⮞│  elementary stream 0 ╞═══════════╡
│          │                     │          │                      │OUTPUT.mp4 │
│          ├─────────────────────┤          └──────────────────────┴───────────┘
│          │                     │ unused
│          │ elementary stream 2 ├────────╳
│          │                     │
└──────────┴─────────────────────┘

The above pipeline can be constructed with the following commandline:

ffmpeg -i INPUT.mkv -map 0:1 -c copy OUTPUT.mp4

In this commandline

  • there is a single input INPUT.mkv;
  • there are no input options for this input;
  • there is a single output OUTPUT.mp4;
  • there are two output options for this output:
    • -map 0:1 selects the input stream to be used - from input with index 0 (i.e. the first one) the stream with index 1 (i.e. the second one);
    • -c copy selects the copy encoder, i.e. streamcopy with no decoding or encoding.

Streamcopy is useful for changing the elementary stream count, container format, or modifying container-level metadata. Since there is no decoding or encoding, it is very fast and there is no quality loss. However, it might not work in some cases because of a variety of factors (e.g. certain information required by the target container is not available in the source). Applying filters is obviously also impossible, since filters work on decoded frames.

More complex streamcopy scenarios can be constructed - e.g. combining streams from two input files into a single output:

┌──────────┬────────────────────┐         ┌────────────────────┬───────────┐
│ demuxer 0│                    │ packets │                    │   muxer   │
╞══════════╡elementary stream 0 ├────────⮞│elementary stream 0 ╞═══════════╡
│INPUT0.mkv│                    │         │                    │OUTPUT.mp4 │
└──────────┴────────────────────┘         ├────────────────────┤           │
┌──────────┬────────────────────┐         │                    │           │
│ demuxer 1│                    │ packets │elementary stream 1 │           │
╞══════════╡elementary stream 0 ├────────⮞│                    │           │
│INPUT1.aac│                    │         └────────────────────┴───────────┘
└──────────┴────────────────────┘

that can be built by the commandline

ffmpeg -i INPUT0.mkv -i INPUT1.aac -map 0:0 -map 1:0 -c copy OUTPUT.mp4

The output -map option is used twice here, creating two streams in the output file - one fed by the first input and one by the second. The single instance of the -c option selects streamcopy for both of those streams. You could also use multiple instances of this option together with Stream specifiers to apply different values to each stream, as will be demonstrated in following sections.

A converse scenario is splitting multiple streams from a single input into multiple outputs:

┌──────────┬─────────────────────┐          ┌───────────────────┬───────────┐
│ demuxer  │                     │ packets  │                   │ muxer 0   │
╞══════════╡ elementary stream 0 ├─────────⮞│elementary stream 0╞═══════════╡
│          │                     │          │                   │OUTPUT0.mp4│
│INPUT.mkv ├─────────────────────┤          └───────────────────┴───────────┘
│          │                     │ packets  ┌───────────────────┬───────────┐
│          │ elementary stream 1 ├─────────⮞│                   │ muxer 1   │
│          │                     │          │elementary stream 0╞═══════════╡
└──────────┴─────────────────────┘          │                   │OUTPUT1.mp4│
                                            └───────────────────┴───────────┘

built with

ffmpeg -i INPUT.mkv -map 0:0 -c copy OUTPUT0.mp4 -map 0:1 -c copy OUTPUT1.mp4

Note how a separate instance of the -c option is needed for every output file even though their values are the same. This is because non-global options (which is most of them) only apply in the context of the file before which they are placed.

These examples can of course be further generalized into arbitrary remappings of any number of inputs into any number of outputs.

3.2 Trancoding

Transcoding is the process of decoding a stream and then encoding it again. Since encoding tends to be computationally expensive and in most cases degrades the stream quality (i.e. it is lossy), you should only transcode when you need to and perform streamcopy otherwise. Typical reasons to transcode are:

  • applying filters - e.g. resizing, deinterlacing, or overlaying video; resampling or mixing audio;
  • you want to feed the stream to something that cannot decode the original codec.

Note that ffmpeg will transcode all audio, video, and subtitle streams unless you specify -c copy for them.

Consider an example pipeline that reads an input file with one audio and one video stream, transcodes the video and copies the audio into a single output file. This can be schematically represented as follows

┌──────────┬─────────────────────┐
│ demuxer  │                     │       audio packets
╞══════════╡ stream 0 (audio)    ├─────────────────────────────────────╮
│          │                     │                                     │
│INPUT.mkv ├─────────────────────┤ video    ┌─────────┐     raw        │
│          │                     │ packets  │  video  │ video frames   │
│          │ stream 1 (video)    ├─────────⮞│ decoder ├──────────────╮ │
│          │                     │          │         │              │ │
└──────────┴─────────────────────┘          └─────────┘              │ │
                                                                     ▼ ▼
                                                                     │ │
┌──────────┬─────────────────────┐ video    ┌─────────┐              │ │
│ muxer    │                     │ packets  │  video  │              │ │
╞══════════╡ stream 0 (video)    │⮜─────────┤ encoder ├──────────────╯ │
│          │                     │          │(libx264)│                │
│OUTPUT.mp4├─────────────────────┤          └─────────┘                │
│          │                     │                                     │
│          │ stream 1 (audio)    │⮜────────────────────────────────────╯
│          │                     │
└──────────┴─────────────────────┘

and implemented with the following commandline:

ffmpeg -i INPUT.mkv -map 0:v -map 0:a -c:v libx264 -c:a copy OUTPUT.mp4

Note how it uses stream specifiers :v and :a to select input streams and apply different values of the -c option to them; see the Stream specifiers section for more details.

3.3 Filtering

When transcoding, audio and video streams can be filtered before encoding, with either a simple or complex filtergraph.

3.3.1 Simple filtergraphs

Simple filtergraphs are those that have exactly one input and output, both of the same type (audio or video). They are configured with the per-stream -filter option (with -vf and -af aliases for -filter:v (video) and -filter:a (audio) respectively). Note that simple filtergraphs are tied to their output stream, so e.g. if you have multiple audio streams, -af will create a separate filtergraph for each one.

Taking the trancoding example from above, adding filtering (and omitting audio, for clarity) makes it look like this:

┌──────────┬───────────────┐
│ demuxer  │               │          ┌─────────┐
╞══════════╡ video stream  │ packets  │  video  │ frames
│INPUT.mkv │               ├─────────⮞│ decoder ├─────⮞───╮
│          │               │          └─────────┘         │
└──────────┴───────────────┘                              │
                                  ╭───────────⮜───────────╯
                                  │   ┌────────────────────────┐
                                  │   │  simple filtergraph    │
                                  │   ╞════════════════════════╡
                                  │   │  ┌───────┐  ┌───────┐  │
                                  ╰──⮞├─⮞│ yadif ├─⮞│ scale ├─⮞├╮
                                      │  └───────┘  └───────┘  ││
                                      └────────────────────────┘│
                                                                │
                                                                │
┌──────────┬───────────────┐ video    ┌─────────┐               │
│ muxer    │               │ packets  │  video  │               │
╞══════════╡ video stream  │⮜─────────┤ encoder ├───────⮜───────╯
│OUTPUT.mp4│               │          │         │
│          │               │          └─────────┘
└──────────┴───────────────┘

3.3.2 Complex filtergraphs

Complex filtergraphs are those which cannot be described as simply a linear processing chain applied to one stream. This is the case, for example, when the graph has more than one input and/or output, or when output stream type is different from input. Complex filtergraphs are configured with the -filter_complex option. Note that this option is global, since a complex filtergraph, by its nature, cannot be unambiguously associated with a single stream or file. Each instance of -filter_complex creates a new complex filtergraph, and there can be any number of them.

A trivial example of a complex filtergraph is the overlay filter, which has two video inputs and one video output, containing one video overlaid on top of the other. Its audio counterpart is the amix filter.

3.4 Loopback decoders

While decoders are normally associated with demuxer streams, it is also possible to create "loopback" decoders that decode the output from some encoder and allow it to be fed back to complex filtergraphs. This is done with the -dec directive, which takes as a parameter the index of the output stream that should be decoded. Every such directive creates a new loopback decoder, indexed with successive integers starting at zero. These indices should then be used to refer to loopback decoders in complex filtergraph link labels, as described in the documentation for -filter_complex.

Decoding AVOptions can be passed to loopback decoders by placing them before -dec, analogously to input/output options.

E.g. the following example:

ffmpeg -i INPUT                                        \
  -map 0:v:0 -c:v libx264 -crf 45 -f null -            \
  -threads 3 -dec 0:0                                  \
  -filter_complex '[0:v][dec:0]hstack[stack]'          \
  -map '[stack]' -c:v ffv1 OUTPUT

reads an input video and

  • (line 2) encodes it with libx264 at low quality;
  • (line 3) decodes this encoded stream using 3 threads;
  • (line 4) places decoded video side by side with the original input video;
  • (line 5) combined video is then losslessly encoded and written into OUTPUT.

Such a transcoding pipeline can be represented with the following diagram:

┌──────────┬───────────────┐
│ demuxer  │               │   ┌─────────┐            ┌─────────┐    ┌────────────────────┐
╞══════════╡ video stream  │   │  video  │            │  video  │    │ null muxer         │
│   INPUT  │               ├──⮞│ decoder ├──┬────────⮞│ encoder ├─┬─⮞│(discards its input)│
│          │               │   └─────────┘  │         │(libx264)│ │  └────────────────────┘
└──────────┴───────────────┘                │         └─────────┘ │
                                 ╭───────⮜──╯   ┌─────────┐       │
                                 │              │loopback │       │
                                 │ ╭─────⮜──────┤ decoder ├────⮜──╯
                                 │ │            └─────────┘
                                 │ │
                                 │ │
                                 │ │  ┌───────────────────┐
                                 │ │  │complex filtergraph│
                                 │ │  ╞═══════════════════╡
                                 │ │  │  ┌─────────────┐  │
                                 ╰─╫─⮞├─⮞│   hstack    ├─⮞├╮
                                   ╰─⮞├─⮞│             │  ││
                                      │  └─────────────┘  ││
                                      └───────────────────┘│
                                                           │
┌──────────┬───────────────┐  ┌─────────┐                  │
│ muxer    │               │  │  video  │                  │
╞══════════╡ video stream  │⮜─┤ encoder ├───────⮜──────────╯
│  OUTPUT  │               │  │ (ffv1)  │
│          │               │  └─────────┘
└──────────┴───────────────┘

4 Stream selection

ffmpeg provides the -map option for manual control of stream selection in each output file. Users can skip -map and let ffmpeg perform automatic stream selection as described below. The -vn / -an / -sn / -dn options can be used to skip inclusion of video, audio, subtitle and data streams respectively, whether manually mapped or automatically selected, except for those streams which are outputs of complex filtergraphs.

4.1 Description

The sub-sections that follow describe the various rules that are involved in stream selection. The examples that follow next show how these rules are applied in practice.

While every effort is made to accurately reflect the behavior of the program, FFmpeg is under continuous development and the code may have changed since the time of this writing.

4.1.1 Automatic stream selection

In the absence of any map options for a particular output file, ffmpeg inspects the output format to check which type of streams can be included in it, viz. video, audio and/or subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available, from among all the inputs.

It will select that stream based upon the following criteria:

  • for video, it is the stream with the highest resolution,
  • for audio, it is the stream with the most channels,
  • for subtitles, it is the first subtitle stream found but there’s a caveat. The output format’s default subtitle encoder can be either text-based or image-based, and only a subtitle stream of the same type will be chosen.

In the case where several streams of the same type rate equally, the stream with the lowest index is chosen.

Data or attachment streams are not automatically selected and can only be included using -map.

4.1.2 Manual stream selection

When -map is used, only user-mapped streams are included in that output file, with one possible exception for filtergraph outputs described below.

4.1.3 Complex filtergraphs

If there are any complex filtergraph output streams with unlabeled pads, they will be added to the first output file. This will lead to a fatal error if the stream type is not supported by the output format. In the absence of the map option, the inclusion of these streams leads to the automatic stream selection of their types being skipped. If map options are present, these filtergraph streams are included in addition to the mapped streams.

Complex filtergraph output streams with labeled pads must be mapped once and exactly once.

4.1.4 Stream handling

Stream handling is independent of stream selection, with an exception for subtitles described below. Stream handling is set via the -codec option addressed to streams within a specific output file. In particular, codec options are applied by ffmpeg after the stream selection process and thus do not influence the latter. If no -codec option is specified for a stream type, ffmpeg will select the default encoder registered by the output file muxer.

An exception exists for subtitles. If a subtitle encoder is specified for an output file, the first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate if the specified encoder can convert the selected stream or if the converted stream is acceptable within the output format. This applies generally as well: when the user sets an encoder manually, the stream selection process cannot check if the encoded stream can be muxed into the output file. If it cannot, ffmpeg will abort and all output files will fail to be processed.

4.2 Examples

The following examples illustrate the behavior, quirks and limitations of ffmpeg’s stream selection methods.

They assume the following three input files.

input file 'A.avi'
      stream 0: video 640x360
      stream 1: audio 2 channels

input file 'B.mp4'
      stream 0: video 1920x1080
      stream 1: audio 2 channels
      stream 2: subtitles (text)
      stream 3: audio 5.1 channels
      stream 4: subtitles (text)

input file 'C.mkv'
      stream 0: video 1280x720
      stream 1: audio 2 channels
      stream 2: subtitles (image)

Example: automatic stream selection

ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov

There are three output files specified, and for the first two, no -map options are set, so ffmpeg will select streams for these two files automatically.

out1.mkv is a Matroska container file and accepts video, audio and subtitle streams, so ffmpeg will try to select one of each type.
For video, it will select stream 0 from B.mp4, which has the highest resolution among all the input video streams.
For audio, it will select stream 3 from B.mp4, since it has the greatest number of channels.
For subtitles, it will select stream 2 from B.mp4, which is the first subtitle stream from among A.avi and B.mp4.

out2.wav accepts only audio streams, so only stream 3 from B.mp4 is selected.

For out3.mov, since a -map option is set, no automatic stream selection will occur. The -map 1:a option will select all audio streams from the second input B.mp4. No other streams will be included in this output file.

For the first two outputs, all included streams will be transcoded. The encoders chosen will be the default ones registered by each output format, which may not match the codec of the selected input streams.

For the third output, codec option for audio streams has been set to copy, so no decoding-filtering-encoding operations will occur, or can occur. Packets of selected streams shall be conveyed from the input file and muxed within the output file.

Example: automatic subtitles selection

ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv

Although out1.mkv is a Matroska container file which accepts subtitle streams, only a video and audio stream shall be selected. The subtitle stream of C.mkv is image-based and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation for the subtitles is expected to fail and hence the stream isn’t selected. However, in out2.mkv, a subtitle encoder is specified in the command and so, the subtitle stream is selected, in addition to the video stream. The presence of -an disables audio stream selection for out2.mkv.

Example: unlabeled filtergraph outputs

ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt

A filtergraph is setup here using the -filter_complex option and consists of a single video filter. The overlay filter requires exactly two video inputs, but none are specified, so the first two available video streams are used, those of A.avi and C.mkv. The output pad of the filter has no label and so is sent to the first output file out1.mp4. Due to this, automatic selection of the video stream is skipped, which would have selected the stream in B.mp4. The audio stream with most channels viz. stream 3 in B.mp4, is chosen automatically. No subtitle stream is chosen however, since the MP4 format has no default subtitle encoder registered, and the user hasn’t specified a subtitle encoder.

The 2nd output file, out2.srt, only accepts text-based subtitle streams. So, even though the first subtitle stream available belongs to C.mkv, it is image-based and hence skipped. The selected stream, stream 2 in B.mp4, is the first text-based subtitle stream.

Example: labeled filtergraph outputs

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
       -map '[outv]' -an        out1.mp4 \
                                out2.mkv \
       -map '[outv]' -map 1:a:0 out3.mkv

The above command will fail, as the output pad labelled [outv] has been mapped twice. None of the output files shall be processed.

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
       -an        out1.mp4 \
                  out2.mkv \
       -map 1:a:0 out3.mkv

This command above will also fail as the hue filter output has a label, [outv], and hasn’t been mapped anywhere.

The command should be modified as follows,

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
        -map '[outv1]' -an        out1.mp4 \
                                  out2.mkv \
        -map '[outv2]' -map 1:a:0 out3.mkv

The video stream from B.mp4 is sent to the hue filter, whose output is cloned once using the split filter, and both outputs labelled. Then a copy each is mapped to the first and third output files.

The overlay filter, requiring two video inputs, uses the first two unused video streams. Those are the streams from A.avi and C.mkv. The overlay output isn’t labelled, so it is sent to the first output file out1.mp4, regardless of the presence of the -map option.

The aresample filter is sent the first unused audio stream, that of A.avi. Since this filter output is also unlabelled, it too is mapped to the first output file. The presence of -an only suppresses automatic or manual stream selection of audio streams, not outputs sent from filtergraphs. Both these mapped streams shall be ordered before the mapped stream in out1.mp4.

The video, audio and subtitle streams mapped to out2.mkv are entirely determined by automatic stream selection.

out3.mkv consists of the cloned video output from the hue filter and the first audio stream from B.mp4.

5 Options

All the numerical options, if not specified otherwise, accept a string representing a number as input, which may be followed by one of the SI unit prefixes, for example: ’K’, ’M’, or ’G’.

If ’i’ is appended to the SI unit prefix, the complete prefix will be interpreted as a unit prefix for binary multiples, which are based on powers of 1024 instead of powers of 1000. Appending ’B’ to the SI unit prefix multiplies the value by 8. This allows using, for example: ’KB’, ’MiB’, ’G’ and ’B’ as number suffixes.

Options which do not take arguments are boolean options, and set the corresponding value to true. They can be set to false by prefixing the option name with "no". For example using "-nofoo" will set the boolean option with name "foo" to false.

Options that take arguments support a special syntax where the argument given on the command line is interpreted as a path to the file from which the actual argument value is loaded. To use this feature, add a forward slash ’/’ immediately before the option name (after the leading dash). E.g.

ffmpeg -i INPUT -/filter:v filter.script OUTPUT

will load a filtergraph description from the file named filter.script.

5.1 Stream specifiers

Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to precisely specify which stream(s) a given option belongs to.

A stream specifier is a string generally appended to the option name and separated from it by a colon. E.g. -codec:a:1 ac3 contains the a:1 stream specifier, which matches the second audio stream. Therefore, it would select the ac3 codec for the second audio stream.

A stream specifier can match several streams, so that the option is applied to all of them. E.g. the stream specifier in -b:a 128k matches all audio streams.

An empty stream specifier matches all streams. For example, -codec copy or -codec: copy would copy all the streams without reencoding.

Possible forms of stream specifiers are:

stream_index

Matches the stream with this index. E.g. -threads:1 4 would set the thread count for the second stream to 4. If stream_index is used as an additional stream specifier (see below), then it selects stream number stream_index from the matching streams. Stream numbering is based on the order of the streams as detected by libavformat except when a stream group specifier or program ID is also specified. In this case it is based on the ordering of the streams in the group or program.

stream_type[:additional_stream_specifier]

stream_type is one of following: ’v’ or ’V’ for video, ’a’ for audio, ’s’ for subtitle, ’d’ for data, and ’t’ for attachments. ’v’ matches all video streams, ’V’ only matches video streams which are not attached pictures, video thumbnails or cover arts. If additional_stream_specifier is used, then it matches streams which both have this type and match the additional_stream_specifier. Otherwise, it matches all streams of the specified type.

g:group_specifier[:additional_stream_specifier]

Matches streams which are in the group with the specifier group_specifier. if additional_stream_specifier is used, then it matches streams which both are part of the group and match the additional_stream_specifier. group_specifier may be one of the following:

group_index

Match the stream with this group index.

#group_id or i:group_id

Match the stream with this group id.

p:program_id[:additional_stream_specifier]

Matches streams which are in the program with the id program_id. If additional_stream_specifier is used, then it matches streams which both are part of the program and match the additional_stream_specifier.

#stream_id or i:stream_id

Match the stream by stream id (e.g. PID in MPEG-TS container).

m:key[:value]

Matches streams with the metadata tag key having the specified value. If value is not given, matches streams that contain the given tag with any value.

disp:dispositions[:additional_stream_specifier]

Matches streams with the given disposition(s). dispositions is a list of one or more dispositions (as printed by the -dispositions option) joined with ’+’.

u

Matches streams with usable configuration, the codec must be defined and the essential information such as video dimension or audio sample rate must be present.

Note that in ffmpeg, matching by metadata will only work properly for input files.

5.2 Generic options

These options are shared amongst the ff* tools.

-L

Show license.

-h, -?, -help, --help [arg]

Show help. An optional parameter may be specified to print help about a specific item. If no argument is specified, only basic (non advanced) tool options are shown.

Possible values of arg are:

long

Print advanced tool options in addition to the basic tool options.

full

Print complete list of options, including shared and private options for encoders, decoders, demuxers, muxers, filters, etc.

decoder=decoder_name

Print detailed information about the decoder named decoder_name. Use the -decoders option to get a list of all decoders.

encoder=encoder_name

Print detailed information about the encoder named encoder_name. Use the -encoders option to get a list of all encoders.

demuxer=demuxer_name

Print detailed information about the demuxer named demuxer_name. Use the -formats option to get a list of all demuxers and muxers.

muxer=muxer_name

Print detailed information about the muxer named muxer_name. Use the -formats option to get a list of all muxers and demuxers.

filter=filter_name

Print detailed information about the filter named filter_name. Use the -filters option to get a list of all filters.

bsf=bitstream_filter_name

Print detailed information about the bitstream filter named bitstream_filter_name. Use the -bsfs option to get a list of all bitstream filters.

protocol=protocol_name

Print detailed information about the protocol named protocol_name. Use the -protocols option to get a list of all protocols.

-version

Show version.

-buildconf

Show the build configuration, one option per line.

-formats

Show available formats (including devices).

-demuxers

Show available demuxers.

-muxers

Show available muxers.

-devices

Show available devices.

-codecs

Show all codecs known to libavcodec.

Note that the term ’codec’ is used throughout this documentation as a shortcut for what is more correctly called a media bitstream format.

-decoders

Show available decoders.

-encoders

Show all available encoders.

-bsfs

Show available bitstream filters.

-protocols

Show available protocols.

-filters

Show available libavfilter filters.

-pix_fmts

Show available pixel formats.

-sample_fmts

Show available sample formats.

-layouts

Show channel names and standard channel layouts.

-dispositions

Show stream dispositions.

-colors

Show recognized color names.

-sources device[,opt1=val1[,opt2=val2]...]

Show autodetected sources of the input device. Some devices may provide system-dependent source names that cannot be autodetected. The returned list cannot be assumed to be always complete.

ffmpeg -sources pulse,server=192.168.0.4
-sinks device[,opt1=val1[,opt2=val2]...]

Show autodetected sinks of the output device. Some devices may provide system-dependent sink names that cannot be autodetected. The returned list cannot be assumed to be always complete.

ffmpeg -sinks pulse,server=192.168.0.4
-loglevel [flags+]loglevel | -v [flags+]loglevel

Set logging level and flags used by the library.

The optional flags prefix can consist of the following values:

repeat

Indicates that repeated log output should not be compressed to the first line and the "Last message repeated n times" line will be omitted.

level

Indicates that log output should add a [level] prefix to each message line. This can be used as an alternative to log coloring, e.g. when dumping the log to file.

Flags can also be used alone by adding a ’+’/’-’ prefix to set/reset a single flag without affecting other flags or changing loglevel. When setting both flags and loglevel, a ’+’ separator is expected between the last flags value and before loglevel.

loglevel is a string or a number containing one of the following values:

quiet, -8

Show nothing at all; be silent.

panic, 0

Only show fatal errors which could lead the process to crash, such as an assertion failure. This is not currently used for anything.

fatal, 8

Only show fatal errors. These are errors after which the process absolutely cannot continue.

error, 16

Show all errors, including ones which can be recovered from.

warning, 24

Show all warnings and errors. Any message related to possibly incorrect or unexpected events will be shown.

info, 32

Show informative messages during processing. This is in addition to warnings and errors. This is the default value.

verbose, 40

Same as info, except more verbose.

debug, 48

Show everything, including debugging information.

trace, 56

For example to enable repeated log output, add the level prefix, and set loglevel to verbose:

ffmpeg -loglevel repeat+level+verbose -i input output

Another example that enables repeated log output without affecting current state of level prefix flag or loglevel:

ffmpeg [...] -loglevel +repeat

By default the program logs to stderr. If coloring is supported by the terminal, colors are used to mark errors and warnings. Log coloring can be disabled setting the environment variable AV_LOG_FORCE_NOCOLOR, or can be forced setting the environment variable AV_LOG_FORCE_COLOR.

-report

Dump full command line and log output to a file named program-YYYYMMDD-HHMMSS.log in the current directory. This file can be useful for bug reports. It also implies -loglevel debug.

Setting the environment variable FFREPORT to any value has the same effect. If the value is a ’:’-separated key=value sequence, these options will affect the report; option values must be escaped if they contain special characters or the options delimiter ’:’ (see the “Quoting and escaping” section in the ffmpeg-utils manual).

The following options are recognized:

file

set the file name to use for the report; %p is expanded to the name of the program, %t is expanded to a timestamp, %% is expanded to a plain %

level

set the log verbosity level using a numerical value (see -loglevel).

For example, to output a report to a file named ffreport.log using a log level of 32 (alias for log level info):

FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output

Errors in parsing the environment variable are not fatal, and will not appear in the report.

-hide_banner

Suppress printing banner.

All FFmpeg tools will normally show a copyright notice, build options and library versions. This option can be used to suppress printing this information.

-cpuflags flags (global)

Allows setting and clearing cpu flags. This option is intended for testing. Do not use it unless you know what you’re doing.

ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...

Possible flags for this option are:

x86
mmx
mmxext
sse
sse2
sse2slow
sse3
sse3slow
ssse3
atom
sse4.1
sse4.2
avx
avx2
xop
fma3
fma4
3dnow
3dnowext
bmi1
bmi2
cmov
ARM
armv5te
armv6
armv6t2
vfp
vfpv3
neon
setend
AArch64
armv8
vfp
neon
PowerPC
altivec
Specific Processors
pentium2
pentium3
pentium4
k6
k62
athlon
athlonxp
k8
-cpucount count (global)

Override detection of CPU count. This option is intended for testing. Do not use it unless you know what you’re doing.

ffmpeg -cpucount 2
-max_alloc bytes

Set the maximum size limit for allocating a block on the heap by ffmpeg’s family of malloc functions. Exercise extreme caution when using this option. Don’t use if you do not understand the full consequence of doing so. Default is INT_MAX.

5.3 AVOptions

These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the list of available AVOptions, use the -help option. They are separated into two categories:

generic

These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs.

private

These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs.

For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the id3v2_version private option of the MP3 muxer:

ffmpeg -i input.flac -id3v2_version 3 out.mp3

All codec AVOptions are per-stream, and thus a stream specifier should be attached to them:

ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4

In the above example, a multichannel audio stream is mapped twice for output. The first instance is encoded with codec ac3 and bitrate 640k. The second instance is downmixed to 2 channels and encoded with codec aac. A bitrate of 128k is specified for it using absolute index of the output stream.

Note: the -nooption syntax cannot be used for boolean AVOptions, use -option 0/-option 1.

Note: the old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name is now obsolete and will be removed soon.

5.4 Main options

-f fmt (input/output)

Force input or output file format. The format is normally auto detected for input files and guessed from the file extension for output files, so this option is not needed in most cases.

-i url (input)

input file url

-y (global)

Overwrite output files without asking.

-n (global)

Do not overwrite output files, and exit immediately if a specified output file already exists.

-stream_loop number (input)

Set number of times input stream shall be looped. Loop 0 means no loop, loop -1 means infinite loop.

-recast_media (global)

Allow forcing a decoder of a different media type than the one detected or designated by the demuxer. Useful for decoding media data muxed as data streams.

-c[:stream_specifier] codec (input/output,per-stream)
-codec[:stream_specifier] codec (input/output,per-stream)

Select an encoder (when used before an output file) or a decoder (when used before an input file) for one or more streams. codec is the name of a decoder/encoder or a special value copy (output only) to indicate that the stream is not to be re-encoded.

For example

ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT

encodes all video streams with libx264 and copies all audio streams.

For each stream, the last matching c option is applied, so

ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT

will copy all the streams except the second video, which will be encoded with libx264, and the 138th audio, which will be encoded with libvorbis.

-t duration (input/output)

When used as an input option (before -i), limit the duration of data read from the input file.

When used as an output option (before an output url), stop writing the output after its duration reaches duration.

duration must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.

-to and -t are mutually exclusive and -t has priority.

-to position (input/output)

Stop writing the output or reading the input at position. position must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.

-to and -t are mutually exclusive and -t has priority.

-fs limit_size (output)

Set the file size limit, expressed in bytes. No further chunk of bytes is written after the limit is exceeded. The size of the output file is slightly more than the requested file size.

-ss position (input/output)

When used as an input option (before -i), seeks in this input file to position. Note that in most formats it is not possible to seek exactly, so ffmpeg will seek to the closest seek point before position. When transcoding and -accurate_seek is enabled (the default), this extra segment between the seek point and position will be decoded and discarded. When doing stream copy or when -noaccurate_seek is used, it will be preserved.

When used as an output option (before an output url), decodes but discards input until the timestamps reach position.

position must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.

-sseof position (input)

Like the -ss option but relative to the "end of file". That is negative values are earlier in the file, 0 is at EOF.

-isync input_index (input)

Assign an input as a sync source.

This will take the difference between the start times of the target and reference inputs and offset the timestamps of the target file by that difference. The source timestamps of the two inputs should derive from the same clock source for expected results. If copyts is set then start_at_zero must also be set. If either of the inputs has no starting timestamp then no sync adjustment is made.

Acceptable values are those that refer to a valid ffmpeg input index. If the sync reference is the target index itself or -1, then no adjustment is made to target timestamps. A sync reference may not itself be synced to any other input.

Default value is -1.

-itsoffset offset (input)

Set the input time offset.

offset must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.

The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by the time duration specified in offset.

-itsscale scale (input,per-stream)

Rescale input timestamps. scale should be a floating point number.

-timestamp date (output)

Set the recording timestamp in the container.

date must be a date specification, see (ffmpeg-utils)the Date section in the ffmpeg-utils(1) manual.

-metadata[:metadata_specifier] key=value (output,per-metadata)

Set a metadata key/value pair.

An optional metadata_specifier may be given to set metadata on streams, chapters or programs. See -map_metadata documentation for details.

This option overrides metadata set with -map_metadata. It is also possible to delete metadata by using an empty value.

For example, for setting the title in the output file:

ffmpeg -i in.avi -metadata title="my title" out.flv

To set the language of the first audio stream:

ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
-disposition[:stream_specifier] value (output,per-stream)

Sets the disposition flags for a stream.

Default value: by default, all disposition flags are copied from the input stream, unless the output stream this option applies to is fed by a complex filtergraph - in that case no disposition flags are set by default.

value is a sequence of disposition flags separated by ’+’ or ’-’. A ’+’ prefix adds the given disposition, ’-’ removes it. If the first flag is also prefixed with ’+’ or ’-’, the resulting disposition is the default value updated by value. If the first flag is not prefixed, the resulting disposition is value. It is also possible to clear the disposition by setting it to 0.

If no -disposition options were specified for an output file, ffmpeg will automatically set the ’default’ disposition flag on the first stream of each type, when there are multiple streams of this type in the output file and no stream of that type is already marked as default.

The -dispositions option lists the known disposition flags.

For example, to make the second audio stream the default stream:

ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv

To make the second subtitle stream the default stream and remove the default disposition from the first subtitle stream:

ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv

To add an embedded cover/thumbnail:

ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4

To add the ’original’ and remove the ’comment’ disposition flag from the first audio stream without removing its other disposition flags:

ffmpeg -i in.mkv -c copy -disposition:a:0 +original-comment out.mkv

To remove the ’original’ and add the ’comment’ disposition flag to the first audio stream without removing its other disposition flags:

ffmpeg -i in.mkv -c copy -disposition:a:0 -original+comment out.mkv

To set only the ’original’ and ’comment’ disposition flags on the first audio stream (and remove its other disposition flags):

ffmpeg -i in.mkv -c copy -disposition:a:0 original+comment out.mkv

To remove all disposition flags from the first audio stream:

ffmpeg -i in.mkv -c copy -disposition:a:0 0 out.mkv

Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.

-program [title=title:][program_num=program_num:]st=stream[:st=stream...] (output)

Creates a program with the specified title, program_num and adds the specified stream(s) to it.

-stream_group [map=input_file_id=stream_group][type=type:]st=stream[:st=stream][:stg=stream_group][:id=stream_group_id...] (output)

Creates a stream group of the specified type and stream_group_id, or by mapping an input group, adding the specified stream(s) and/or previously defined stream_group(s) to it.

type can be one of the following:

iamf_audio_element

Groups streams that belong to the same IAMF Audio Element

For this group type, the following options are available

audio_element_type

The Audio Element type. The following values are supported:

channel

Scalable channel audio representation

scene

Ambisonics representation

demixing

Demixing information used to reconstruct a scalable channel audio representation. This option must be separated from the rest with a ’,’, and takes the following key=value options

parameter_id

An identifier parameters blocks in frames may refer to

dmixp_mode

A pre-defined combination of demixing parameters

recon_gain

Recon gain information used to reconstruct a scalable channel audio representation. This option must be separated from the rest with a ’,’, and takes the following key=value options

parameter_id

An identifier parameters blocks in frames may refer to

layer

A layer defining a Channel Layout in the Audio Element. This option must be separated from the rest with a ’,’. Several ’,’ separated entries can be defined, and at least one must be set.

It takes the following ":"-separated key=value options

ch_layout

The layer’s channel layout

flags

The following flags are available:

recon_gain

Wether to signal if recon_gain is present as metadata in parameter blocks within frames

output_gain
output_gain_flags

Which channels output_gain applies to. The following flags are available:

FL
FR
BL
BR
TFL
TFR
ambisonics_mode

The ambisonics mode. This has no effect if audio_element_type is set to channel.

The following values are supported:

mono

Each ambisonics channel is coded as an individual mono stream in the group

default_w

Default weight value

iamf_mix_presentation

Groups streams that belong to all IAMF Audio Element the same IAMF Mix Presentation references

For this group type, the following options are available

submix

A sub-mix within the Mix Presentation. This option must be separated from the rest with a ’,’. Several ’,’ separated entries can be defined, and at least one must be set.

It takes the following ":"-separated key=value options

parameter_id

An identifier parameters blocks in frames may refer to, for post-processing the mixed audio signal to generate the audio signal for playback

parameter_rate

The sample rate duration fields in parameters blocks in frames that refer to this parameter_id are expressed as

default_mix_gain

Default mix gain value to apply when there are no parameter blocks sharing the same parameter_id for a given frame

element

References an Audio Element used in this Mix Presentation to generate the final output audio signal for playback. This option must be separated from the rest with a ’|’. Several ’|’ separated entries can be defined, and at least one must be set.

It takes the following ":"-separated key=value options:

stg

The stream_group_id for an Audio Element which this sub-mix refers to

parameter_id

An identifier parameters blocks in frames may refer to, for applying any processing to the referenced and rendered Audio Element before being summed with other processed Audio Elements

parameter_rate

The sample rate duration fields in parameters blocks in frames that refer to this parameter_id are expressed as

default_mix_gain

Default mix gain value to apply when there are no parameter blocks sharing the same parameter_id for a given frame

annotations

A key=value string describing the sub-mix element where "key" is a string conforming to BCP-47 that specifies the language for the "value" string. "key" must be the same as the one in the mix’s annotations

headphones_rendering_mode

Indicates whether the input channel-based Audio Element is rendered to stereo loudspeakers or spatialized with a binaural renderer when played back on headphones. This has no effect if the referenced Audio Element’s audio_element_type is set to channel.

The following values are supported:

stereo
binaural
layout

Specifies the layouts for this sub-mix on which the loudness information was measured. This option must be separated from the rest with a ’|’. Several ’|’ separated entries can be defined, and at least one must be set.

It takes the following ":"-separated key=value options:

layout_type
loudspeakers

The layout follows the loudspeaker sound system convention of ITU-2051-3.

binaural

The layout is binaural.

sound_system

Channel layout matching one of Sound Systems A to J of ITU-2051-3, plus 7.1.2 and 3.1.2 This has no effect if layout_type is set to binaural.

integrated_loudness

The program integrated loudness information, as defined in ITU-1770-4.

digital_peak

The digital (sampled) peak value of the audio signal, as defined in ITU-1770-4.

true_peak

The true peak of the audio signal, as defined in ITU-1770-4.

dialog_anchored_loudness

The Dialogue loudness information, as defined in ITU-1770-4.

album_anchored_loudness

The Album loudness information, as defined in ITU-1770-4.

annotations

A key=value string string describing the mix where "key" is a string conforming to BCP-47 that specifies the language for the "value" string. "key" must be the same as the ones in all sub-mix element’s annotationss

E.g. to create an scalable 5.1 IAMF file from several WAV input files

ffmpeg -i front.wav -i back.wav -i center.wav -i lfe.wav
-map 0:0 -map 1:0 -map 2:0 -map 3:0 -c:a opus
-stream_group type=iamf_audio_element:id=1:st=0:st=1:st=2:st=3,
demixing=parameter_id=998,
recon_gain=parameter_id=101,
layer=ch_layout=stereo,
layer=ch_layout=5.1,
-stream_group type=iamf_mix_presentation:id=2:stg=0:annotations=en-us=Mix_Presentation,
submix=parameter_id=100:parameter_rate=48000|element=stg=0:parameter_id=100:annotations=en-us=Scalable_Submix|layout=sound_system=stereo|layout=sound_system=5.1
-streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 output.iamf

To copy the two stream groups (Audio Element and Mix Presentation) from an input IAMF file with four streams into an mp4 output

ffmpeg -i input.iamf -c:a copy -stream_group map=0=0:st=0:st=1:st=2:st=3 -stream_group map=0=1:stg=0
-streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 output.mp4
-target type (output)

Specify target file type (vcd, svcd, dvd, dv, dv50). type may be prefixed with pal-, ntsc- or film- to use the corresponding standard. All the format options (bitrate, codecs, buffer sizes) are then set automatically. You can just type:

ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg

Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in:

ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg

The parameters set for each target are as follows.

VCD

pal:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x288 -r 25
-codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

ntsc:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 30000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

film:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 24000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

SVCD

pal:
-f svcd -packetsize 2324
-s 480x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

ntsc:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

film:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

DVD

pal:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

ntsc:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

film:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

DV

pal:
-f dv
-s 720x576 -pix_fmt yuv420p -r 25
-ar 48000 -ac 2

ntsc:
-f dv
-s 720x480 -pix_fmt yuv411p -r 30000/1001
-ar 48000 -ac 2

film:
-f dv
-s 720x480 -pix_fmt yuv411p -r 24000/1001
-ar 48000 -ac 2

The dv50 target is identical to the dv target except that the pixel format set is yuv422p for all three standards.

Any user-set value for a parameter above will override the target preset value. In that case, the output may not comply with the target standard.

-dn (input/output)

As an input option, blocks all data streams of a file from being filtered or being automatically selected or mapped for any output. See -discard option to disable streams individually.

As an output option, disables data recording i.e. automatic selection or mapping of any data stream. For full manual control see the -map option.

-dframes number (output)

Set the number of data frames to output. This is an obsolete alias for -frames:d, which you should use instead.

-frames[:stream_specifier] framecount (output,per-stream)

Stop writing to the stream after framecount frames.

-q[:stream_specifier] q (output,per-stream)
-qscale[:stream_specifier] q (output,per-stream)

Use fixed quality scale (VBR). The meaning of q/qscale is codec-dependent. If qscale is used without a stream_specifier then it applies only to the video stream, this is to maintain compatibility with previous behavior and as specifying the same codec specific value to 2 different codecs that is audio and video generally is not what is intended when no stream_specifier is used.

-filter[:stream_specifier] filtergraph (output,per-stream)

Create the filtergraph specified by filtergraph and use it to filter the stream.

filtergraph is a description of the filtergraph to apply to the stream, and must have a single input and a single output of the same type of the stream. In the filtergraph, the input is associated to the label in, and the output to the label out. See the ffmpeg-filters manual for more information about the filtergraph syntax.

See the -filter_complex option if you want to create filtergraphs with multiple inputs and/or outputs.

-reinit_filter[:stream_specifier] integer (input,per-stream)

This boolean option determines if the filtergraph(s) to which this stream is fed gets reinitialized when input frame parameters change mid-stream. This option is enabled by default as most video and all audio filters cannot handle deviation in input frame properties. Upon reinitialization, existing filter state is lost, like e.g. the frame count n reference available in some filters. Any frames buffered at time of reinitialization are lost. The properties where a change triggers reinitialization are, for video, frame resolution or pixel format; for audio, sample format, sample rate, channel count or channel layout.

-filter_threads nb_threads (global)

Defines how many threads are used to process a filter pipeline. Each pipeline will produce a thread pool with this many threads available for parallel processing. The default is the number of available CPUs.

-pre[:stream_specifier] preset_name (output,per-stream)

Specify the preset for matching stream(s).

-stats (global)

Log encoding progress/statistics as "info"-level log (see -loglevel). It is on by default, to explicitly disable it you need to specify -nostats.

-stats_period time (global)

Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.

-progress url (global)

Send program-friendly progress information to url.

Progress information is written periodically and at the end of the encoding process. It is made of "key=value" lines. key consists of only alphanumeric characters. The last key of a sequence of progress information is always "progress" with the value "continue" or "end".

The update period is set using -stats_period.

For example, log progress information to stdout:

ffmpeg -progress pipe:1 -i in.mkv out.mkv
-stdin

Enable interaction on standard input. On by default unless standard input is used as an input. To explicitly disable interaction you need to specify -nostdin.

Disabling interaction on standard input is useful, for example, if ffmpeg is in the background process group. Roughly the same result can be achieved with ffmpeg ... < /dev/null but it requires a shell.

-debug_ts (global)

Print timestamp/latency information. It is off by default. This option is mostly useful for testing and debugging purposes, and the output format may change from one version to another, so it should not be employed by portable scripts.

See also the option -fdebug ts.

-attach filename (output)

Add an attachment to the output file. This is supported by a few formats like Matroska for e.g. fonts used in rendering subtitles. Attachments are implemented as a specific type of stream, so this option will add a new stream to the file. It is then possible to use per-stream options on this stream in the usual way. Attachment streams created with this option will be created after all the other streams (i.e. those created with -map or automatic mappings).

Note that for Matroska you also have to set the mimetype metadata tag:

ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv

(assuming that the attachment stream will be third in the output file).

-dump_attachment[:stream_specifier] filename (input,per-stream)

Extract the matching attachment stream into a file named filename. If filename is empty, then the value of the filename metadata tag will be used.

E.g. to extract the first attachment to a file named ’out.ttf’:

ffmpeg -dump_attachment:t:0 out.ttf -i INPUT

To extract all attachments to files determined by the filename tag:

ffmpeg -dump_attachment:t "" -i INPUT

Technical note – attachments are implemented as codec extradata, so this option can actually be used to extract extradata from any stream, not just attachments.

5.5 Video Options

-vframes number (output)

Set the number of video frames to output. This is an obsolete alias for -frames:v, which you should use instead.

-r[:stream_specifier] fps (input/output,per-stream)

Set frame rate (Hz value, fraction or abbreviation).

As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate fps. This is not the same as the -framerate option used for some input formats like image2 or v4l2 (it used to be the same in older versions of FFmpeg). If in doubt use -framerate instead of the input option -r.

As an output option:

video encoding

Duplicate or drop frames right before encoding them to achieve constant output frame rate fps.

video streamcopy

Indicate to the muxer that fps is the stream frame rate. No data is dropped or duplicated in this case. This may produce invalid files if fps does not match the actual stream frame rate as determined by packet timestamps. See also the setts bitstream filter.

-fpsmax[:stream_specifier] fps (output,per-stream)

Set maximum frame rate (Hz value, fraction or abbreviation).

Clamps output frame rate when output framerate is auto-set and is higher than this value. Useful in batch processing or when input framerate is wrongly detected as very high. It cannot be set together with -r. It is ignored during streamcopy.

-s[:stream_specifier] size (input/output,per-stream)

Set frame size.

As an input option, this is a shortcut for the video_size private option, recognized by some demuxers for which the frame size is either not stored in the file or is configurable – e.g. raw video or video grabbers.

As an output option, this inserts the scale video filter to the end of the corresponding filtergraph. Please use the scale filter directly to insert it at the beginning or some other place.

The format is ‘wxh’ (default - same as source).

-aspect[:stream_specifier] aspect (output,per-stream)

Set the video display aspect ratio specified by aspect.

aspect can be a floating point number string, or a string of the form num:den, where num and den are the numerator and denominator of the aspect ratio. For example "4:3", "16:9", "1.3333", and "1.7777" are valid argument values.

If used together with -vcodec copy, it will affect the aspect ratio stored at container level, but not the aspect ratio stored in encoded frames, if it exists.

-display_rotation[:stream_specifier] rotation (input,per-stream)

Set video rotation metadata.

rotation is a decimal number specifying the amount in degree by which the video should be rotated counter-clockwise before being displayed.

This option overrides the rotation/display transform metadata stored in the file, if any. When the video is being transcoded (rather than copied) and -autorotate is enabled, the video will be rotated at the filtering stage. Otherwise, the metadata will be written into the output file if the muxer supports it.

If the -display_hflip and/or -display_vflip options are given, they are applied after the rotation specified by this option.

-display_hflip[:stream_specifier] (input,per-stream)

Set whether on display the image should be horizontally flipped.

See the -display_rotation option for more details.

-display_vflip[:stream_specifier] (input,per-stream)

Set whether on display the image should be vertically flipped.

See the -display_rotation option for more details.

-vn (input/output)

As an input option, blocks all video streams of a file from being filtered or being automatically selected or mapped for any output. See -discard option to disable streams individually.

As an output option, disables video recording i.e. automatic selection or mapping of any video stream. For full manual control see the -map option.

-vcodec codec (output)

Set the video codec. This is an alias for -codec:v.

-pass[:stream_specifier] n (output,per-stream)

Select the pass number (1 or 2). It is used to do two-pass video encoding. The statistics of the video are recorded in the first pass into a log file (see also the option -passlogfile), and in the second pass that log file is used to generate the video at the exact requested bitrate. On pass 1, you may just deactivate audio and set output to null, examples for Windows and Unix:

ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL
ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null
-passlogfile[:stream_specifier] prefix (output,per-stream)

Set two-pass log file name prefix to prefix, the default file name prefix is “ffmpeg2pass”. The complete file name will be PREFIX-N.log, where N is a number specific to the output stream

-vf filtergraph (output)

Create the filtergraph specified by filtergraph and use it to filter the stream.

This is an alias for -filter:v, see the -filter option.

-autorotate

Automatically rotate the video according to file metadata. Enabled by default, use -noautorotate to disable it.

-autoscale

Automatically scale the video according to the resolution of first frame. Enabled by default, use -noautoscale to disable it. When autoscale is disabled, all output frames of filter graph might not be in the same resolution and may be inadequate for some encoder/muxer. Therefore, it is not recommended to disable it unless you really know what you are doing. Disable autoscale at your own risk.

5.6 Advanced Video options

-pix_fmt[:stream_specifier] format (input/output,per-stream)

Set pixel format. Use -pix_fmts to show all the supported pixel formats. If the selected pixel format can not be selected, ffmpeg will print a warning and select the best pixel format supported by the encoder. If pix_fmt is prefixed by a +, ffmpeg will exit with an error if the requested pixel format can not be selected, and automatic conversions inside filtergraphs are disabled. If pix_fmt is a single +, ffmpeg selects the same pixel format as the input (or graph output) and automatic conversions are disabled.

-sws_flags flags (input/output)

Set default flags for the libswscale library. These flags are used by automatically inserted scale filters and those within simple filtergraphs, if not overridden within the filtergraph definition.

See the (ffmpeg-scaler)ffmpeg-scaler manual for a list of scaler options.

-rc_override[:stream_specifier] override (output,per-stream)

Rate control override for specific intervals, formatted as "int,int,int" list separated with slashes. Two first values are the beginning and end frame numbers, last one is quantizer to use if positive, or quality factor if negative.

-vstats

Dump video coding statistics to vstats_HHMMSS.log. See the vstats file format section for the format description.

-vstats_file file

Dump video coding statistics to file. See the vstats file format section for the format description.

-vstats_version file

Specify which version of the vstats format to use. Default is 2. See the vstats file format section for the format description.

-vtag fourcc/tag (output)

Force video tag/fourcc. This is an alias for -tag:v.

-force_key_frames[:stream_specifier] time[,time...] (output,per-stream)
-force_key_frames[:stream_specifier] expr:expr (output,per-stream)
-force_key_frames[:stream_specifier] source (output,per-stream)

force_key_frames can take arguments of the following form:

time[,time...]

If the argument consists of timestamps, ffmpeg will round the specified times to the nearest output timestamp as per the encoder time base and force a keyframe at the first frame having timestamp equal or greater than the computed timestamp. Note that if the encoder time base is too coarse, then the keyframes may be forced on frames with timestamps lower than the specified time. The default encoder time base is the inverse of the output framerate but may be set otherwise via -enc_time_base.

If one of the times is "chapters[delta]", it is expanded into the time of the beginning of all chapters in the file, shifted by delta, expressed as a time in seconds. This option can be useful to ensure that a seek point is present at a chapter mark or any other designated place in the output file.

For example, to insert a key frame at 5 minutes, plus key frames 0.1 second before the beginning of every chapter:

-force_key_frames 0:05:00,chapters-0.1
expr:expr

If the argument is prefixed with expr:, the string expr is interpreted like an expression and is evaluated for each frame. A key frame is forced in case the evaluation is non-zero.

The expression in expr can contain the following constants:

n

the number of current processed frame, starting from 0

n_forced

the number of forced frames

prev_forced_n

the number of the previous forced frame, it is NAN when no keyframe was forced yet

prev_forced_t

the time of the previous forced frame, it is NAN when no keyframe was forced yet

t

the time of the current processed frame

For example to force a key frame every 5 seconds, you can specify:

-force_key_frames expr:gte(t,n_forced*5)

To force a key frame 5 seconds after the time of the last forced one, starting from second 13:

-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
source

If the argument is source, ffmpeg will force a key frame if the current frame being encoded is marked as a key frame in its source. In cases where this particular source frame has to be dropped, enforce the next available frame to become a key frame instead.

Note that forcing too many keyframes is very harmful for the lookahead algorithms of certain encoders: using fixed-GOP options or similar would be more efficient.

-apply_cropping[:stream_specifier] source (input,per-stream)

Automatically crop the video after decoding according to file metadata. Default is all.

none (0)

Don’t apply any cropping metadata.

all (1)

Apply both codec and container level croppping. This is the default mode.

codec (2)

Apply codec level croppping.

container (3)

Apply container level croppping.

-copyinkf[:stream_specifier] (output,per-stream)

When doing stream copy, copy also non-key frames found at the beginning.

-init_hw_device type[=name][:device[,key=value...]]

Initialise a new hardware device of type type called name, using the given device parameters. If no name is specified it will receive a default name of the form "type%d".

The meaning of device and the following arguments depends on the device type:

cuda

device is the number of the CUDA device.

The following options are recognized:

primary_ctx

If set to 1, uses the primary device context instead of creating a new one.

Examples:

-init_hw_device cuda:1

Choose the second device on the system.

-init_hw_device cuda:0,primary_ctx=1

Choose the first device and use the primary device context.

dxva2

device is the number of the Direct3D 9 display adapter.

d3d11va

device is the number of the Direct3D 11 display adapter. If not specified, it will attempt to use the default Direct3D 11 display adapter or the first Direct3D 11 display adapter whose hardware VendorId is specified by ‘vendor_id’.

Examples:

-init_hw_device d3d11va

Create a d3d11va device on the default Direct3D 11 display adapter.

-init_hw_device d3d11va:1

Create a d3d11va device on the Direct3D 11 display adapter specified by index 1.

-init_hw_device d3d11va:,vendor_id=0x8086

Create a d3d11va device on the first Direct3D 11 display adapter whose hardware VendorId is 0x8086.

vaapi

device is either an X11 display name, a DRM render node or a DirectX adapter index. If not specified, it will attempt to open the default X11 display ($DISPLAY) and then the first DRM render node (/dev/dri/renderD128), or the default DirectX adapter on Windows.

The following options are recognized:

kernel_driver

When device is not specified, use this option to specify the name of the kernel driver associated with the desired device. This option is available only when the hardware acceleration method drm and vaapi are enabled.

vendor_id

When device and kernel_driver are not specified, use this option to specify the vendor id associated with the desired device. This option is available only when the hardware acceleration method drm and vaapi are enabled and kernel_driver is not specified.

Examples:

-init_hw_device vaapi

Create a vaapi device on the default device.

-init_hw_device vaapi:/dev/dri/renderD129

Create a vaapi device on DRM render node /dev/dri/renderD129.

-init_hw_device vaapi:1

Create a vaapi device on DirectX adapter 1.

-init_hw_device vaapi:,kernel_driver=i915

Create a vaapi device on a device associated with kernel driver ‘i915’.

-init_hw_device vaapi:,vendor_id=0x8086

Create a vaapi device on a device associated with vendor id ‘0x8086’.

vdpau

device is an X11 display name. If not specified, it will attempt to open the default X11 display ($DISPLAY).

qsv

device selects a value in ‘MFX_IMPL_*’. Allowed values are:

auto
sw
hw
auto_any
hw_any
hw2
hw3
hw4

If not specified, ‘auto_any’ is used. (Note that it may be easier to achieve the desired result for QSV by creating the platform-appropriate subdevice (‘dxva2’ or ‘d3d11va’ or ‘vaapi’) and then deriving a QSV device from that.)

The following options are recognized:

child_device

Specify a DRM render node on Linux or DirectX adapter on Windows.

child_device_type

Choose platform-appropriate subdevice type. On Windows ‘d3d11va’ is used as default subdevice type when --enable-libvpl is specified at configuration time, ‘dxva2’ is used as default subdevice type when --enable-libmfx is specified at configuration time. On Linux user can use ‘vaapi’ only as subdevice type.

Examples:

-init_hw_device qsv:hw,child_device=/dev/dri/renderD129

Create a QSV device with ‘MFX_IMPL_HARDWARE’ on DRM render node /dev/dri/renderD129.

-init_hw_device qsv:hw,child_device=1

Create a QSV device with ‘MFX_IMPL_HARDWARE’ on DirectX adapter 1.

-init_hw_device qsv:hw,child_device_type=d3d11va

Choose the GPU subdevice with type ‘d3d11va’ and create QSV device with ‘MFX_IMPL_HARDWARE’.

-init_hw_device qsv:hw,child_device_type=dxva2

Choose the GPU subdevice with type ‘dxva2’ and create QSV device with ‘MFX_IMPL_HARDWARE’.

-init_hw_device qsv:hw,child_device=1,child_device_type=d3d11va

Create a QSV device with ‘MFX_IMPL_HARDWARE’ on DirectX adapter 1 with subdevice type ‘d3d11va’.

-init_hw_device vaapi=va:/dev/dri/renderD129 -init_hw_device qsv=hw1@va

Create a VAAPI device called ‘va’ on /dev/dri/renderD129, then derive a QSV device called ‘hw1’ from device ‘va’.

opencl

device selects the platform and device as platform_index.device_index.

The set of devices can also be filtered using the key-value pairs to find only devices matching particular platform or device strings.

The strings usable as filters are:

platform_profile
platform_version
platform_name
platform_vendor
platform_extensions
device_name
device_vendor
driver_version
device_version
device_profile
device_extensions
device_type

The indices and filters must together uniquely select a device.

Examples:

-init_hw_device opencl:0.1

Choose the second device on the first platform.

-init_hw_device opencl:,device_name=Foo9000

Choose the device with a name containing the string Foo9000.

-init_hw_device opencl:1,device_type=gpu,device_extensions=cl_khr_fp16

Choose the GPU device on the second platform supporting the cl_khr_fp16 extension.

vulkan

If device is an integer, it selects the device by its index in a system-dependent list of devices. If device is any other string, it selects the first device with a name containing that string as a substring.

The following options are recognized:

debug

If set to 1, enables the validation layer, if installed.

linear_images

If set to 1, images allocated by the hwcontext will be linear and locally mappable.

instance_extensions

A plus separated list of additional instance extensions to enable.

device_extensions

A plus separated list of additional device extensions to enable.

Examples:

-init_hw_device vulkan:1

Choose the second device on the system.

-init_hw_device vulkan:RADV

Choose the first device with a name containing the string RADV.

-init_hw_device vulkan:0,instance_extensions=VK_KHR_wayland_surface+VK_KHR_xcb_surface

Choose the first device and enable the Wayland and XCB instance extensions.

-init_hw_device type[=name]@source

Initialise a new hardware device of type type called name, deriving it from the existing device with the name source.

-init_hw_device list

List all hardware device types supported in this build of ffmpeg.

-filter_hw_device name

Pass the hardware device called name to all filters in any filter graph. This can be used to set the device to upload to with the hwupload filter, or the device to map to with the hwmap filter. Other filters may also make use of this parameter when they require a hardware device. Note that this is typically only required when the input is not already in hardware frames - when it is, filters will derive the device they require from the context of the frames they receive as input.

This is a global setting, so all filters will receive the same device.

-hwaccel[:stream_specifier] hwaccel (input,per-stream)

Use hardware acceleration to decode the matching stream(s). The allowed values of hwaccel are:

none

Do not use any hardware acceleration (the default).

auto

Automatically select the hardware acceleration method.

vdpau

Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.

dxva2

Use DXVA2 (DirectX Video Acceleration) hardware acceleration.

d3d11va

Use D3D11VA (DirectX Video Acceleration) hardware acceleration.

vaapi

Use VAAPI (Video Acceleration API) hardware acceleration.

qsv

Use the Intel QuickSync Video acceleration for video transcoding.

Unlike most other values, this option does not enable accelerated decoding (that is used automatically whenever a qsv decoder is selected), but accelerated transcoding, without copying the frames into the system memory.

For it to work, both the decoder and the encoder must support QSV acceleration and no filters must be used.

This option has no effect if the selected hwaccel is not available or not supported by the chosen decoder.

Note that most acceleration methods are intended for playback and will not be faster than software decoding on modern CPUs. Additionally, ffmpeg will usually need to copy the decoded frames from the GPU memory into the system memory, resulting in further performance loss. This option is thus mainly useful for testing.

-hwaccel_device[:stream_specifier] hwaccel_device (input,per-stream)

Select a device to use for hardware acceleration.

This option only makes sense when the -hwaccel option is also specified. It can either refer to an existing device created with -init_hw_device by name, or it can create a new device as if ‘-init_hw_devicetype:hwaccel_device were called immediately before.

-hwaccels

List all hardware acceleration components enabled in this build of ffmpeg. Actual runtime availability depends on the hardware and its suitable driver being installed.

-fix_sub_duration_heartbeat[:stream_specifier]

Set a specific output video stream as the heartbeat stream according to which to split and push through currently in-progress subtitle upon receipt of a random access packet.

This lowers the latency of subtitles for which the end packet or the following subtitle has not yet been received. As a drawback, this will most likely lead to duplication of subtitle events in order to cover the full duration, so when dealing with use cases where latency of when the subtitle event is passed on to output is not relevant this option should not be utilized.

Requires -fix_sub_duration to be set for the relevant input subtitle stream for this to have any effect, as well as for the input subtitle stream having to be directly mapped to the same output in which the heartbeat stream resides.

5.7 Audio Options

-aframes number (output)

Set the number of audio frames to output. This is an obsolete alias for -frames:a, which you should use instead.

-ar[:stream_specifier] freq (input/output,per-stream)

Set the audio sampling frequency. For output streams it is set by default to the frequency of the corresponding input stream. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.

-aq q (output)

Set the audio quality (codec-specific, VBR). This is an alias for -q:a.

-ac[:stream_specifier] channels (input/output,per-stream)

Set the number of audio channels. For output streams it is set by default to the number of input audio channels. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.

-an (input/output)

As an input option, blocks all audio streams of a file from being filtered or being automatically selected or mapped for any output. See -discard option to disable streams individually.

As an output option, disables audio recording i.e. automatic selection or mapping of any audio stream. For full manual control see the -map option.

-acodec codec (input/output)

Set the audio codec. This is an alias for -codec:a.

-sample_fmt[:stream_specifier] sample_fmt (output,per-stream)

Set the audio sample format. Use -sample_fmts to get a list of supported sample formats.

-af filtergraph (output)

Create the filtergraph specified by filtergraph and use it to filter the stream.

This is an alias for -filter:a, see the -filter option.

5.8 Advanced Audio options

-atag fourcc/tag (output)

Force audio tag/fourcc. This is an alias for -tag:a.

-ch_layout[:stream_specifier] layout (input/output,per-stream)

Alias for -channel_layout.

-channel_layout[:stream_specifier] layout (input/output,per-stream)

Set the audio channel layout. For output streams it is set by default to the input channel layout. For input streams it overrides the channel layout of the input. Not all decoders respect the overridden channel layout. This option also sets the channel layout for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer option.

-guess_layout_max channels (input,per-stream)

If some input channel layout is not known, try to guess only if it corresponds to at most the specified number of channels. For example, 2 tells to ffmpeg to recognize 1 channel as mono and 2 channels as stereo but not 6 channels as 5.1. The default is to always try to guess. Use 0 to disable all guessing. Using the -channel_layout option to explicitly specify an input layout also disables guessing.

5.9 Subtitle options

-scodec codec (input/output)

Set the subtitle codec. This is an alias for -codec:s.

-sn (input/output)

As an input option, blocks all subtitle streams of a file from being filtered or being automatically selected or mapped for any output. See -discard option to disable streams individually.

As an output option, disables subtitle recording i.e. automatic selection or mapping of any subtitle stream. For full manual control see the -map option.

5.10 Advanced Subtitle options

-fix_sub_duration

Fix subtitles durations. For each subtitle, wait for the next packet in the same stream and adjust the duration of the first to avoid overlap. This is necessary with some subtitles codecs, especially DVB subtitles, because the duration in the original packet is only a rough estimate and the end is actually marked by an empty subtitle frame. Failing to use this option when necessary can result in exaggerated durations or muxing failures due to non-monotonic timestamps.

Note that this option will delay the output of all data until the next subtitle packet is decoded: it may increase memory consumption and latency a lot.

-canvas_size size

Set the size of the canvas used to render subtitles.

5.11 Advanced options

-map [-]input_file_id[:stream_specifier][:view_specifier][?] | [linklabel] (output)

Create one or more streams in the output file. This option has two forms for specifying the data source(s): the first selects one or more streams from some input file (specified with -i), the second takes an output from some complex filtergraph (specified with -filter_complex).

In the first form, an output stream is created for every stream from the input file with the index input_file_id. If stream_specifier is given, only those streams that match the specifier are used (see the Stream specifiers section for the stream_specifier syntax).

A - character before the stream identifier creates a "negative" mapping. It disables matching streams from already created mappings.

An optional view_specifier may be given after the stream specifier, which for multiview video specifies the view to be used. The view specifier may have one of the following formats:

view:view_id

select a view by its ID; view_id may be set to ’all’ to use all the views interleaved into one stream;

vidx:view_idx

select a view by its index; i.e. 0 is the base view, 1 is the first non-base view, etc.

vpos:position

select a view by its display position; position may be left or right

The default for transcoding is to only use the base view, i.e. the equivalent of vidx:0. For streamcopy, view specifiers are not supported and all views are always copied.

A trailing ? after the stream index will allow the map to be optional: if the map matches no streams the map will be ignored instead of failing. Note the map will still fail if an invalid input file index is used; such as if the map refers to a non-existent input.

An alternative [linklabel] form will map outputs from complex filter graphs (see the -filter_complex option) to the output file. linklabel must correspond to a defined output link label in the graph.

This option may be specified multiple times, each adding more streams to the output file. Any given input stream may also be mapped any number of times as a source for different output streams, e.g. in order to use different encoding options and/or filters. The streams are created in the output in the same order in which the -map options are given on the commandline.

Using this option disables the default mappings for this output file.

Examples:

map everything

To map ALL streams from the first input file to output

ffmpeg -i INPUT -map 0 output
select specific stream

If you have two audio streams in the first input file, these streams are identified by 0:0 and 0:1. You can use -map to select which streams to place in an output file. For example:

ffmpeg -i INPUT -map 0:1 out.wav

will map the second input stream in INPUT to the (single) output stream in out.wav.

create multiple streams

To select the stream with index 2 from input file a.mov (specified by the identifier 0:2), and stream with index 6 from input b.mov (specified by the identifier 1:6), and copy them to the output file out.mov:

ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov
create multiple streams 2

To select all video and the third audio stream from an input file:

ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT
negative map

To map all the streams except the second audio, use negative mappings

ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
optional map

To map the video and audio streams from the first input, and using the trailing ?, ignore the audio mapping if no audio streams exist in the first input:

ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
map by language

To pick the English audio stream:

ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
-ignore_unknown

Ignore input streams with unknown type instead of failing if copying such streams is attempted.

-copy_unknown

Allow input streams with unknown type to be copied instead of failing if copying such streams is attempted.

-map_metadata[:metadata_spec_out] infile[:metadata_spec_in] (output,per-metadata)

Set metadata information of the next output file from infile. Note that those are file indices (zero-based), not filenames. Optional metadata_spec_in/out parameters specify, which metadata to copy. A metadata specifier can have the following forms:

g

global metadata, i.e. metadata that applies to the whole file

s[:stream_spec]

per-stream metadata. stream_spec is a stream specifier as described in the Stream specifiers chapter. In an input metadata specifier, the first matching stream is copied from. In an output metadata specifier, all matching streams are copied to.

c:chapter_index

per-chapter metadata. chapter_index is the zero-based chapter index.

p:program_index

per-program metadata. program_index is the zero-based program index.

If metadata specifier is omitted, it defaults to global.

By default, global metadata is copied from the first input file, per-stream and per-chapter metadata is copied along with streams/chapters. These default mappings are disabled by creating any mapping of the relevant type. A negative file index can be used to create a dummy mapping that just disables automatic copying.

For example to copy metadata from the first stream of the input file to global metadata of the output file:

ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3

To do the reverse, i.e. copy global metadata to all audio streams:

ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv

Note that simple 0 would work as well in this example, since global metadata is assumed by default.

-map_chapters input_file_index (output)

Copy chapters from input file with index input_file_index to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying.

-benchmark (global)

Show benchmarking information at the end of an encode. Shows real, system and user time used and maximum memory consumption. Maximum memory consumption is not supported on all systems, it will usually display as 0 if not supported.

-benchmark_all (global)

Show benchmarking information during the encode. Shows real, system and user time used in various steps (audio/video encode/decode).

-timelimit duration (global)

Exit after ffmpeg has been running for duration seconds in CPU user time.

-dump (global)

Dump each input packet to stderr.

-hex (global)

When dumping packets, also dump the payload.

-readrate speed (input)

Limit input read speed.

Its value is a floating-point positive number which represents the maximum duration of media, in seconds, that should be ingested in one second of wallclock time. Default value is zero and represents no imposed limitation on speed of ingestion. Value 1 represents real-time speed and is equivalent to -re.

Mainly used to simulate a capture device or live input stream (e.g. when reading from a file). Should not be used with a low value when input is an actual capture device or live stream as it may cause packet loss.

It is useful for when flow speed of output packets is important, such as live streaming.

-re (input)

Read input at native frame rate. This is equivalent to setting -readrate 1.

-readrate_initial_burst seconds

Set an initial read burst time, in seconds, after which -re/-readrate will be enforced.

-vsync parameter (global)
-fps_mode[:stream_specifier] parameter (output,per-stream)

Set video sync method / framerate mode. vsync is applied to all output video streams but can be overridden for a stream by setting fps_mode. vsync is deprecated and will be removed in the future.

For compatibility reasons some of the values for vsync can be specified as numbers (shown in parentheses in the following table).

passthrough (0)

Each frame is passed with its timestamp from the demuxer to the muxer.

cfr (1)

Frames will be duplicated and dropped to achieve exactly the requested constant frame rate.

vfr (2)

Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp.

auto (-1)

Chooses between cfr and vfr depending on muxer capabilities. This is the default method.

Note that the timestamps may be further modified by the muxer, after this. For example, in the case that the format option avoid_negative_ts is enabled.

With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one.

-frame_drop_threshold parameter

Frame drop threshold, which specifies how much behind video frames can be before they are dropped. In frame rate units, so 1.0 is one frame. The default is -1.1. One possible usecase is to avoid framedrops in case of noisy timestamps or to increase frame drop precision in case of exact timestamps.

-apad parameters (output,per-stream)

Pad the output audio stream(s). This is the same as applying -af apad. Argument is a string of filter parameters composed the same as with the apad filter. -shortest must be set for this output for the option to take effect.

-copyts

Do not process input timestamps, but keep their values without trying to sanitize them. In particular, do not remove the initial start time offset value.

Note that, depending on the vsync option or on specific muxer processing (e.g. in case the format option avoid_negative_ts is enabled) the output timestamps may mismatch with the input timestamps even when this option is selected.

-start_at_zero

When used with copyts, shift input timestamps so they start at zero.

This means that using e.g. -ss 50 will make output timestamps start at 50 seconds, regardless of what timestamp the input file started at.

-copytb mode

Specify how to set the encoder timebase when stream copying. mode is an integer numeric value, and can assume one of the following values:

1

Use the demuxer timebase.

The time base is copied to the output encoder from the corresponding input demuxer. This is sometimes required to avoid non monotonically increasing timestamps when copying video streams with variable frame rate.

0

Use the decoder timebase.

The time base is copied to the output encoder from the corresponding input decoder.

-1

Try to make the choice automatically, in order to generate a sane output.

Default value is -1.

-enc_time_base[:stream_specifier] timebase (output,per-stream)

Set the encoder timebase. timebase can assume one of the following values:

0

Assign a default value according to the media type.

For video - use 1/framerate, for audio - use 1/samplerate.

demux

Use the timebase from the demuxer.

filter

Use the timebase from the filtergraph.

a positive number

Use the provided number as the timebase.

This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000) or as a decimal number (e.g. 0.04166, 2.0833e-5)

Default value is 0.

-bitexact (input/output)

Enable bitexact mode for (de)muxer and (de/en)coder

-shortest (output)

Finish encoding when the shortest output stream ends.

Note that this option may require buffering frames, which introduces extra latency. The maximum amount of this latency may be controlled with the -shortest_buf_duration option.

-shortest_buf_duration duration (output)

The -shortest option may require buffering potentially large amounts of data when at least one of the streams is "sparse" (i.e. has large gaps between frames – this is typically the case for subtitles).

This option controls the maximum duration of buffered frames in seconds. Larger values may allow the -shortest option to produce more accurate results, but increase memory use and latency.

The default value is 10 seconds.

-dts_delta_threshold threshold

Timestamp discontinuity delta threshold, expressed as a decimal number of seconds.

The timestamp discontinuity correction enabled by this option is only applied to input formats accepting timestamp discontinuity (for which the AVFMT_TS_DISCONT flag is enabled), e.g. MPEG-TS and HLS, and is automatically disabled when employing the -copyts option (unless wrapping is detected).

If a timestamp discontinuity is detected whose absolute value is greater than threshold, ffmpeg will remove the discontinuity by decreasing/increasing the current DTS and PTS by the corresponding delta value.

The default value is 10.

-dts_error_threshold threshold

Timestamp error delta threshold, expressed as a decimal number of seconds.

The timestamp correction enabled by this option is only applied to input formats not accepting timestamp discontinuity (for which the AVFMT_TS_DISCONT flag is not enabled).

If a timestamp discontinuity is detected whose absolute value is greater than threshold, ffmpeg will drop the PTS/DTS timestamp value.

The default value is 3600*30 (30 hours), which is arbitrarily picked and quite conservative.

-muxdelay seconds (output)

Set the maximum demux-decode delay.

-muxpreload seconds (output)

Set the initial demux-decode delay.

-streamid output-stream-index:new-value (output)

Assign a new stream-id value to an output stream. This option should be specified prior to the output filename to which it applies. For the situation where multiple output files exist, a streamid may be reassigned to a different value.

For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for an output mpegts file:

ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
-bsf[:stream_specifier] bitstream_filters (input/output,per-stream)

Apply bitstream filters to matching streams. The filters are applied to each packet as it is received from the demuxer (when used as an input option) or before it is sent to the muxer (when used as an output option).

bitstream_filters is a comma-separated list of bitstream filter specifications, each of the form

filter[=optname0=optval0:optname1=optval1:...]

Any of the ’,=:’ characters that are to be a part of an option value need to be escaped with a backslash.

Use the -bsfs option to get the list of bitstream filters.

E.g.

ffmpeg -bsf:v h264_mp4toannexb -i h264.mp4 -c:v copy -an out.h264

applies the h264_mp4toannexb bitstream filter (which converts MP4-encapsulated H.264 stream to Annex B) to the input video stream.

On the other hand,

ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt

applies the mov2textsub bitstream filter (which extracts text from MOV subtitles) to the output subtitle stream. Note, however, that since both examples use -c copy, it matters little whether the filters are applied on input or output - that would change if transcoding was happening.

-tag[:stream_specifier] codec_tag (input/output,per-stream)

Force a tag/fourcc for matching streams.

-timecode hh:mm:ssSEPff

Specify Timecode for writing. SEP is ’:’ for non drop timecode and ’;’ (or ’.’) for drop.

ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
-filter_complex filtergraph (global)

Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. For simple graphs – those with one input and one output of the same type – see the -filter options. filtergraph is a description of the filtergraph, as described in the “Filtergraph syntax” section of the ffmpeg-filters manual. This option may be specified multiple times - each use creates a new complex filtergraph.

Inputs to a complex filtergraph may come from different source types, distinguished by the format of the corresponding link label:

  • To connect an input stream, use [file_index:stream_specifier] (i.e. the same syntax as -map). If stream_specifier matches multiple streams, the first one will be used. For multiview video, the stream specifier may be followed by the view specifier, see documentation for the -map option for its syntax.
  • To connect a loopback decoder use [dec:dec_idx], where dec_idx is the index of the loopback decoder to be connected to given input. For multiview video, the decoder index may be followed by the view specifier, see documentation for the -map option for its syntax.
  • To connect an output from another complex filtergraph, use its link label. E.g the following example:
    ffmpeg -i input.mkv \
      -filter_complex '[0:v]scale=size=hd1080,split=outputs=2[for_enc][orig_scaled]' \
      -c:v libx264 -map '[for_enc]' output.mkv \
      -dec 0:0 \
      -filter_complex '[dec:0][orig_scaled]hstack[stacked]' \
      -map '[stacked]' -c:v ffv1 comparison.mkv
    

    reads an input video and

    • (line 2) uses a complex filtergraph with one input and two outputs to scale the video to 1920x1080 and duplicate the result to both outputs;
    • (line 3) encodes one scaled output with libx264 and writes the result to output.mkv;
    • (line 4) decodes this encoded stream with a loopback decoder;
    • (line 5) places the output of the loopback decoder (i.e. the libx264-encoded video) side by side with the scaled original input;
    • (line 6) combined video is then losslessly encoded and written into comparison.mkv.

    Note that the two filtergraphs cannot be combined into one, because then there would be a cycle in the transcoding pipeline (filtergraph output goes to encoding, from there to decoding, then back to the same graph), and such cycles are not allowed.

An unlabeled input will be connected to the first unused input stream of the matching type.

Output link labels are referred to with -map. Unlabeled outputs are added to the first output file.

Note that with this option it is possible to use only lavfi sources without normal input files.

For example, to overlay an image over video

ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map
'[out]' out.mkv

Here [0:v] refers to the first video stream in the first input file, which is linked to the first (main) input of the overlay filter. Similarly the first video stream in the second input is linked to the second (overlay) input of overlay.

Assuming there is only one video stream in each input file, we can omit input labels, so the above is equivalent to

ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map
'[out]' out.mkv

Furthermore we can omit the output label and the single output from the filter graph will be added to the output file automatically, so we can simply write

ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv

As a special exception, you can use a bitmap subtitle stream as input: it will be converted into a video with the same size as the largest video in the file, or 720x576 if no video is present. Note that this is an experimental and temporary solution. It will be removed once libavfilter has proper support for subtitles.

For example, to hardcode subtitles on top of a DVB-T recording stored in MPEG-TS format, delaying the subtitles by 1 second:

ffmpeg -i input.ts -filter_complex \
  '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
  -sn -map '#0x2dc' output.mkv

(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video, audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)

To generate 5 seconds of pure red video using lavfi color source:

ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
-filter_complex_threads nb_threads (global)

Defines how many threads are used to process a filter_complex graph. Similar to filter_threads but used for -filter_complex graphs only. The default is the number of available CPUs.

-lavfi filtergraph (global)

Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. Equivalent to -filter_complex.

-accurate_seek (input)

This option enables or disables accurate seeking in input files with the -ss option. It is enabled by default, so seeking is accurate when transcoding. Use -noaccurate_seek to disable it, which may be useful e.g. when copying some streams and transcoding the others.

-seek_timestamp (input)

This option enables or disables seeking by timestamp in input files with the -ss option. It is disabled by default. If enabled, the argument to the -ss option is considered an actual timestamp, and is not offset by the start time of the file. This matters only for files which do not start from timestamp 0, such as transport streams.

-thread_queue_size size (input/output)

For input, this option sets the maximum number of queued packets when reading from the file or device. With low latency / high rate live streams, packets may be discarded if they are not read in a timely manner; setting this value can force ffmpeg to use a separate input thread and read packets as soon as they arrive. By default ffmpeg only does this if multiple inputs are specified.

For output, this option specified the maximum number of packets that may be queued to each muxing thread.

-sdp_file file (global)

Print sdp information for an output stream to file. This allows dumping sdp information when at least one output isn’t an rtp stream. (Requires at least one of the output formats to be rtp).

-discard (input)

Allows discarding specific streams or frames from streams. Any input stream can be fully discarded, using value all whereas selective discarding of frames from a stream occurs at the demuxer and is not supported by all demuxers.

none

Discard no frame.

default

Default, which discards no frames.

noref

Discard all non-reference frames.

bidir

Discard all bidirectional frames.

nokey

Discard all frames excepts keyframes.

all

Discard all frames.

-abort_on flags (global)

Stop and abort on various conditions. The following flags are available:

empty_output

No packets were passed to the muxer, the output is empty.

empty_output_stream

No packets were passed to the muxer in some of the output streams.

-max_error_rate (global)

Set fraction of decoding frame failures across all inputs which when crossed ffmpeg will return exit code 69. Crossing this threshold does not terminate processing. Range is a floating-point number between 0 to 1. Default is 2/3.

-xerror (global)

Stop and exit on error

-max_muxing_queue_size packets (output,per-stream)

When transcoding audio and/or video streams, ffmpeg will not begin writing into the output until it has one packet for each such stream. While waiting for that to happen, packets for other streams are buffered. This option sets the size of this buffer, in packets, for the matching output stream.

The default value of this option should be high enough for most uses, so only touch this option if you are sure that you need it.

-muxing_queue_data_threshold bytes (output,per-stream)

This is a minimum threshold until which the muxing queue size is not taken into account. Defaults to 50 megabytes per stream, and is based on the overall size of packets passed to the muxer.

-auto_conversion_filters (global)

Enable automatically inserting format conversion filters in all filter graphs, including those defined by -vf, -af, -filter_complex and -lavfi. If filter format negotiation requires a conversion, the initialization of the filters will fail. Conversions can still be performed by inserting the relevant conversion filter (scale, aresample) in the graph. On by default, to explicitly disable it you need to specify -noauto_conversion_filters.

-bits_per_raw_sample[:stream_specifier] value (output,per-stream)

Declare the number of bits per raw sample in the given output stream to be value. Note that this option sets the information provided to the encoder/muxer, it does not change the stream to conform to this value. Setting values that do not match the stream properties may result in encoding failures or invalid output files.

-stats_enc_pre[:stream_specifier] path (output,per-stream)
-stats_enc_post[:stream_specifier] path (output,per-stream)
-stats_mux_pre[:stream_specifier] path (output,per-stream)

Write per-frame encoding information about the matching streams into the file given by path.

-stats_enc_pre writes information about raw video or audio frames right before they are sent for encoding, while -stats_enc_post writes information about encoded packets as they are received from the encoder. -stats_mux_pre writes information about packets just as they are about to be sent to the muxer. Every frame or packet produces one line in the specified file. The format of this line is controlled by -stats_enc_pre_fmt / -stats_enc_post_fmt / -stats_mux_pre_fmt.

When stats for multiple streams are written into a single file, the lines corresponding to different streams will be interleaved. The precise order of this interleaving is not specified and not guaranteed to remain stable between different invocations of the program, even with the same options.

-stats_enc_pre_fmt[:stream_specifier] format_spec (output,per-stream)
-stats_enc_post_fmt[:stream_specifier] format_spec (output,per-stream)
-stats_mux_pre_fmt[:stream_specifier] format_spec (output,per-stream)

Specify the format for the lines written with -stats_enc_pre / -stats_enc_post / -stats_mux_pre.

format_spec is a string that may contain directives of the form {fmt}. format_spec is backslash-escaped — use \{, \}, and \\ to write a literal {, }, or \, respectively, into the output.

The directives given with fmt may be one of the following:

fidx

Index of the output file.

sidx

Index of the output stream in the file.

n

Frame number. Pre-encoding: number of frames sent to the encoder so far. Post-encoding: number of packets received from the encoder so far. Muxing: number of packets submitted to the muxer for this stream so far.

ni

Input frame number. Index of the input frame (i.e. output by a decoder) that corresponds to this output frame or packet. -1 if unavailable.

tb

Timebase in which this frame/packet’s timestamps are expressed, as a rational number num/den. Note that encoder and muxer may use different timebases.

tbi

Timebase for ptsi, as a rational number num/den. Available when ptsi is available, 0/1 otherwise.

pts

Presentation timestamp of the frame or packet, as an integer. Should be multiplied by the timebase to compute presentation time.

ptsi

Presentation timestamp of the input frame (see ni), as an integer. Should be multiplied by tbi to compute presentation time. Printed as (2^63 - 1 = 9223372036854775807) when not available.

t

Presentation time of the frame or packet, as a decimal number. Equal to pts multiplied by tb.

ti

Presentation time of the input frame (see ni), as a decimal number. Equal to ptsi multiplied by tbi. Printed as inf when not available.

dts (packet)

Decoding timestamp of the packet, as an integer. Should be multiplied by the timebase to compute presentation time.

dt (packet)

Decoding time of the frame or packet, as a decimal number. Equal to dts multiplied by tb.

sn (frame,audio)

Number of audio samples sent to the encoder so far.

samp (frame,audio)

Number of audio samples in the frame.

size (packet)

Size of the encoded packet in bytes.

br (packet)

Current bitrate in bits per second.

abr (packet)

Average bitrate for the whole stream so far, in bits per second, -1 if it cannot be determined at this point.

key (packet)

Character ’K’ if the packet contains a keyframe, character ’N’ otherwise.

Directives tagged with packet may only be used with -stats_enc_post_fmt and -stats_mux_pre_fmt.

Directives tagged with frame may only be used with -stats_enc_pre_fmt.

Directives tagged with audio may only be used with audio streams.

The default format strings are:

pre-encoding

{fidx} {sidx} {n} {t}

post-encoding

{fidx} {sidx} {n} {t}

In the future, new items may be added to the end of the default formatting strings. Users who depend on the format staying exactly the same, should prescribe it manually.

Note that stats for different streams written into the same file may have different formats.

5.12 Preset files

A preset file contains a sequence of option=value pairs, one for each line, specifying a sequence of options which would be awkward to specify on the command line. Lines starting with the hash (’#’) character are ignored and are used to provide comments. Check the presets directory in the FFmpeg source tree for examples.

There are two types of preset files: ffpreset and avpreset files.

5.12.1 ffpreset files

ffpreset files are specified with the vpre, apre, spre, and fpre options. The fpre option takes the filename of the preset instead of a preset name as input and can be used for any kind of codec. For the vpre, apre, and spre options, the options specified in a preset file are applied to the currently selected codec of the same type as the preset option.

The argument passed to the vpre, apre, and spre preset options identifies the preset file to use according to the following rules:

First ffmpeg searches for a file named arg.ffpreset in the directories $FFMPEG_DATADIR (if set), and $HOME/.ffmpeg, and in the datadir defined at configuration time (usually PREFIX/share/ffmpeg) or in a ffpresets folder along the executable on win32, in that order. For example, if the argument is libvpx-1080p, it will search for the file libvpx-1080p.ffpreset.

If no such file is found, then ffmpeg will search for a file named codec_name-arg.ffpreset in the above-mentioned directories, where codec_name is the name of the codec to which the preset file options will be applied. For example, if you select the video codec with -vcodec libvpx and use -vpre 1080p, then it will search for the file libvpx-1080p.ffpreset.

5.12.2 avpreset files

avpreset files are specified with the pre option. They work similar to ffpreset files, but they only allow encoder- specific options. Therefore, an option=value pair specifying an encoder cannot be used.

When the pre option is specified, ffmpeg will look for files with the suffix .avpreset in the directories $AVCONV_DATADIR (if set), and $HOME/.avconv, and in the datadir defined at configuration time (usually PREFIX/share/ffmpeg), in that order.

First ffmpeg searches for a file named codec_name-arg.avpreset in the above-mentioned directories, where codec_name is the name of the codec to which the preset file options will be applied. For example, if you select the video codec with -vcodec libvpx and use -pre 1080p, then it will search for the file libvpx-1080p.avpreset.

If no such file is found, then ffmpeg will search for a file named arg.avpreset in the same directories.

5.13 vstats file format

The -vstats and -vstats_file options enable generation of a file containing statistics about the generated video outputs.

The -vstats_version option controls the format version of the generated file.

With version 1 the format is:

frame= FRAME q= FRAME_QUALITY PSNR= PSNR f_size= FRAME_SIZE s_size= STREAM_SIZEkB time= TIMESTAMP br= BITRATEkbits/s avg_br= AVERAGE_BITRATEkbits/s

With version 2 the format is:

out= OUT_FILE_INDEX st= OUT_FILE_STREAM_INDEX frame= FRAME_NUMBER q= FRAME_QUALITYf PSNR= PSNR f_size= FRAME_SIZE s_size= STREAM_SIZEkB time= TIMESTAMP br= BITRATEkbits/s avg_br= AVERAGE_BITRATEkbits/s

The value corresponding to each key is described below:

avg_br

average bitrate expressed in Kbits/s

br

bitrate expressed in Kbits/s

frame

number of encoded frame

out

out file index

PSNR

Peak Signal to Noise Ratio

q

quality of the frame

f_size

encoded packet size expressed as number of bytes

s_size

stream size expressed in KiB

st

out file stream index

time

time of the packet

type

picture type

See also the -stats_enc options for an alternative way to show encoding statistics.

6 Examples

6.1 Video and Audio grabbing

If you specify the input format and device then ffmpeg can grab video and audio directly.

ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg

Or with an ALSA audio source (mono input, card id 1) instead of OSS:

ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg

Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer such as xawtv by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.

6.2 X11 grabbing

Grab the X11 display with ffmpeg via

ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg

0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable.

ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg

0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. 10 is the x-offset and 20 the y-offset for the grabbing.

6.3 Video and Audio file format conversion

Any supported file format and protocol can serve as input to ffmpeg:

Examples:

  • You can use YUV files as input:
    ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
    

    It will use the files:

    /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
    /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
    

    The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the -s option if ffmpeg cannot guess it.

  • You can input from a raw YUV420P file:
    ffmpeg -i /tmp/test.yuv /tmp/out.avi
    

    test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.

  • You can output to a raw YUV420P file:
    ffmpeg -i mydivx.avi hugefile.yuv
    
  • You can set several input files and output files:
    ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
    

    Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.

  • You can also do audio and video conversions at the same time:
    ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
    

    Converts a.wav to MPEG audio at 22050 Hz sample rate.

  • You can encode to several formats at the same time and define a mapping from input stream to output streams:
    ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2
    

    Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. ’-map file:index’ specifies which input stream is used for each output stream, in the order of the definition of output streams.

  • You can transcode decrypted VOBs:
    ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi
    

    This is a typical DVD ripping example; the input is a VOB file, the output an AVI file with MPEG-4 video and MP3 audio. Note that in this command we use B-frames so the MPEG-4 stream is DivX5 compatible, and GOP size is 300 which means one intra frame every 10 seconds for 29.97fps input video. Furthermore, the audio stream is MP3-encoded so you need to enable LAME support by passing --enable-libmp3lame to configure. The mapping is particularly useful for DVD transcoding to get the desired audio language.

    NOTE: To see the supported input formats, use ffmpeg -demuxers.

  • You can extract images from a video, or create a video from many images:

    For extracting images from a video:

    ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
    

    This will extract one video frame per second from the video and will output them in files named foo-001.jpeg, foo-002.jpeg, etc. Images will be rescaled to fit the new WxH values.

    If you want to extract just a limited number of frames, you can use the above command in combination with the -frames:v or -t option, or in combination with -ss to start extracting from a certain point in time.

    For creating a video from many images:

    ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
    

    The syntax foo-%03d.jpeg specifies to use a decimal number composed of three digits padded with zeroes to express the sequence number. It is the same syntax supported by the C printf function, but only formats accepting a normal integer are suitable.

    When importing an image sequence, -i also supports expanding shell-like wildcard patterns (globbing) internally, by selecting the image2-specific -pattern_type glob option.

    For example, for creating a video from filenames matching the glob pattern foo-*.jpeg:

    ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
    
  • You can put many streams of the same type in the output:
    ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
    

    The resulting output file test12.nut will contain the first four streams from the input files in reverse order.

  • To force CBR video output:
    ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
    
  • The four options lmin, lmax, mblmin and mblmax use ’lambda’ units, but you may use the QP2LAMBDA constant to easily convert from ’q’ units:
    ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
    

7 Syntax

This section documents the syntax and formats employed by the FFmpeg libraries and tools.

7.1 Quoting and escaping

FFmpeg adopts the following quoting and escaping mechanism, unless explicitly specified. The following rules are applied:

  • '’ and ‘\’ are special characters (respectively used for quoting and escaping). In addition to them, there might be other special characters depending on the specific syntax where the escaping and quoting are employed.
  • A special character is escaped by prefixing it with a ‘\’.
  • All characters enclosed between ‘''’ are included literally in the parsed string. The quote character ‘'’ itself cannot be quoted, so you may need to close the quote and escape it.
  • Leading and trailing whitespaces, unless escaped or quoted, are removed from the parsed string.

Note that you may need to add a second level of escaping when using the command line or a script, which depends on the syntax of the adopted shell language.

The function av_get_token defined in libavutil/avstring.h can be used to parse a token quoted or escaped according to the rules defined above.

The tool tools/ffescape in the FFmpeg source tree can be used to automatically quote or escape a string in a script.

7.1.1 Examples

  • Escape the string Crime d'Amour containing the ' special character:
    Crime d\'Amour
    
  • The string above contains a quote, so the ' needs to be escaped when quoting it:
    'Crime d'\''Amour'
    
  • Include leading or trailing whitespaces using quoting:
    '  this string starts and ends with whitespaces  '
    
  • Escaping and quoting can be mixed together:
    ' The string '\'string\'' is a string '
    
  • To include a literal ‘\’ you can use either escaping or quoting:
    'c:\foo' can be written as c:\\foo
    

7.2 Date

The accepted syntax is:

[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
now

If the value is "now" it takes the current time.

Time is local time unless Z is appended, in which case it is interpreted as UTC. If the year-month-day part is not specified it takes the current year-month-day.

7.3 Time duration

There are two accepted syntaxes for expressing time duration.

[-][HH:]MM:SS[.m...]

HH expresses the number of hours, MM the number of minutes for a maximum of 2 digits, and SS the number of seconds for a maximum of 2 digits. The m at the end expresses decimal value for SS.

or

[-]S+[.m...][s|ms|us]

S expresses the number of seconds, with the optional decimal part m. The optional literal suffixes ‘s’, ‘ms’ or ‘us’ indicate to interpret the value as seconds, milliseconds or microseconds, respectively.

In both expressions, the optional ‘-’ indicates negative duration.

7.3.1 Examples

The following examples are all valid time duration:

55

55 seconds

0.2

0.2 seconds

200ms

200 milliseconds, that’s 0.2s

200000us

200000 microseconds, that’s 0.2s

12:03:45

12 hours, 03 minutes and 45 seconds

23.189

23.189 seconds

7.4 Video size

Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size abbreviation.

The following abbreviations are recognized:

ntsc

720x480

pal

720x576

qntsc

352x240

qpal

352x288

sntsc

640x480

spal

768x576

film

352x240

ntsc-film

352x240

sqcif

128x96

qcif

176x144

cif

352x288

4cif

704x576

16cif

1408x1152

qqvga

160x120

qvga

320x240

vga

640x480

svga

800x600

xga

1024x768

uxga

1600x1200

qxga

2048x1536

sxga

1280x1024

qsxga

2560x2048

hsxga

5120x4096

wvga

852x480

wxga

1366x768

wsxga

1600x1024

wuxga

1920x1200

woxga

2560x1600

wqsxga

3200x2048

wquxga

3840x2400

whsxga

6400x4096

whuxga

7680x4800

cga

320x200

ega

640x350

hd480

852x480

hd720

1280x720

hd1080

1920x1080

2k

2048x1080

2kflat

1998x1080

2kscope

2048x858

4k

4096x2160

4kflat

3996x2160

4kscope

4096x1716

nhd

640x360

hqvga

240x160

wqvga

400x240

fwqvga

432x240

hvga

480x320

qhd

960x540

2kdci

2048x1080

4kdci

4096x2160

uhd2160

3840x2160

uhd4320

7680x4320

7.5 Video rate

Specify the frame rate of a video, expressed as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation.

The following abbreviations are recognized:

ntsc

30000/1001

pal

25/1

qntsc

30000/1001

qpal

25/1

sntsc

30000/1001

spal

25/1

film

24/1

ntsc-film

24000/1001

7.6 Ratio

A ratio can be expressed as an expression, or in the form numerator:denominator.

Note that a ratio with infinite (1/0) or negative value is considered valid, so you should check on the returned value if you want to exclude those values.

The undefined value can be expressed using the "0:0" string.

7.7 Color

It can be the name of a color as defined below (case insensitive match) or a [0x|#]RRGGBB[AA] sequence, possibly followed by @ and a string representing the alpha component.

The alpha component may be a string composed by "0x" followed by an hexadecimal number or a decimal number between 0.0 and 1.0, which represents the opacity value (‘0x00’ or ‘0.0’ means completely transparent, ‘0xff’ or ‘1.0’ completely opaque). If the alpha component is not specified then ‘0xff’ is assumed.

The string ‘random’ will result in a random color.

The following names of colors are recognized:

AliceBlue

0xF0F8FF

AntiqueWhite

0xFAEBD7

Aqua

0x00FFFF

Aquamarine

0x7FFFD4

Azure

0xF0FFFF

Beige

0xF5F5DC

Bisque

0xFFE4C4

Black

0x000000

BlanchedAlmond

0xFFEBCD

Blue

0x0000FF

BlueViolet

0x8A2BE2

Brown

0xA52A2A

BurlyWood

0xDEB887

CadetBlue

0x5F9EA0

Chartreuse

0x7FFF00

Chocolate

0xD2691E

Coral

0xFF7F50

CornflowerBlue

0x6495ED

Cornsilk

0xFFF8DC

Crimson

0xDC143C

Cyan

0x00FFFF

DarkBlue

0x00008B

DarkCyan

0x008B8B

DarkGoldenRod

0xB8860B

DarkGray

0xA9A9A9

DarkGreen

0x006400

DarkKhaki

0xBDB76B

DarkMagenta

0x8B008B

DarkOliveGreen

0x556B2F

Darkorange

0xFF8C00

DarkOrchid

0x9932CC

DarkRed

0x8B0000

DarkSalmon

0xE9967A

DarkSeaGreen

0x8FBC8F

DarkSlateBlue

0x483D8B

DarkSlateGray

0x2F4F4F

DarkTurquoise

0x00CED1

DarkViolet

0x9400D3

DeepPink

0xFF1493

DeepSkyBlue

0x00BFFF

DimGray

0x696969

DodgerBlue

0x1E90FF

FireBrick

0xB22222

FloralWhite

0xFFFAF0

ForestGreen

0x228B22

Fuchsia

0xFF00FF

Gainsboro

0xDCDCDC

GhostWhite

0xF8F8FF

Gold

0xFFD700

GoldenRod

0xDAA520

Gray

0x808080

Green

0x008000

GreenYellow

0xADFF2F

HoneyDew

0xF0FFF0

HotPink

0xFF69B4

IndianRed

0xCD5C5C

Indigo

0x4B0082

Ivory

0xFFFFF0

Khaki

0xF0E68C

Lavender

0xE6E6FA

LavenderBlush

0xFFF0F5

LawnGreen

0x7CFC00

LemonChiffon

0xFFFACD

LightBlue

0xADD8E6

LightCoral

0xF08080

LightCyan

0xE0FFFF

LightGoldenRodYellow

0xFAFAD2

LightGreen

0x90EE90

LightGrey

0xD3D3D3

LightPink

0xFFB6C1

LightSalmon

0xFFA07A

LightSeaGreen

0x20B2AA

LightSkyBlue

0x87CEFA

LightSlateGray

0x778899

LightSteelBlue

0xB0C4DE

LightYellow

0xFFFFE0

Lime

0x00FF00

LimeGreen

0x32CD32

Linen

0xFAF0E6

Magenta

0xFF00FF

Maroon

0x800000

MediumAquaMarine

0x66CDAA

MediumBlue

0x0000CD

MediumOrchid

0xBA55D3

MediumPurple

0x9370D8

MediumSeaGreen

0x3CB371

MediumSlateBlue

0x7B68EE

MediumSpringGreen

0x00FA9A

MediumTurquoise

0x48D1CC

MediumVioletRed

0xC71585

MidnightBlue

0x191970

MintCream

0xF5FFFA

MistyRose

0xFFE4E1

Moccasin

0xFFE4B5

NavajoWhite

0xFFDEAD

Navy

0x000080

OldLace

0xFDF5E6

Olive

0x808000

OliveDrab

0x6B8E23

Orange

0xFFA500

OrangeRed

0xFF4500

Orchid

0xDA70D6

PaleGoldenRod

0xEEE8AA

PaleGreen

0x98FB98

PaleTurquoise

0xAFEEEE

PaleVioletRed

0xD87093

PapayaWhip

0xFFEFD5

PeachPuff

0xFFDAB9

Peru

0xCD853F

Pink

0xFFC0CB

Plum

0xDDA0DD

PowderBlue

0xB0E0E6

Purple

0x800080

Red

0xFF0000

RosyBrown

0xBC8F8F

RoyalBlue

0x4169E1

SaddleBrown

0x8B4513

Salmon

0xFA8072

SandyBrown

0xF4A460

SeaGreen

0x2E8B57

SeaShell

0xFFF5EE

Sienna

0xA0522D

Silver

0xC0C0C0

SkyBlue

0x87CEEB

SlateBlue

0x6A5ACD

SlateGray

0x708090

Snow

0xFFFAFA

SpringGreen

0x00FF7F

SteelBlue

0x4682B4

Tan

0xD2B48C

Teal

0x008080

Thistle

0xD8BFD8

Tomato

0xFF6347

Turquoise

0x40E0D0

Violet

0xEE82EE

Wheat

0xF5DEB3

White

0xFFFFFF

WhiteSmoke

0xF5F5F5

Yellow

0xFFFF00

YellowGreen

0x9ACD32

7.8 Channel Layout

A channel layout specifies the spatial disposition of the channels in a multi-channel audio stream. To specify a channel layout, FFmpeg makes use of a special syntax.

Individual channels are identified by an id, as given by the table below:

FL

front left

FR

front right

FC

front center

LFE

low frequency

BL

back left

BR

back right

FLC

front left-of-center

FRC

front right-of-center

BC

back center

SL

side left

SR

side right

TC

top center

TFL

top front left

TFC

top front center

TFR

top front right

TBL

top back left

TBC

top back center

TBR

top back right

DL

downmix left

DR

downmix right

WL

wide left

WR

wide right

SDL

surround direct left

SDR

surround direct right

LFE2

low frequency 2

Standard channel layout compositions can be specified by using the following identifiers:

mono

FC

stereo

FL+FR

2.1

FL+FR+LFE

3.0

FL+FR+FC

3.0(back)

FL+FR+BC

4.0

FL+FR+FC+BC

quad

FL+FR+BL+BR

quad(side)

FL+FR+SL+SR

3.1

FL+FR+FC+LFE

5.0

FL+FR+FC+BL+BR

5.0(side)

FL+FR+FC+SL+SR

4.1

FL+FR+FC+LFE+BC

5.1

FL+FR+FC+LFE+BL+BR

5.1(side)

FL+FR+FC+LFE+SL+SR

6.0

FL+FR+FC+BC+SL+SR

6.0(front)

FL+FR+FLC+FRC+SL+SR

3.1.2

FL+FR+FC+LFE+TFL+TFR

hexagonal

FL+FR+FC+BL+BR+BC

6.1

FL+FR+FC+LFE+BC+SL+SR

6.1

FL+FR+FC+LFE+BL+BR+BC

6.1(front)

FL+FR+LFE+FLC+FRC+SL+SR

7.0

FL+FR+FC+BL+BR+SL+SR

7.0(front)

FL+FR+FC+FLC+FRC+SL+SR

7.1

FL+FR+FC+LFE+BL+BR+SL+SR

7.1(wide)

FL+FR+FC+LFE+BL+BR+FLC+FRC

7.1(wide-side)

FL+FR+FC+LFE+FLC+FRC+SL+SR

5.1.2

FL+FR+FC+LFE+BL+BR+TFL+TFR

octagonal

FL+FR+FC+BL+BR+BC+SL+SR

cube

FL+FR+BL+BR+TFL+TFR+TBL+TBR

5.1.4

FL+FR+FC+LFE+BL+BR+TFL+TFR+TBL+TBR

7.1.2

FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR

7.1.4

FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR

7.2.3

FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBC+LFE2

9.1.4

FL+FR+FC+LFE+BL+BR+FLC+FRC+SL+SR+TFL+TFR+TBL+TBR

hexadecagonal

FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR

downmix

DL+DR

22.2

FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR

A custom channel layout can be specified as a sequence of terms, separated by ’+’. Each term can be:

  • the name of a single channel (e.g. ‘FL’, ‘FR’, ‘FC’, ‘LFE’, etc.), each optionally containing a custom name after a ’@’, (e.g. ‘FL@Left’, ‘FR@Right’, ‘FC@Center’, ‘LFE@Low_Frequency’, etc.)

A standard channel layout can be specified by the following:

  • the name of a single channel (e.g. ‘FL’, ‘FR’, ‘FC’, ‘LFE’, etc.)
  • the name of a standard channel layout (e.g. ‘mono’, ‘stereo’, ‘4.0’, ‘quad’, ‘5.0’, etc.)
  • a number of channels, in decimal, followed by ’c’, yielding the default channel layout for that number of channels (see the function av_channel_layout_default). Note that not all channel counts have a default layout.
  • a number of channels, in decimal, followed by ’C’, yielding an unknown channel layout with the specified number of channels. Note that not all channel layout specification strings support unknown channel layouts.
  • a channel layout mask, in hexadecimal starting with "0x" (see the AV_CH_* macros in libavutil/channel_layout.h.

Before libavutil version 53 the trailing character "c" to specify a number of channels was optional, but now it is required, while a channel layout mask can also be specified as a decimal number (if and only if not followed by "c" or "C").

See also the function av_channel_layout_from_string defined in libavutil/channel_layout.h.

8 Expression Evaluation

When evaluating an arithmetic expression, FFmpeg uses an internal formula evaluator, implemented through the libavutil/eval.h interface.

An expression may contain unary, binary operators, constants, and functions.

Two expressions expr1 and expr2 can be combined to form another expression "expr1;expr2". expr1 and expr2 are evaluated in turn, and the new expression evaluates to the value of expr2.

The following binary operators are available: +, -, *, /, ^.

The following unary operators are available: +, -.

Some internal variables can be used to store and load intermediary results. They can be accessed using the ld and st functions with an index argument varying from 0 to 9 to specify which internal variable to access.

The following functions are available:

abs(x)

Compute absolute value of x.

acos(x)

Compute arccosine of x.

asin(x)

Compute arcsine of x.

atan(x)

Compute arctangent of x.

atan2(y, x)

Compute principal value of the arc tangent of y/x.

between(x, min, max)

Return 1 if x is greater than or equal to min and lesser than or equal to max, 0 otherwise.

bitand(x, y)
bitor(x, y)

Compute bitwise and/or operation on x and y.

The results of the evaluation of x and y are converted to integers before executing the bitwise operation.

Note that both the conversion to integer and the conversion back to floating point can lose precision. Beware of unexpected results for large numbers (usually 2^53 and larger).

ceil(expr)

Round the value of expression expr upwards to the nearest integer. For example, "ceil(1.5)" is "2.0".

clip(x, min, max)

Return the value of x clipped between min and max.

cos(x)

Compute cosine of x.

cosh(x)

Compute hyperbolic cosine of x.

eq(x, y)

Return 1 if x and y are equivalent, 0 otherwise.

exp(x)

Compute exponential of x (with base e, the Euler’s number).

floor(expr)

Round the value of expression expr downwards to the nearest integer. For example, "floor(-1.5)" is "-2.0".

gauss(x)

Compute Gauss function of x, corresponding to exp(-x*x/2) / sqrt(2*PI).

gcd(x, y)

Return the greatest common divisor of x and y. If both x and y are 0 or either or both are less than zero then behavior is undefined.

gt(x, y)

Return 1 if x is greater than y, 0 otherwise.

gte(x, y)

Return 1 if x is greater than or equal to y, 0 otherwise.

hypot(x, y)

This function is similar to the C function with the same name; it returns "sqrt(x*x + y*y)", the length of the hypotenuse of a right triangle with sides of length x and y, or the distance of the point (x, y) from the origin.

if(x, y)

Evaluate x, and if the result is non-zero return the result of the evaluation of y, return 0 otherwise.

if(x, y, z)

Evaluate x, and if the result is non-zero return the evaluation result of y, otherwise the evaluation result of z.

ifnot(x, y)

Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.

ifnot(x, y, z)

Evaluate x, and if the result is zero return the evaluation result of y, otherwise the evaluation result of z.

isinf(x)

Return 1.0 if x is +/-INFINITY, 0.0 otherwise.

isnan(x)

Return 1.0 if x is NAN, 0.0 otherwise.

ld(idx)

Load the value of the internal variable with index idx, which was previously stored with st(idx, expr). The function returns the loaded value.

lerp(x, y, z)

Return linear interpolation between x and y by amount of z.

log(x)

Compute natural logarithm of x.

lt(x, y)

Return 1 if x is lesser than y, 0 otherwise.

lte(x, y)

Return 1 if x is lesser than or equal to y, 0 otherwise.

max(x, y)

Return the maximum between x and y.

min(x, y)

Return the minimum between x and y.

mod(x, y)

Compute the remainder of division of x by y.

not(expr)

Return 1.0 if expr is zero, 0.0 otherwise.

pow(x, y)

Compute the power of x elevated y, it is equivalent to "(x)^(y)".

print(t)
print(t, l)

Print the value of expression t with loglevel l. If l is not specified then a default log level is used. Return the value of the expression printed.

random(idx)

Return a pseudo random value between 0.0 and 1.0. idx is the index of the internal variable used to save the seed/state, which can be previously stored with st(idx).

To initialize the seed, you need to store the seed value as a 64-bit unsigned integer in the internal variable with index idx.

For example, to store the seed with value 42 in the internal variable with index 0 and print a few random values:

st(0,42); print(random(0)); print(random(0)); print(random(0))
randomi(idx, min, max)

Return a pseudo random value in the interval between min and max. idx is the index of the internal variable which will be used to save the seed/state, which can be previously stored with st(idx).

To initialize the seed, you need to store the seed value as a 64-bit unsigned integer in the internal variable with index idx.

root(expr, max)

Find an input value for which the function represented by expr with argument ld(0) is 0 in the interval 0..max.

The expression in expr must denote a continuous function or the result is undefined.

ld(0) is used to represent the function input value, which means that the given expression will be evaluated multiple times with various input values that the expression can access through ld(0). When the expression evaluates to 0 then the corresponding input value will be returned.

round(expr)

Round the value of expression expr to the nearest integer. For example, "round(1.5)" is "2.0".

sgn(x)

Compute sign of x.

sin(x)

Compute sine of x.

sinh(x)

Compute hyperbolic sine of x.

sqrt(expr)

Compute the square root of expr. This is equivalent to "(expr)^.5".

squish(x)

Compute expression 1/(1 + exp(4*x)).

st(idx, expr)

Store the value of the expression expr in an internal variable. idx specifies the index of the variable where to store the value, and it is a value ranging from 0 to 9. The function returns the value stored in the internal variable.

The stored value can be retrieved with ld(var).

Note: variables are currently not shared between expressions.

tan(x)

Compute tangent of x.

tanh(x)

Compute hyperbolic tangent of x.

taylor(expr, x)
taylor(expr, x, idx)

Evaluate a Taylor series at x, given an expression representing the ld(idx)-th derivative of a function at 0.

When the series does not converge the result is undefined.

ld(idx) is used to represent the derivative order in expr, which means that the given expression will be evaluated multiple times with various input values that the expression can access through ld(idx). If idx is not specified then 0 is assumed.

Note, when you have the derivatives at y instead of 0, taylor(expr, x-y) can be used.

time(0)

Return the current (wallclock) time in seconds.

trunc(expr)

Round the value of expression expr towards zero to the nearest integer. For example, "trunc(-1.5)" is "-1.0".

while(cond, expr)

Evaluate expression expr while the expression cond is non-zero, and returns the value of the last expr evaluation, or NAN if cond was always false.

The following constants are available:

PI

area of the unit disc, approximately 3.14

E

exp(1) (Euler’s number), approximately 2.718

PHI

golden ratio (1+sqrt(5))/2, approximately 1.618

Assuming that an expression is considered "true" if it has a non-zero value, note that:

* works like AND

+ works like OR

For example the construct:

if (A AND B) then C

is equivalent to:

if(A*B, C)

In your C code, you can extend the list of unary and binary functions, and define recognized constants, so that they are available for your expressions.

The evaluator also recognizes the International System unit prefixes. If ’i’ is appended after the prefix, binary prefixes are used, which are based on powers of 1024 instead of powers of 1000. The ’B’ postfix multiplies the value by 8, and can be appended after a unit prefix or used alone. This allows using for example ’KB’, ’MiB’, ’G’ and ’B’ as number postfix.

The list of available International System prefixes follows, with indication of the corresponding powers of 10 and of 2.

y

10^-24 / 2^-80

z

10^-21 / 2^-70

a

10^-18 / 2^-60

f

10^-15 / 2^-50

p

10^-12 / 2^-40

n

10^-9 / 2^-30

u

10^-6 / 2^-20

m

10^-3 / 2^-10

c

10^-2

d

10^-1

h

10^2

k

10^3 / 2^10

K

10^3 / 2^10

M

10^6 / 2^20

G

10^9 / 2^30

T

10^12 / 2^40

P

10^15 / 2^50

E

10^18 / 2^60

Z

10^21 / 2^70

Y

10^24 / 2^80

9 Codec Options

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 nonsensical 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:

b integer (encoding,audio,video)

Set bitrate in bits/s. Default value is 200K.

ab integer (encoding,audio)

Set audio bitrate (in bits/s). Default value is 128K.

bt integer (encoding,video)

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.

flags flags (decoding/encoding,audio,video,subtitles)

Set generic flags.

Possible values:

mv4

Use four motion vector by macroblock (mpeg4).

qpel

Use 1/4 pel motion compensation.

loop

Use loop filter.

qscale

Use fixed qscale.

pass1

Use internal 2pass ratecontrol in first pass mode.

pass2

Use internal 2pass ratecontrol in second pass mode.

gray

Only decode/encode grayscale.

psnr

Set error[?] variables during encoding.

truncated

Input bitstream might be randomly truncated.

drop_changed

Don’t output frames whose parameters differ from first decoded frame in stream. Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.

ildct

Use interlaced DCT.

low_delay

Force low delay.

global_header

Place global headers in extradata instead of every keyframe.

bitexact

Only write platform-, build- and time-independent data. (except (I)DCT). This ensures that file and data checksums are reproducible and match between platforms. Its primary use is for regression testing.

aic

Apply H263 advanced intra coding / mpeg4 ac prediction.

ilme

Apply interlaced motion estimation.

cgop

Use closed gop.

output_corrupt

Output even potentially corrupted frames.

time_base rational number

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.

g integer (encoding,video)

Set the group of picture (GOP) size. Default value is 12.

ar integer (decoding/encoding,audio)

Set audio sampling rate (in Hz).

ac integer (decoding/encoding,audio)

Set number of audio channels.

cutoff integer (encoding,audio)

Set cutoff bandwidth. (Supported only by selected encoders, see their respective documentation sections.)

frame_size integer (encoding,audio)

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.

frame_number integer

Set the frame number.

delay integer
qcomp float (encoding,video)

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.

qblur float (encoding,video)

Set video quantizer scale blur (VBR).

qmin integer (encoding,video)

Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2.

qmax integer (encoding,video)

Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31.

qdiff integer (encoding,video)

Set max difference between the quantizer scale (VBR).

bf integer (encoding,video)

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.

b_qfactor float (encoding,video)

Set qp factor between P and B frames.

codec_tag integer
bug flags (decoding,video)

Workaround not auto detected encoder bugs.

Possible values:

autodetect
xvid_ilace

Xvid interlacing bug (autodetected if fourcc==XVIX)

ump4

(autodetected if fourcc==UMP4)

no_padding

padding bug (autodetected)

amv
qpel_chroma
std_qpel

old standard qpel (autodetected per fourcc/version)

qpel_chroma2
direct_blocksize

direct-qpel-blocksize bug (autodetected per fourcc/version)

edge

edge padding bug (autodetected per fourcc/version)

hpel_chroma
dc_clip
ms

Workaround various bugs in microsoft broken decoders.

trunc

trancated frames

strict integer (decoding/encoding,audio,video)

Specify how strictly to follow the standards.

Possible values:

very

strictly conform to an older more strict version of the spec or reference software

strict

strictly conform to all the things in the spec no matter what consequences

normal
unofficial

allow unofficial extensions

experimental

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.

b_qoffset float (encoding,video)

Set QP offset between P and B frames.

err_detect flags (decoding,audio,video)

Set error detection flags.

Possible values:

crccheck

verify embedded CRCs

bitstream

detect bitstream specification deviations

buffer

detect improper bitstream length

explode

abort decoding on minor error detection

ignore_err

ignore decoding errors, and continue decoding. This is useful if you want to analyze the content of a video and thus want everything to be decoded no matter what. This option will not result in a video that is pleasing to watch in case of errors.

careful

consider things that violate the spec and have not been seen in the wild as errors

compliant

consider all spec non compliancies as errors

aggressive

consider things that a sane encoder should not do as an error

has_b_frames integer
block_align integer
rc_override_count integer
maxrate integer (encoding,audio,video)

Set max bitrate tolerance (in bits/s). Requires bufsize to be set.

minrate integer (encoding,audio,video)

Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise.

bufsize integer (encoding,audio,video)

Set ratecontrol buffer size (in bits).

i_qfactor float (encoding,video)

Set QP factor between P and I frames.

i_qoffset float (encoding,video)

Set QP offset between P and I frames.

dct integer (encoding,video)

Set DCT algorithm.

Possible values:

auto

autoselect a good one (default)

fastint

fast integer

int

accurate integer

mmx
altivec
faan

floating point AAN DCT

lumi_mask float (encoding,video)

Compress bright areas stronger than medium ones.

tcplx_mask float (encoding,video)

Set temporal complexity masking.

scplx_mask float (encoding,video)

Set spatial complexity masking.

p_mask float (encoding,video)

Set inter masking.

dark_mask float (encoding,video)

Compress dark areas stronger than medium ones.

idct integer (decoding/encoding,video)

Select IDCT implementation.

Possible values:

auto
int
simple
simplemmx
simpleauto

Automatically pick a IDCT compatible with the simple one

arm
altivec
sh4
simplearm
simplearmv5te
simplearmv6
simpleneon
xvid
faani

floating point AAN IDCT

slice_count integer
ec flags (decoding,video)

Set error concealment strategy.

Possible values:

guess_mvs

iterative motion vector (MV) search (slow)

deblock

use strong deblock filter for damaged MBs

favor_inter

favor predicting from the previous frame instead of the current

bits_per_coded_sample integer
aspect rational number (encoding,video)

Set sample aspect ratio.

sar rational number (encoding,video)

Set sample aspect ratio. Alias to aspect.

debug flags (decoding/encoding,audio,video,subtitles)

Print specific debug info.

Possible values:

pict

picture info

rc

rate control

bitstream
mb_type

macroblock (MB) type

qp

per-block quantization parameter (QP)

dct_coeff
green_metadata

display complexity metadata for the upcoming frame, GoP or for a given duration.

skip
startcode
er

error recognition

mmco

memory management control operations (H.264)

bugs
buffers

picture buffer allocations

thread_ops

threading operations

nomc

skip motion compensation

cmp integer (encoding,video)

Set full pel me compare function.

Possible values:

sad

sum of absolute differences, fast (default)

sse

sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

bit

number of bits needed for the block

rd

rate distortion optimal, slow

zero

0

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

w53

5/3 wavelet, only used in snow

w97

9/7 wavelet, only used in snow

dctmax
chroma
subcmp integer (encoding,video)

Set sub pel me compare function.

Possible values:

sad

sum of absolute differences, fast (default)

sse

sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

bit

number of bits needed for the block

rd

rate distortion optimal, slow

zero

0

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

w53

5/3 wavelet, only used in snow

w97

9/7 wavelet, only used in snow

dctmax
chroma
mbcmp integer (encoding,video)

Set macroblock compare function.

Possible values:

sad

sum of absolute differences, fast (default)

sse

sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

bit

number of bits needed for the block

rd

rate distortion optimal, slow

zero

0

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

w53

5/3 wavelet, only used in snow

w97

9/7 wavelet, only used in snow

dctmax
chroma
ildctcmp integer (encoding,video)

Set interlaced dct compare function.

Possible values:

sad

sum of absolute differences, fast (default)

sse

sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

bit

number of bits needed for the block

rd

rate distortion optimal, slow

zero

0

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

w53

5/3 wavelet, only used in snow

w97

9/7 wavelet, only used in snow

dctmax
chroma
dia_size integer (encoding,video)

Set diamond type & size for motion estimation.

(1024, INT_MAX)

full motion estimation(slowest)

(768, 1024]

umh motion estimation

(512, 768]

hex motion estimation

(256, 512]

l2s diamond motion estimation

[2,256]

var diamond motion estimation

(-1, 2)

small diamond motion estimation

-1

funny diamond motion estimation

(INT_MIN, -1)

sab diamond motion estimation

last_pred integer (encoding,video)

Set amount of motion predictors from the previous frame.

precmp integer (encoding,video)

Set pre motion estimation compare function.

Possible values:

sad

sum of absolute differences, fast (default)

sse

sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

bit

number of bits needed for the block

rd

rate distortion optimal, slow

zero

0

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

w53

5/3 wavelet, only used in snow

w97

9/7 wavelet, only used in snow

dctmax
chroma
pre_dia_size integer (encoding,video)

Set diamond type & size for motion estimation pre-pass.

subq integer (encoding,video)

Set sub pel motion estimation quality.

me_range integer (encoding,video)

Set limit motion vectors range (1023 for DivX player).

global_quality integer (encoding,audio,video)
slice_flags integer
mbd integer (encoding,video)

Set macroblock decision algorithm (high quality mode).

Possible values:

simple

use mbcmp (default)

bits

use fewest bits

rd

use best rate distortion

rc_init_occupancy integer (encoding,video)

Set number of bits which should be loaded into the rc buffer before decoding starts.

flags2 flags (decoding/encoding,audio,video,subtitles)

Possible values:

fast

Allow non spec compliant speedup tricks.

noout

Skip bitstream encoding.

ignorecrop

Ignore cropping information from sps.

local_header

Place global headers at every keyframe instead of in extradata.

chunks

Frame data might be split into multiple chunks.

showall

Show all frames before the first keyframe.

export_mvs

Export motion vectors into frame side-data (see AV_FRAME_DATA_MOTION_VECTORS) for codecs that support it. See also doc/examples/export_mvs.c.

skip_manual

Do not skip samples and export skip information as frame side data.

ass_ro_flush_noop

Do not reset ASS ReadOrder field on flush.

icc_profiles

Generate/parse embedded ICC profiles from/to colorimetry tags.

export_side_data flags (decoding/encoding,audio,video,subtitles)

Possible values:

mvs

Export motion vectors into frame side-data (see AV_FRAME_DATA_MOTION_VECTORS) for codecs that support it. See also doc/examples/export_mvs.c.

prft

Export encoder Producer Reference Time into packet side-data (see AV_PKT_DATA_PRFT) for codecs that support it.

venc_params

Export video encoding parameters through frame side data (see AV_FRAME_DATA_VIDEO_ENC_PARAMS) for codecs that support it. At present, those are H.264 and VP9.

film_grain

Export film grain parameters through frame side data (see AV_FRAME_DATA_FILM_GRAIN_PARAMS). Supported at present by AV1 decoders.

threads integer (decoding/encoding,video)

Set the number of threads to be used, in case the selected codec implementation supports multi-threading.

Possible values:

auto, 0

automatically select the number of threads to set

Default value is ‘auto’.

dc integer (encoding,video)

Set intra_dc_precision.

nssew integer (encoding,video)

Set nsse weight.

skip_top integer (decoding,video)

Set number of macroblock rows at the top which are skipped.

skip_bottom integer (decoding,video)

Set number of macroblock rows at the bottom which are skipped.

profile integer (encoding,audio,video)

Set encoder codec profile. Default value is ‘unknown’. Encoder specific profiles are documented in the relevant encoder documentation.

level integer (encoding,audio,video)

Set the encoder level. This level depends on the specific codec, and might correspond to the profile level. It is set by default to ‘unknown’.

Possible values:

unknown
lowres integer (decoding,audio,video)

Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.

mblmin integer (encoding,video)

Set min macroblock lagrange factor (VBR).

mblmax integer (encoding,video)

Set max macroblock lagrange factor (VBR).

skip_loop_filter integer (decoding,video)
skip_idct integer (decoding,video)
skip_frame integer (decoding,video)

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:

none

Discard no frame.

default

Discard useless frames like 0-sized frames.

noref

Discard all non-reference frames.

bidir

Discard all bidirectional frames.

nokey

Discard all frames excepts keyframes.

nointra

Discard all frames except I frames.

all

Discard all frames.

Default value is ‘default’.

bidir_refine integer (encoding,video)

Refine the two motion vectors used in bidirectional macroblocks.

keyint_min integer (encoding,video)

Set minimum interval between IDR-frames.

refs integer (encoding,video)

Set reference frames to consider for motion compensation.

trellis integer (encoding,audio,video)

Set rate-distortion optimal quantization.

mv0_threshold integer (encoding,video)
compression_level integer (encoding,audio,video)
bits_per_raw_sample integer
channel_layout integer (decoding/encoding,audio)

See (ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

rc_max_vbv_use float (encoding,video)
rc_min_vbv_use float (encoding,video)
color_primaries integer (decoding/encoding,video)

Possible values:

bt709

BT.709

bt470m

BT.470 M

bt470bg

BT.470 BG

smpte170m

SMPTE 170 M

smpte240m

SMPTE 240 M

film

Film

bt2020

BT.2020

smpte428
smpte428_1

SMPTE ST 428-1

smpte431

SMPTE 431-2

smpte432

SMPTE 432-1

jedec-p22

JEDEC P22

color_trc integer (decoding/encoding,video)

Possible values:

bt709

BT.709

gamma22

BT.470 M

gamma28

BT.470 BG

smpte170m

SMPTE 170 M

smpte240m

SMPTE 240 M

linear

Linear

log
log100

Log

log_sqrt
log316

Log square root

iec61966_2_4
iec61966-2-4

IEC 61966-2-4

bt1361
bt1361e

BT.1361

iec61966_2_1
iec61966-2-1

IEC 61966-2-1

bt2020_10
bt2020_10bit

BT.2020 - 10 bit

bt2020_12
bt2020_12bit

BT.2020 - 12 bit

smpte2084

SMPTE ST 2084

smpte428
smpte428_1

SMPTE ST 428-1

arib-std-b67

ARIB STD-B67

colorspace integer (decoding/encoding,video)

Possible values:

rgb

RGB

bt709

BT.709

fcc

FCC

bt470bg

BT.470 BG

smpte170m

SMPTE 170 M

smpte240m

SMPTE 240 M

ycocg

YCOCG

bt2020nc
bt2020_ncl

BT.2020 NCL

bt2020c
bt2020_cl

BT.2020 CL

smpte2085

SMPTE 2085

chroma-derived-nc

Chroma-derived NCL

chroma-derived-c

Chroma-derived CL

ictcp

ICtCp

color_range integer (decoding/encoding,video)

If used as input parameter, it serves as a hint to the decoder, which color_range the input has. Possible values:

tv
mpeg
limited

MPEG (219*2^(n-8))

pc
jpeg
full

JPEG (2^n-1)

chroma_sample_location integer (decoding/encoding,video)

Possible values:

left
center
topleft
top
bottomleft
bottom
log_level_offset integer

Set the log level offset.

slices integer (encoding,video)

Number of slices, used in parallelized encoding.

thread_type flags (decoding/encoding,video)

Select which multithreading methods to use.

Use of ‘frame’ will increase decoding delay by one frame per thread, so clients which cannot provide future frames should not use it.

Possible values:

slice

Decode more than one part of a single frame at once.

Multithreading using slices works only when the video was encoded with slices.

frame

Decode more than one frame at once.

Default value is ‘slice+frame’.

audio_service_type integer (encoding,audio)

Set audio service type.

Possible values:

ma

Main Audio Service

ef

Effects

vi

Visually Impaired

hi

Hearing Impaired

di

Dialogue

co

Commentary

em

Emergency

vo

Voice Over

ka

Karaoke

request_sample_fmt sample_fmt (decoding,audio)

Set sample format audio decoders should prefer. Default value is none.

pkt_timebase rational number
sub_charenc encoding (decoding,subtitles)

Set the input subtitles character encoding.

field_order field_order (video)

Set/override the field order of the video. Possible values:

progressive

Progressive video

tt

Interlaced video, top field coded and displayed first

bb

Interlaced video, bottom field coded and displayed first

tb

Interlaced video, top coded first, bottom displayed first

bt

Interlaced video, bottom coded first, top displayed first

skip_alpha bool (decoding,video)

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.

codec_whitelist list (input)

"," separated list of allowed decoders. By default all are allowed.

dump_separator string (input)

Separator used to separate the fields printed on the command line about the Stream parameters. For example, to separate the fields with newlines and indentation:

ffprobe -dump_separator "
                          "  -i ~/videos/matrixbench_mpeg2.mpg
max_pixels integer (decoding/encoding,video)

Maximum number of pixels per image. This value can be used to avoid out of memory failures due to large images.

apply_cropping bool (decoding,video)

Enable cropping if cropping parameters are multiples of the required alignment for the left and top parameters. If the alignment is not met the cropping will be partially applied to maintain alignment. Default is 1 (enabled). Note: The required alignment depends on if AV_CODEC_FLAG_UNALIGNED is set and the CPU. AV_CODEC_FLAG_UNALIGNED cannot be changed from the command line. Also hardware decoders will not apply left/top Cropping.

10 Decoders

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.

11 Video Decoders

A description of some of the currently available video decoders follows.

11.1 av1

AOMedia Video 1 (AV1) decoder.

11.1.1 Options

operating_point

Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.

11.2 hevc

HEVC (AKA ITU-T H.265 or ISO/IEC 23008-2) decoder.

The decoder supports MV-HEVC multiview streams with at most two views. Views to be output are selected by supplying a list of view IDs to the decoder (the view_ids option). This option may be set either statically before decoder init, or from the get_format() callback - useful for the case when the view count or IDs change dynamically during decoding.

Only the base layer is decoded by default.

Note that if you are using the ffmpeg CLI tool, you should be using view specifiers as documented in its manual, rather than the options documented here.

11.2.1 Options

view_ids (MV-HEVC)

Specify a list of view IDs that should be output. This option can also be set to a single ’-1’, which will cause all views defined in the VPS to be decoded and output.

view_ids_available (MV-HEVC)

This option may be read by the caller to retrieve an array of view IDs available in the active VPS. The array is empty for single-layer video.

The value of this option is guaranteed to be accurate when read from the get_format() callback. It may also be set at other times (e.g. after opening the decoder), but the value is informational only and may be incorrect (e.g. when the stream contains multiple distinct VPS NALUs).

view_pos_available (MV-HEVC)

This option may be read by the caller to retrieve an array of view positions (left, right, or unspecified) available in the active VPS, as AVStereo3DView values. When the array is available, its elements apply to the corresponding elements of view_ids_available, i.e. view_pos_available[i] contains the position of view with ID view_ids_available[i].

Same validity restrictions as for view_ids_available apply to this option.

11.3 rawvideo

Raw video decoder.

This decoder decodes rawvideo streams.

11.3.1 Options

top top_field_first

Specify the assumed field type of the input video.

-1

the video is assumed to be progressive (default)

0

bottom-field-first is assumed

1

top-field-first is assumed

11.4 libdav1d

dav1d AV1 decoder.

libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec. Requires the presence of the libdav1d headers and library during configuration. You need to explicitly configure the build with --enable-libdav1d.

11.4.1 Options

The following options are supported by the libdav1d wrapper.

framethreads

Set amount of frame threads to use during decoding. The default value is 0 (autodetect). This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the option max_frame_delay and the global option threads instead.

tilethreads

Set amount of tile threads to use during decoding. The default value is 0 (autodetect). This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the global option threads instead.

max_frame_delay

Set max amount of frames the decoder may buffer internally. The default value is 0 (autodetect).

filmgrain

Apply film grain to the decoded video if present in the bitstream. Defaults to the internal default of the library. This option is deprecated and will be removed in the future. See the global option export_side_data to export Film Grain parameters instead of applying it.

oppoint

Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the internal default of the library.

alllayers

Output all spatial layers of a scalable AV1 bitstream. The default value is false.

11.5 libdavs2

AVS2-P2/IEEE1857.4 video decoder wrapper.

This decoder allows libavcodec to decode AVS2 streams with davs2 library.

11.6 libuavs3d

AVS3-P2/IEEE1857.10 video decoder.

libuavs3d allows libavcodec to decode AVS3 streams. Requires the presence of the libuavs3d headers and library during configuration. You need to explicitly configure the build with --enable-libuavs3d.

11.6.1 Options

The following option is supported by the libuavs3d wrapper.

frame_threads

Set amount of frame threads to use during decoding. The default value is 0 (autodetect).

11.7 libxevd

eXtra-fast Essential Video Decoder (XEVD) MPEG-5 EVC decoder wrapper.

This decoder requires the presence of the libxevd headers and library during configuration. You need to explicitly configure the build with --enable-libxevd.

The xevd project website is at https://github.com/mpeg5/xevd.

11.7.1 Options

The following options are supported by the libxevd wrapper. The xevd-equivalent options or values are listed in parentheses for easy migration.

To get a more accurate and extensive documentation of the libxevd options, invoke the command xevd_app --help or consult the libxevd documentation.

threads (threads)

Force to use a specific number of threads

11.8 QSV Decoders

The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC, JPEG/MJPEG, VP8, VP9, AV1, VVC).

11.8.1 Common Options

The following options are supported by all qsv decoders.

async_depth

Internal parallelization depth, the higher the value the higher the latency.

gpu_copy

A GPU-accelerated copy between video and system memory

default
on
off

11.8.2 HEVC Options

Extra options for hevc_qsv.

load_plugin

A user plugin to load in an internal session

none
hevc_sw
hevc_hw
load_plugins

A :-separate list of hexadecimal plugin UIDs to load in an internal session

11.9 v210

Uncompressed 4:2:2 10-bit decoder.

11.9.1 Options

custom_stride

Set the line size of the v210 data in bytes. The default value is 0 (autodetect). You can use the special -1 value for a strideless v210 as seen in BOXX files.

12 Audio Decoders

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

12.1 ac3

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).

12.1.1 AC-3 Decoder Options

-drc_scale value

Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This factor is applied exponentially. The default value is 1. There are 3 notable scale factor ranges:

drc_scale == 0

DRC disabled. Produces full range audio.

0 < drc_scale <= 1

DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression.

drc_scale > 1

DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds are enhanced.

12.2 flac

FLAC audio decoder.

This decoder aims to implement the complete FLAC specification from Xiph.

12.2.1 FLAC Decoder options

-use_buggy_lpc

The lavc FLAC encoder used to produce buggy streams with high lpc values (like the default value). This option makes it possible to decode such streams correctly by using lavc’s old buggy lpc logic for decoding.

12.3 ffwavesynth

Internal wave synthesizer.

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.

12.4 libcelt

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.

12.5 libgsm

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.

12.6 libilbc

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.

12.6.1 Options

The following option is supported by the libilbc wrapper.

enhance

Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).

12.7 libopencore-amrnb

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.

12.8 libopencore-amrwb

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.

12.9 libopus

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.

An FFmpeg native decoder for Opus exists, so users can decode Opus without this library.

13 Subtitles Decoders

13.1 libaribb24

ARIB STD-B24 caption decoder.

Implements profiles A and C of the ARIB STD-B24 standard.

13.1.1 libaribb24 Decoder Options

-aribb24-base-path path

Sets the base path for the libaribb24 library. This is utilized for reading of configuration files (for custom unicode conversions), and for dumping of non-text symbols as images under that location.

Unset by default.

-aribb24-skip-ruby-text boolean

Tells the decoder wrapper to skip text blocks that contain half-height ruby text.

Enabled by default.

13.2 libaribcaption

Yet another ARIB STD-B24 caption decoder using external libaribcaption library.

Implements profiles A and C of the Japanse ARIB STD-B24 standard, Brazilian ABNT NBR 15606-1, and Philippines version of ISDB-T.

Requires the presence of the libaribcaption headers and library (https://github.com/xqq/libaribcaption) during configuration. You need to explicitly configure the build with --enable-libaribcaption. If both libaribb24 and libaribcaption are enabled, libaribcaption decoder precedes.

13.2.1 libaribcaption Decoder Options

-sub_type subtitle_type

Specifies the format of the decoded subtitles.

bitmap

Graphical image.

ass

ASS formatted text.

text

Simple text based output without formatting.

The default is ass as same as libaribb24 decoder. Some present players (e.g., mpv) expect ASS format for ARIB caption.

-caption_encoding encoding_scheme

Specifies the encoding scheme of input subtitle text.

auto

Automatically detect text encoding (default).

jis

8bit-char JIS encoding defined in ARIB STD B24. This encoding used in Japan for ISDB captions.

utf8

UTF-8 encoding defined in ARIB STD B24. This encoding is used in Philippines for ISDB-T captions.

latin

Latin character encoding defined in ABNT NBR 15606-1. This encoding is used in South America for SBTVD / ISDB-Tb captions.

-font font_name[,font_name2,...]

Specify comma-separated list of font family names to be used for bitmap or ass type subtitle rendering. Only first font name is used for ass type subtitle.

If not specified, use internaly defined default font family.

-ass_single_rect boolean

ARIB STD-B24 specifies that some captions may be displayed at different positions at a time (multi-rectangle subtitle). Since some players (e.g., old mpv) can’t handle multiple ASS rectangles in a single AVSubtitle, or multiple ASS rectangles of indeterminate duration with the same start timestamp, this option can change the behavior so that all the texts are displayed in a single ASS rectangle.

The default is false.

If your player cannot handle AVSubtitles with multiple ASS rectangles properly, set this option to true or define ASS_SINGLE_RECT=1 to change default behavior at compilation.

-force_outline_text boolean

Specify whether always render outline text for all characters regardless of the indication by charactor style.

The default is false.

-outline_width number (0.0 - 3.0)

Specify width for outline text, in dots (relative).

The default is 1.5.

-ignore_background boolean

Specify whether to ignore background color rendering.

The default is false.

-ignore_ruby boolean

Specify whether to ignore rendering for ruby-like (furigana) characters.

The default is false.

-replace_drcs boolean

Specify whether to render replaced DRCS characters as Unicode characters.

The default is true.

-replace_msz_ascii boolean

Specify whether to replace MSZ (Middle Size; half width) fullwidth alphanumerics with halfwidth alphanumerics.

The default is true.

-replace_msz_japanese boolean

Specify whether to replace some MSZ (Middle Size; half width) fullwidth japanese special characters with halfwidth ones.

The default is true.

-replace_msz_glyph boolean

Specify whether to replace MSZ (Middle Size; half width) characters with halfwidth glyphs if the fonts supports it. This option works under FreeType or DirectWrite renderer with Adobe-Japan1 compliant fonts. e.g., IBM Plex Sans JP, Morisawa BIZ UDGothic, Morisawa BIZ UDMincho, Yu Gothic, Yu Mincho, and Meiryo.

The default is true.

-canvas_size image_size

Specify the resolution of the canvas to render subtitles to; usually, this should be frame size of input video. This only applies when -subtitle_type is set to bitmap.

The libaribcaption decoder assumes input frame size for bitmap rendering as below:

  1. PROFILE_A : 1440 x 1080 with SAR (PAR) 4:3
  2. PROFILE_C : 320 x 180 with SAR (PAR) 1:1

If actual frame size of input video does not match above assumption, the rendered captions may be distorted. To make the captions undistorted, add -canvas_size option to specify actual input video size.

Note that the -canvas_size option is not required for video with different size but same aspect ratio. In such cases, the caption will be stretched or shrunk to actual video size if -canvas_size option is not specified. If -canvas_size option is specified with different size, the caption will be stretched or shrunk as specified size with calculated SAR.

13.2.2 libaribcaption decoder usage examples

Display MPEG-TS file with ARIB subtitle by ffplay tool:

ffplay -sub_type bitmap MPEG.TS

Display MPEG-TS file with input frame size 1920x1080 by ffplay tool:

ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS

Embed ARIB subtitle in transcoded video:

ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h264 dest.mp4

13.3 dvbsub

13.3.1 Options

compute_clut
-2

Compute clut once if no matching CLUT is in the stream.

-1

Compute clut if no matching CLUT is in the stream.

0

Never compute CLUT

1

Always compute CLUT and override the one provided in the stream.

dvb_substream

Selects the dvb substream, or all substreams if -1 which is default.

13.4 dvdsub

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.

13.4.1 Options

palette

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 commas, for example 0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b.

ifo_palette

Specify the IFO file from which the global palette is obtained. (experimental)

forced_subs_only

Only decode subtitle entries marked as forced. Some titles have forced and non-forced subtitles in the same track. Setting this flag to 1 will only keep the forced subtitles. Default value is 0.

13.5 libzvbi-teletext

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.

13.5.1 Options

txt_page

List of teletext page numbers to decode. Pages that do not match the specified list are dropped. You may use the special * string to match all pages, or subtitle to match all subtitle pages. Default value is *.

txt_default_region

Set default character set used for decoding, a value between 0 and 87 (see ETS 300 706, Section 15, Table 32). Default value is -1, which does not override the libzvbi default. This option is needed for some legacy level 1.0 transmissions which cannot signal the proper charset.

txt_chop_top

Discards the top teletext line. Default value is 1.

txt_format

Specifies the format of the decoded subtitles.

bitmap

The default format, you should use this for teletext pages, because certain graphics and colors cannot be expressed in simple text or even ASS.

text

Simple text based output without formatting.

ass

Formatted ASS output, subtitle pages and teletext pages are returned in different styles, subtitle pages are stripped down to text, but an effort is made to keep the text alignment and the formatting.

txt_left

X offset of generated bitmaps, default is 0.

txt_top

Y offset of generated bitmaps, default is 0.

txt_chop_spaces

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 characters. Default value is 1.

txt_duration

Sets the display duration of the decoded teletext pages or subtitles in milliseconds. Default value is -1 which means infinity or until the next subtitle event comes.

txt_transparent

Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque background.

txt_opacity

Sets the opacity (0-255) of the teletext background. If txt_transparent is not set, it only affects characters between a start box and an end box, typically subtitles. Default value is 0 if txt_transparent is set, 255 otherwise.

14 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 --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.

15 Audio Encoders

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

15.1 aac

Advanced Audio Coding (AAC) encoder.

This encoder is the default AAC encoder, natively implemented into FFmpeg.

15.1.1 Options

b

Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode. If this option is unspecified it is set to 128kbps.

q

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.

cutoff

Set cutoff frequency. If unspecified will allow the encoder to dynamically adjust the cutoff to improve clarity on low bitrates.

aac_coder

Set AAC encoder coding method. Possible values:

twoloop

Two loop searching (TLS) method. This is the default 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. Will tune itself based on whether aac_is, aac_ms and aac_pns are enabled.

anmr

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

This is an experimental coder which currently produces a lower quality, is more unstable and is slower than the default twoloop coder but has potential. Currently has no support for the aac_is or aac_pns options. Not currently recommended.

fast

Constant quantizer method.

Uses a cheaper version of twoloop algorithm that doesn’t try to do as many clever adjustments. Worse with low bitrates (less than 64kbps), but is better and much faster at higher bitrates.

aac_ms

Sets mid/side coding mode. The default value of "auto" will automatically use M/S with bands which will benefit from such coding. Can be forced for all bands using the value "enable", which is mainly useful for debugging or disabled using "disable".

aac_is

Sets intensity stereo coding tool usage. By default, it’s enabled and will automatically toggle IS for similar pairs of stereo bands if it’s beneficial. Can be disabled for debugging by setting the value to "disable".

aac_pns

Uses perceptual noise substitution to replace low entropy high frequency bands with imperceptible white noise during the decoding process. By default, it’s enabled, but can be disabled for debugging purposes by using "disable".

aac_tns

Enables the use of a multitap FIR filter which spans through the high frequency bands to hide quantization noise during the encoding process and is reverted by the decoder. As well as decreasing unpleasant artifacts in the high range this also reduces the entropy in the high bands and allows for more bits to be used by the mid-low bands. By default it’s enabled but can be disabled for debugging by setting the option to "disable".

aac_ltp

Enables the use of the long term prediction extension which increases coding efficiency in very low bandwidth situations such as encoding of voice or solo piano music by extending constant harmonic peaks in bands throughout frames. This option is implied by profile:a aac_low and is incompatible with aac_pred. Use in conjunction with -ar to decrease the samplerate.

aac_pred

Enables the use of a more traditional style of prediction where the spectral coefficients transmitted are replaced by the difference of the current coefficients minus the previous "predicted" coefficients. In theory and sometimes in practice this can improve quality for low to mid bitrate audio. This option implies the aac_main profile and is incompatible with aac_ltp.

profile

Sets the encoding profile, possible values:

aac_low

The default, AAC "Low-complexity" profile. Is the most compatible and produces decent quality.

mpeg2_aac_low

Equivalent to -profile:a aac_low -aac_pns 0. PNS was introduced with the MPEG4 specifications.

aac_ltp

Long term prediction profile, is enabled by and will enable the aac_ltp option. Introduced in MPEG4.

aac_main

Main-type prediction profile, is enabled by and will enable the aac_pred option. Introduced in MPEG2.

If this option is unspecified it is set to ‘aac_low’.

15.2 ac3 and ac3_fixed

AC-3 audio encoders.

These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366.

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 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.

15.2.1 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.

15.2.1.1 Metadata Control Options

-per_frame_metadata boolean

Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.

0

The metadata values set at initialization will be used for every frame in the stream. (default)

1

Metadata values can be changed before encoding each frame.

15.2.1.2 Downmix Levels

-center_mixlev 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:

0.707

Apply -3dB gain

0.595

Apply -4.5dB gain (default)

0.500

Apply -6dB gain

-surround_mixlev 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:

0.707

Apply -3dB gain

0.500

Apply -6dB gain (default)

0.000

Silence Surround Channel(s)

15.2.1.3 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.

-mixing_level 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 room_type option is not the default value, the mixing_level option must not be -1.

-room_type 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 mixing_level option and the room_type option have the default values.

0
notindicated

Not Indicated (default)

1
large

Large Room

2
small

Small Room

15.2.1.4 Other Metadata Options

-copyright boolean

Copyright Indicator. Specifies whether a copyright exists for this audio.

0
off

No Copyright Exists (default)

1
on

Copyright Exists

-dialnorm 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.

-dsur_mode 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 NOT mean the encoder will actually apply Dolby Surround processing.

0
notindicated

Not Indicated (default)

1
off

Not Dolby Surround Encoded

2
on

Dolby Surround Encoded

-original boolean

Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.

0
off

Not Original Source

1
on

Original Source (default)

15.2.2 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 center_mixlev and surround_mixlev options if it supports the Alternate Bit Stream Syntax.

15.2.2.1 Extended Bitstream Information - Part 1

-dmix_mode 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.

0
notindicated

Not Indicated (default)

1
ltrt

Lt/Rt Downmix Preferred

2
loro

Lo/Ro Downmix Preferred

-ltrt_cmixlev 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.

1.414

Apply +3dB gain

1.189

Apply +1.5dB gain

1.000

Apply 0dB gain

0.841

Apply -1.5dB gain

0.707

Apply -3.0dB gain

0.595

Apply -4.5dB gain (default)

0.500

Apply -6.0dB gain

0.000

Silence Center Channel

-ltrt_surmixlev 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.

0.841

Apply -1.5dB gain

0.707

Apply -3.0dB gain

0.595

Apply -4.5dB gain

0.500

Apply -6.0dB gain (default)

0.000

Silence Surround Channel(s)

-loro_cmixlev 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.

1.414

Apply +3dB gain

1.189

Apply +1.5dB gain

1.000

Apply 0dB gain

0.841

Apply -1.5dB gain

0.707

Apply -3.0dB gain

0.595

Apply -4.5dB gain (default)

0.500

Apply -6.0dB gain

0.000

Silence Center Channel

-loro_surmixlev 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.

0.841

Apply -1.5dB gain

0.707

Apply -3.0dB gain

0.595

Apply -4.5dB gain

0.500

Apply -6.0dB gain (default)

0.000

Silence Surround Channel(s)

15.2.2.2 Extended Bitstream Information - Part 2

-dsurex_mode mode

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.

0
notindicated

Not Indicated (default)

1
on

Dolby Surround EX Off

2
off

Dolby Surround EX On

-dheadphone_mode 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 NOT mean the encoder will actually apply Dolby Headphone processing.

0
notindicated

Not Indicated (default)

1
on

Dolby Headphone Off

2
off

Dolby Headphone On

-ad_conv_type type

A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.

0
standard

Standard A/D Converter (default)

1
hdcd

HDCD A/D Converter

15.2.3 Other AC-3 Encoding Options

-stereo_rematrixing 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.

cutoff frequency

Set lowpass cutoff frequency. If unspecified, the encoder selects a default determined by various other encoding parameters.

15.2.4 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.

-channel_coupling 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.

-1
auto

Selected by Encoder (default)

0
off

Disable Channel Coupling

1
on

Enable Channel Coupling

-cpl_start_band 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 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.

-1
auto

Selected by Encoder (default)

15.3 flac

FLAC (Free Lossless Audio Codec) Encoder

15.3.1 Options

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

compression_level

Sets the compression level, which chooses defaults for many other options if they are not set explicitly. Valid values are from 0 to 12, 5 is the default.

frame_size

Sets the size of the frames in samples per channel.

lpc_coeff_precision

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

lpc_type

Sets the first stage LPC algorithm

none

LPC is not used

fixed

fixed LPC coefficients

levinson
cholesky
lpc_passes

Number of passes to use for Cholesky factorization during LPC analysis

min_partition_order

The minimum partition order

max_partition_order

The maximum partition order

prediction_order_method
estimation
2level
4level
8level
search

Bruteforce search

log
ch_mode

Channel mode

auto

The mode is chosen automatically for each frame

indep

Channels are independently coded

left_side
right_side
mid_side
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.

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.

15.4 opus

Opus encoder.

This is a native FFmpeg encoder for the Opus format. Currently, it’s in development and only implements the CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder.

15.4.1 Options

b

Set bit rate in bits/s. If unspecified it uses the number of channels and the layout to make a good guess.

opus_delay

Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality.

15.5 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 --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 has support for the AAC-HE profiles.

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/.

15.5.1 Options

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

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.

ar

Set audio sampling rate (in Hz).

channels

Set the number of audio channels.

flags +qscale

Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the vbr value is positive.

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.

profile

Set audio profile.

The following profiles are recognized:

aac_low

Low Complexity AAC (LC)

aac_he

High Efficiency AAC (HE-AAC)

aac_he_v2

High Efficiency AAC version 2 (HE-AACv2)

aac_ld

Low Delay AAC (LD)

aac_eld

Enhanced Low Delay AAC (ELD)

If not specified it is set to ‘aac_low’.

The following are private options of the libfdk_aac encoder.

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.

eld_sbr

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

Default value is 0.

eld_v2

Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if set to 1, disabled if set to 0.

Note that option is available when fdk-aac version (AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) > (4.0.0).

Default value is 0.

signaling

Set SBR/PS signaling style.

It can assume one of the following values:

default

choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)

implicit

implicit backwards compatible signaling

explicit_sbr

explicit SBR, implicit PS signaling

explicit_hierarchical

explicit hierarchical signaling

Default value is ‘default’.

latm

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

Default value is 0.

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.

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 ‘aac_low’ profile supports VBR encoding.

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

1

32 kbps/channel

2

40 kbps/channel

3

48-56 kbps/channel

4

64 kbps/channel

5

about 80-96 kbps/channel

Default value is 0.

frame_length

Set the audio frame length in samples. Default value is the internal default of the library. Refer to the library’s documentation for information about supported values.

15.5.2 Examples

  • Use 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
    
  • Use 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
    

15.6 liblc3

liblc3 LC3 (Low Complexity Communication Codec) encoder wrapper.

Requires the presence of the liblc3 headers and library during configuration. You need to explicitly configure the build with --enable-liblc3.

This encoder has support for the Bluetooth SIG LC3 codec for the LE Audio protocol, and the following features of LC3plus:

  • Frame duration of 2.5 and 5ms.
  • High-Resolution mode, 48 KHz, and 96 kHz sampling rates.

For more information see the liblc3 project at https://github.com/google/liblc3.

15.6.1 Options

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

b bitrate

Set the bit rate in bits/s. This will determine the fixed size of the encoded frames, for a selected frame duration.

ar frequency

Set the audio sampling rate (in Hz).

channels

Set the number of audio channels.

frame_duration

Set the audio frame duration in milliseconds. Default value is 10ms. Allowed frame durations are 2.5ms, 5ms, 7.5ms and 10ms. LC3 (Bluetooth LE Audio), allows 7.5ms and 10ms; and LC3plus 2.5ms, 5ms and 10ms.

The 10ms frame duration is available in LC3 and LC3 plus standard. In this mode, the produced bitstream can be referenced either as LC3 or LC3plus.

high_resolution boolean

Enable the high-resolution mode if set to 1. The high-resolution mode is available with all LC3plus frame durations and for a sampling rate of 48 KHz, and 96 KHz.

The encoder automatically turns off this mode at lower sampling rates and activates it at 96 KHz.

This mode should be preferred at high bitrates. In this mode, the audio bandwidth is always up to the Nyquist frequency, compared to LC3 at 48 KHz, which limits the bandwidth to 20 KHz.

15.7 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 --enable-libmp3lame.

See libshine for a fixed-point MP3 encoder, although with a lower quality.

15.7.1 Options

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

b (-b)

Set bitrate expressed in bits/s for CBR or ABR. LAME bitrate is expressed in kilobits/s.

q (-V)

Set constant quality setting for VBR. This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.

compression_level (-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.

cutoff (--lowpass)

Set lowpass cutoff frequency. If unspecified, the encoder dynamically adjusts the cutoff.

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 --nores option.

joint_stereo (-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.

abr (--abr)

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.

copyright (-c)

Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).

original (-o)

Set MPEG audio original flag when set to 1. The default value is 1 (enabled).

15.8 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 --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.

15.8.1 Options

b

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

4750
5150
5900
6700
7400
7950
10200
12200
dtx

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

15.9 libopus

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.

15.9.1 Option Mapping

Most libopus options are modelled 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.

b (bitrate)

Set the bit rate in bits/s. FFmpeg’s b option is expressed in bits/s, while opusenc’s bitrate in kilobits/s.

vbr (vbr, hard-cbr, and cvbr)

Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the opusenc equivalent options in parentheses:

off (hard-cbr)

Use constant bit rate encoding.

on (vbr)

Use variable bit rate encoding (the default).

constrained (cvbr)

Use constrained variable bit rate encoding.

compression_level (comp)

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.

frame_duration (framesize)

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.

packet_loss (expect-loss)

Set expected packet loss percentage. The default is 0.

fec (n/a)

Enable inband forward error correction. packet_loss must be non-zero to take advantage - frequency of FEC ’side-data’ is proportional to expected packet loss. Default is disabled.

application (N.A.)

Set intended application type. Valid options are listed below:

voip

Favor improved speech intelligibility.

audio

Favor faithfulness to the input (the default).

lowdelay

Restrict to only the lowest delay modes by disabling voice-optimized modes.

cutoff (N.A.)

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). Note that libopus forces a wideband cutoff for bitrates < 15 kbps, unless CELT-only (application set to ‘lowdelay’) mode is used.

mapping_family (mapping_family)

Set channel mapping family to be used by the encoder. The default value of -1 uses mapping family 0 for mono and stereo inputs, and mapping family 1 otherwise. The default also disables the surround masking and LFE bandwidth optimzations in libopus, and requires that the input contains 8 channels or fewer.

Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE bandwidth optimizations, and 255 for independent streams with an unspecified channel layout.

apply_phase_inv (N.A.) (requires libopus >= 1.2)

If set to 0, disables the use of phase inversion for intensity stereo, improving the quality of mono downmixes, but slightly reducing normal stereo quality. The default is 1 (phase inversion enabled).

15.10 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 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.

15.10.1 Options

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

b (-b)

Set bitrate expressed in bits/s for CBR. shineenc -b option is expressed in kilobits/s.

15.11 libtwolame

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.

15.11.1 Options

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

b (-b)

Set bitrate expressed in bits/s for CBR. twolame b option is expressed in kilobits/s. Default value is 128k.

q (-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 ffmpeg command-line tool. For library interface users, use global_quality.

mode (--mode)

Set the mode of the resulting audio. Possible values:

auto

Choose mode automatically based on the input. This is the default.

stereo

Stereo

joint_stereo

Joint stereo

dual_channel

Dual channel

mono

Mono

psymodel (--psyc-mode)

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.

energy_levels (--energy)

Enable energy levels extensions when set to 1. The default value is 0 (disabled).

error_protection (--protect)

Enable CRC error protection when set to 1. The default value is 0 (disabled).

copyright (--copyright)

Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).

original (--original)

Set MPEG audio original flag when set to 1. The default value is 0 (disabled).

15.12 libvo-amrwbenc

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.

15.12.1 Options

b

Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.

6600
8850
12650
14250
15850
18250
19850
23050
23850
dtx

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

15.13 libvorbis

libvorbis encoder wrapper.

Requires the presence of the libvorbisenc headers and library during configuration. You need to explicitly configure the build with --enable-libvorbis.

15.13.1 Options

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).

b (-b)

Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in kilobits/s.

q (-q)

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.

cutoff (--advanced-encode-option lowpass_frequency=N)

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).

minrate (-m)

Set minimum bitrate expressed in bits/s. oggenc -m is expressed in kilobits/s.

maxrate (-M)

Set maximum bitrate expressed in bits/s. oggenc -M is expressed in kilobits/s. This only has effect on ABR mode.

iblock (--advanced-encode-option impulse_noisetune=N)

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.

15.14 mjpeg

Motion JPEG encoder.

15.14.1 Options

huffman

Set the huffman encoding strategy. Possible values:

default

Use the default huffman tables. This is the default strategy.

optimal

Compute and use optimal huffman tables.

15.15 wavpack

WavPack lossless audio encoder.

15.15.1 Options

The equivalent options for wavpack command line utility are listed in parentheses.

15.15.1.1 Shared options

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.

frame_size (--blocksize)

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.

compression_level (-f, -h, -hh, and -x)

15.15.1.2 Private options

joint_stereo (-j)

Set whether to enable joint stereo. Valid values are:

on (1)

Force mid/side audio encoding.

off (0)

Force left/right audio encoding.

auto

Let the encoder decide automatically.

optimize_mono

Set whether to enable optimization for mono. This option is only effective for non-mono streams. Available values:

on

enabled

off

disabled

16 Video Encoders

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

16.1 a64_multi, a64_multi5

A64 / Commodore 64 multicolor charset encoder. a64_multi5 is extended with 5th color (colram).

16.2 Cinepak

Cinepak aka CVID encoder. Compatible with Windows 3.1 and vintage MacOS.

16.2.1 Options

g integer

Keyframe interval. A keyframe is inserted at least every -g frames, sometimes sooner.

q:v integer

Quality factor. Lower is better. Higher gives lower bitrate. The following table lists bitrates when encoding akiyo_cif.y4m for various values of -q:v with -g 100:

-q:v 1 1918 kb/s
-q:v 2 1735 kb/s
-q:v 4 1500 kb/s
-q:v 10 1041 kb/s
-q:v 20 826 kb/s
-q:v 40 553 kb/s
-q:v 100 394 kb/s
-q:v 200 312 kb/s
-q:v 400 266 kb/s
-q:v 1000 237 kb/s
max_extra_cb_iterations integer

Max extra codebook recalculation passes, more is better and slower.

skip_empty_cb boolean

Avoid wasting bytes, ignore vintage MacOS decoder.

max_strips integer
min_strips integer

The minimum and maximum number of strips to use. Wider range sometimes improves quality. More strips is generally better quality but costs more bits. Fewer strips tend to yield more keyframes. Vintage compatible is 1..3.

strip_number_adaptivity integer

How much number of strips is allowed to change between frames. Higher is better but slower.

16.3 GIF

GIF image/animation encoder.

16.3.1 Options

gifflags integer

Sets the flags used for GIF encoding.

offsetting

Enables picture offsetting.

Default is enabled.

transdiff

Enables transparency detection between frames.

Default is enabled.

gifimage integer

Enables encoding one full GIF image per frame, rather than an animated GIF.

Default value is 0.

global_palette integer

Writes a palette to the global GIF header where feasible.

If disabled, every frame will always have a palette written, even if there is a global palette supplied.

Default value is 1.

16.4 Hap

Vidvox Hap video encoder.

16.4.1 Options

format integer

Specifies the Hap format to encode.

hap
hap_alpha
hap_q

Default value is hap.

chunks integer

Specifies the number of chunks to split frames into, between 1 and 64. This permits multithreaded decoding of large frames, potentially at the cost of data-rate. The encoder may modify this value to divide frames evenly.

Default value is 1.

compressor integer

Specifies the second-stage compressor to use. If set to none, chunks will be limited to 1, as chunked uncompressed frames offer no benefit.

none
snappy

Default value is snappy.

16.5 jpeg2000

The native jpeg 2000 encoder is lossy by default, the -q:v option can be used to set the encoding quality. Lossless encoding can be selected with -pred 1.

16.5.1 Options

format integer

Can be set to either j2k or jp2 (the default) that makes it possible to store non-rgb pix_fmts.

tile_width integer

Sets tile width. Range is 1 to 1073741824. Default is 256.

tile_height integer

Sets tile height. Range is 1 to 1073741824. Default is 256.

pred integer

Allows setting the discrete wavelet transform (DWT) type

dwt97int (Lossy)
dwt53 (Lossless)

Default is dwt97int

sop boolean

Enable this to add SOP marker at the start of each packet. Disabled by default.

eph boolean

Enable this to add EPH marker at the end of each packet header. Disabled by default.

prog integer

Sets the progression order to be used by the encoder. Possible values are:

lrcp
rlcp
rpcl
pcrl
cprl

Set to lrcp by default.

layer_rates string

By default, when this option is not used, compression is done using the quality metric. This option allows for compression using compression ratio. The compression ratio for each level could be specified. The compression ratio of a layer l species the what ratio of total file size is contained in the first l layers.

Example usage:

ffmpeg -i input.bmp -c:v jpeg2000 -layer_rates "100,10,1" output.j2k

This would compress the image to contain 3 layers, where the data contained in the first layer would be compressed by 1000 times, compressed by 100 in the first two layers, and shall contain all data while using all 3 layers.

16.6 librav1e

rav1e AV1 encoder wrapper.

Requires the presence of the rav1e headers and library during configuration. You need to explicitly configure the build with --enable-librav1e.

16.6.1 Options

qmax

Sets the maximum quantizer to use when using bitrate mode.

qmin

Sets the minimum quantizer to use when using bitrate mode.

qp

Uses quantizer mode to encode at the given quantizer (0-255).

speed

Selects the speed preset (0-10) to encode with.

tiles

Selects how many tiles to encode with.

tile-rows

Selects how many rows of tiles to encode with.

tile-columns

Selects how many columns of tiles to encode with.

rav1e-params

Set rav1e options using a list of key=value pairs separated by ":". See rav1e --help for a list of options.

For example to specify librav1e encoding options with -rav1e-params:

ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4

16.7 libaom-av1

libaom AV1 encoder wrapper.

Requires the presence of the libaom headers and library during configuration. You need to explicitly configure the build with --enable-libaom.

16.7.1 Options

The wrapper supports the following standard libavcodec options:

b

Set bitrate target in bits/second. By default this will use variable-bitrate mode. If maxrate and minrate are also set to the same value then it will use constant-bitrate mode, otherwise if crf is set as well then it will use constrained-quality mode.

g keyint_min

Set key frame placement. The GOP size sets the maximum distance between key frames; if zero the output stream will be intra-only. The minimum distance is ignored unless it is the same as the GOP size, in which case key frames will always appear at a fixed interval. Not set by default, so without this option the library has completely free choice about where to place key frames.

qmin qmax

Set minimum/maximum quantisation values. Valid range is from 0 to 63 (warning: this does not match the quantiser values actually used by AV1 - divide by four to map real quantiser values to this range). Defaults to min/max (no constraint).

minrate maxrate bufsize rc_init_occupancy

Set rate control buffering parameters. Not used if not set - defaults to unconstrained variable bitrate.

threads

Set the number of threads to use while encoding. This may require the tiles or row-mt options to also be set to actually use the specified number of threads fully. Defaults to the number of hardware threads supported by the host machine.

profile

Set the encoding profile. Defaults to using the profile which matches the bit depth and chroma subsampling of the input.

The wrapper also has some specific options:

cpu-used

Set the quality/encoding speed tradeoff. Valid range is from 0 to 8, higher numbers indicating greater speed and lower quality. The default value is 1, which will be slow and high quality.

auto-alt-ref

Enable use of alternate reference frames. Defaults to the internal default of the library.

arnr-max-frames (frames)

Set altref noise reduction max frame count. Default is -1.

arnr-strength (strength)

Set altref noise reduction filter strength. Range is -1 to 6. Default is -1.

aq-mode (aq-mode)

Set adaptive quantization mode. Possible values:

none (0)

Disabled.

variance (1)

Variance-based.

complexity (2)

Complexity-based.

cyclic (3)

Cyclic refresh.

tune (tune)

Set the distortion metric the encoder is tuned with. Default is psnr.

psnr (0)
ssim (1)
lag-in-frames

Set the maximum number of frames which the encoder may keep in flight at any one time for lookahead purposes. Defaults to the internal default of the library.

error-resilience

Enable error resilience features:

default

Improve resilience against losses of whole frames.

Not enabled by default.

crf

Set the quality/size tradeoff for constant-quality (no bitrate target) and constrained-quality (with maximum bitrate target) modes. Valid range is 0 to 63, higher numbers indicating lower quality and smaller output size. Only used if set; by default only the bitrate target is used.

static-thresh

Set a change threshold on blocks below which they will be skipped by the encoder. Defined in arbitrary units as a nonnegative integer, defaulting to zero (no blocks are skipped).

drop-threshold

Set a threshold for dropping frames when close to rate control bounds. Defined as a percentage of the target buffer - when the rate control buffer falls below this percentage, frames will be dropped until it has refilled above the threshold. Defaults to zero (no frames are dropped).

denoise-noise-level (level)

Amount of noise to be removed for grain synthesis. Grain synthesis is disabled if this option is not set or set to 0.

denoise-block-size (pixels)

Block size used for denoising for grain synthesis. If not set, AV1 codec uses the default value of 32.

undershoot-pct (pct)

Set datarate undershoot (min) percentage of the target bitrate. Range is -1 to 100. Default is -1.

overshoot-pct (pct)

Set datarate overshoot (max) percentage of the target bitrate. Range is -1 to 1000. Default is -1.

minsection-pct (pct)

Minimum percentage variation of the GOP bitrate from the target bitrate. If minsection-pct is not set, the libaomenc wrapper computes it as follows: (minrate * 100 / bitrate). Range is -1 to 100. Default is -1 (unset).

maxsection-pct (pct)

Maximum percentage variation of the GOP bitrate from the target bitrate. If maxsection-pct is not set, the libaomenc wrapper computes it as follows: (maxrate * 100 / bitrate). Range is -1 to 5000. Default is -1 (unset).

frame-parallel (boolean)

Enable frame parallel decodability features. Default is true.

tiles

Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow greater parallelism in both encoding and decoding, but may decrease coding efficiency. Defaults to the minimum number of tiles required by the size of the input video (this is 1x1 (that is, a single tile) for sizes up to and including 4K).

tile-columns tile-rows

Set the number of tiles as log2 of the number of tile rows and columns. Provided for compatibility with libvpx/VP9.

row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2)

Enable row based multi-threading. Disabled by default.

enable-cdef (boolean)

Enable Constrained Directional Enhancement Filter. The libaom-av1 encoder enables CDEF by default.

enable-restoration (boolean)

Enable Loop Restoration Filter. Default is true for libaom-av1.

enable-global-motion (boolean)

Enable the use of global motion for block prediction. Default is true.

enable-intrabc (boolean)

Enable block copy mode for intra block prediction. This mode is useful for screen content. Default is true.

enable-rect-partitions (boolean) (Requires libaom >= v2.0.0)

Enable rectangular partitions. Default is true.

enable-1to4-partitions (boolean) (Requires libaom >= v2.0.0)

Enable 1:4/4:1 partitions. Default is true.

enable-ab-partitions (boolean) (Requires libaom >= v2.0.0)

Enable AB shape partitions. Default is true.

enable-angle-delta (boolean) (Requires libaom >= v2.0.0)

Enable angle delta intra prediction. Default is true.

enable-cfl-intra (boolean) (Requires libaom >= v2.0.0)

Enable chroma predicted from luma intra prediction. Default is true.

enable-filter-intra (boolean) (Requires libaom >= v2.0.0)

Enable filter intra predictor. Default is true.

enable-intra-edge-filter (boolean) (Requires libaom >= v2.0.0)

Enable intra edge filter. Default is true.

enable-smooth-intra (boolean) (Requires libaom >= v2.0.0)

Enable smooth intra prediction mode. Default is true.

enable-paeth-intra (boolean) (Requires libaom >= v2.0.0)

Enable paeth predictor in intra prediction. Default is true.

enable-palette (boolean) (Requires libaom >= v2.0.0)

Enable palette prediction mode. Default is true.

enable-flip-idtx (boolean) (Requires libaom >= v2.0.0)

Enable extended transform type, including FLIPADST_DCT, DCT_FLIPADST, FLIPADST_FLIPADST, ADST_FLIPADST, FLIPADST_ADST, IDTX, V_DCT, H_DCT, V_ADST, H_ADST, V_FLIPADST, H_FLIPADST. Default is true.

enable-tx64 (boolean) (Requires libaom >= v2.0.0)

Enable 64-pt transform. Default is true.

reduced-tx-type-set (boolean) (Requires libaom >= v2.0.0)

Use reduced set of transform types. Default is false.

use-intra-dct-only (boolean) (Requires libaom >= v2.0.0)

Use DCT only for INTRA modes. Default is false.

use-inter-dct-only (boolean) (Requires libaom >= v2.0.0)

Use DCT only for INTER modes. Default is false.

use-intra-default-tx-only (boolean) (Requires libaom >= v2.0.0)

Use Default-transform only for INTRA modes. Default is false.

enable-ref-frame-mvs (boolean) (Requires libaom >= v2.0.0)

Enable temporal mv prediction. Default is true.

enable-reduced-reference-set (boolean) (Requires libaom >= v2.0.0)

Use reduced set of single and compound references. Default is false.

enable-obmc (boolean) (Requires libaom >= v2.0.0)

Enable obmc. Default is true.

enable-dual-filter (boolean) (Requires libaom >= v2.0.0)

Enable dual filter. Default is true.

enable-diff-wtd-comp (boolean) (Requires libaom >= v2.0.0)

Enable difference-weighted compound. Default is true.

enable-dist-wtd-comp (boolean) (Requires libaom >= v2.0.0)

Enable distance-weighted compound. Default is true.

enable-onesided-comp (boolean) (Requires libaom >= v2.0.0)

Enable one sided compound. Default is true.

enable-interinter-wedge (boolean) (Requires libaom >= v2.0.0)

Enable interinter wedge compound. Default is true.

enable-interintra-wedge (boolean) (Requires libaom >= v2.0.0)

Enable interintra wedge compound. Default is true.

enable-masked-comp (boolean) (Requires libaom >= v2.0.0)

Enable masked compound. Default is true.

enable-interintra-comp (boolean) (Requires libaom >= v2.0.0)

Enable interintra compound. Default is true.

enable-smooth-interintra (boolean) (Requires libaom >= v2.0.0)

Enable smooth interintra mode. Default is true.

aom-params

Set libaom options using a list of key=value pairs separated by ":". For a list of supported options, see aomenc --help under the section "AV1 Specific Options".

For example to specify libaom encoding options with -aom-params:

ffmpeg -i input -c:v libaom-av1 -b:v 500K -aom-params tune=psnr:enable-tpl-model=1 output.mp4

16.8 libsvtav1

SVT-AV1 encoder wrapper.

Requires the presence of the SVT-AV1 headers and library during configuration. You need to explicitly configure the build with --enable-libsvtav1.

16.8.1 Options

profile

Set the encoding profile.

main
high
professional
level

Set the operating point level. For example: ’4.0’

hielevel

Set the Hierarchical prediction levels.

3level
4level

This is the default.

tier

Set the operating point tier.

main

This is the default.

high
qmax

Set the maximum quantizer to use when using a bitrate mode.

qmin

Set the minimum quantizer to use when using a bitrate mode.

crf

Constant rate factor value used in crf rate control mode (0-63).

qp

Set the quantizer used in cqp rate control mode (0-63).

sc_detection

Enable scene change detection.

la_depth

Set number of frames to look ahead (0-120).

preset

Set the quality-speed tradeoff, in the range 0 to 13. Higher values are faster but lower quality.

tile_rows

Set log2 of the number of rows of tiles to use (0-6).

tile_columns

Set log2 of the number of columns of tiles to use (0-4).

svtav1-params

Set SVT-AV1 options using a list of key=value pairs separated by ":". See the SVT-AV1 encoder user guide for a list of accepted parameters.

16.9 libjxl

libjxl JPEG XL encoder wrapper.

Requires the presence of the libjxl headers and library during configuration. You need to explicitly configure the build with --enable-libjxl.

16.9.1 Options

The libjxl wrapper supports the following options:

distance

Set the target Butteraugli distance. This is a quality setting: lower distance yields higher quality, with distance=1.0 roughly comparable to libjpeg Quality 90 for photographic content. Setting distance=0.0 yields true lossless encoding. Valid values range between 0.0 and 15.0, and sane values rarely exceed 5.0. Setting distance=0.1 usually attains transparency for most input. The default is 1.0.

effort

Set the encoding effort used. Higher effort values produce more consistent quality and usually produces a better quality/bpp curve, at the cost of more CPU time required. Valid values range from 1 to 9, and the default is 7.

modular

Force the encoder to use Modular mode instead of choosing automatically. The default is to use VarDCT for lossy encoding and Modular for lossless. VarDCT is generally superior to Modular for lossy encoding but does not support lossless encoding.

16.10 libkvazaar

Kvazaar H.265/HEVC encoder.

Requires the presence of the libkvazaar headers and library during configuration. You need to explicitly configure the build with --enable-libkvazaar.

16.10.1 Options

b

Set target video bitrate in bit/s and enable rate control.

kvazaar-params

Set kvazaar parameters as a list of name=value pairs separated by commas (,). See kvazaar documentation for a list of options.

16.11 libopenh264

Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

This encoder requires the presence of the libopenh264 headers and library during configuration. You need to explicitly configure the build with --enable-libopenh264. The library is detected using pkg-config.

For more information about the library see http://www.openh264.org.

16.11.1 Options

The following FFmpeg global options affect the configurations of the libopenh264 encoder.

b

Set the bitrate (as a number of bits per second).

g

Set the GOP size.

maxrate

Set the max bitrate (as a number of bits per second).

flags +global_header

Set global header in the bitstream.

slices

Set the number of slices, used in parallelized encoding. Default value is 0. This is only used when slice_mode is set to ‘fixed’.

loopfilter

Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0.

profile

Set profile restrictions. If set to the value of ‘main’ enable CABAC (set the SEncParamExt.iEntropyCodingModeFlag flag to 1).

max_nal_size

Set maximum NAL size in bytes.

allow_skip_frames

Allow skipping frames to hit the target bitrate if set to 1.

16.12 libtheora

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 information about the libtheora project see http://www.theora.org/.

16.12.1 Options

The following global options are mapped to internal libtheora options which affect the quality and the bitrate of the encoded stream.

b

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.

flags

Used to enable constant quality mode (VBR) encoding through the qscale flag, and to enable the pass1 and pass2 modes.

g

Set the GOP size.

global_quality

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.

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 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.

16.12.2 Examples

  • Set maximum constant quality (VBR) encoding with ffmpeg:
    ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
    
  • Use ffmpeg to convert a CBR 1000 kbps Theora video stream:
    ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
    

16.13 libvpx

VP8/VP9 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.

16.13.1 Options

The following options are supported by the libvpx wrapper. The vpxenc-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 more documentation of the libvpx options, invoke the command ffmpeg -h encoder=libvpx, ffmpeg -h encoder=libvpx-vp9 or vpxenc --help. Further information is available in the libvpx API documentation.

b (target-bitrate)

Set bitrate in bits/s. Note that FFmpeg’s b option is expressed in bits/s, while vpxenc’s target-bitrate is in kilobits/s.

g (kf-max-dist)
keyint_min (kf-min-dist)
qmin (min-q)

Minimum (Best Quality) Quantizer.

qmax (max-q)

Maximum (Worst Quality) Quantizer. Can be changed per-frame.

bufsize (buf-sz, buf-optimal-sz)

Set ratecontrol buffer size (in bits). Note vpxenc’s options are specified in milliseconds, the libvpx wrapper converts this value as follows: buf-sz = bufsize * 1000 / bitrate, buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6.

rc_init_occupancy (buf-initial-sz)

Set number of bits which should be loaded into the rc buffer before decoding starts. Note vpxenc’s option is specified in milliseconds, the libvpx wrapper converts this value as follows: rc_init_occupancy * 1000 / bitrate.

undershoot-pct

Set datarate undershoot (min) percentage of the target bitrate.

overshoot-pct

Set datarate overshoot (max) percentage of the target bitrate.

skip_threshold (drop-frame)
qcomp (bias-pct)
maxrate (maxsection-pct)

Set GOP max bitrate in bits/s. Note vpxenc’s option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: (maxrate * 100 / bitrate).

minrate (minsection-pct)

Set GOP min bitrate in bits/s. Note vpxenc’s option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: (minrate * 100 / bitrate).

minrate, maxrate, b end-usage=cbr

(minrate == maxrate == bitrate).

crf (end-usage=cq, cq-level)
tune (tune)
psnr (psnr)
ssim (ssim)
quality, deadline (deadline)
best

Use best quality deadline. Poorly named and quite slow, this option should be avoided as it may give worse quality output than good.

good

Use good quality deadline. This is a good trade-off between speed and quality when used with the cpu-used option.

realtime

Use realtime quality deadline.

speed, cpu-used (cpu-used)

Set quality/speed ratio modifier. Higher values speed up the encode at the cost of quality.

nr (noise-sensitivity)
static-thresh

Set a change threshold on blocks below which they will be skipped by the encoder.

slices (token-parts)

Note that FFmpeg’s slices option gives the total number of partitions, while vpxenc’s token-parts is given as log2(partitions).

max-intra-rate

Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means unlimited.

force_key_frames

VPX_EFLAG_FORCE_KF

Alternate reference frame related
auto-alt-ref

Enable use of alternate reference frames (2-pass only). Values greater than 1 enable multi-layer alternate reference frames (VP9 only).

arnr-maxframes

Set altref noise reduction max frame count.

arnr-type

Set altref noise reduction filter type: backward, forward, centered.

arnr-strength

Set altref noise reduction filter strength.

rc-lookahead, lag-in-frames (lag-in-frames)

Set number of frames to look ahead for frametype and ratecontrol.

min-gf-interval

Set minimum golden/alternate reference frame interval (VP9 only).

error-resilient

Enable error resiliency features.

sharpness integer

Increase sharpness at the expense of lower PSNR. The valid range is [0, 7].

ts-parameters

Sets the temporal scalability configuration using a :-separated list of key=value pairs. For example, to specify temporal scalability parameters with ffmpeg:

ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\
ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT

Below is a brief explanation of each of the parameters, please refer to struct vpx_codec_enc_cfg in vpx/vpx_encoder.h for more details.

ts_number_layers

Number of temporal coding layers.

ts_target_bitrate

Target bitrate for each temporal layer (in kbps). (bitrate should be inclusive of the lower temporal layer).

ts_rate_decimator

Frame rate decimation factor for each temporal layer.

ts_periodicity

Length of the sequence defining frame temporal layer membership.

ts_layer_id

Template defining the membership of frames to temporal layers.

ts_layering_mode

(optional) Selecting the temporal structure from a set of pre-defined temporal layering modes. Currently supports the following options.

0

No temporal layering flags are provided internally, relies on flags being passed in using metadata field in AVFrame with following keys.

vp8-flags

Sets the flags passed into the encoder to indicate the referencing scheme for the current frame. Refer to function vpx_codec_encode in vpx/vpx_encoder.h for more details.

temporal_id

Explicitly sets the temporal id of the current frame to encode.

2

Two temporal layers. 0-1...

3

Three temporal layers. 0-2-1-2...; with single reference frame.

4

Same as option "3", except there is a dependency between the two temporal layer 2 frames within the temporal period.

VP8-specific options
screen-content-mode

Screen content mode, one of: 0 (off), 1 (screen), 2 (screen with more aggressive rate control).

VP9-specific options
lossless

Enable lossless mode.

tile-columns

Set number of tile columns to use. Note this is given as log2(tile_columns). For example, 8 tile columns would be requested by setting the tile-columns option to 3.

tile-rows

Set number of tile rows to use. Note this is given as log2(tile_rows). For example, 4 tile rows would be requested by setting the tile-rows option to 2.

frame-parallel

Enable frame parallel decodability features.

aq-mode

Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3: cyclic refresh, 4: equator360).

colorspace color-space

Set input color space. The VP9 bitstream supports signaling the following colorspaces:

rgbsRGB
bt709bt709
unspecifiedunknown
bt470bgbt601
smpte170msmpte170
smpte240msmpte240
bt2020_nclbt2020
row-mt boolean

Enable row based multi-threading.

tune-content

Set content type: default (0), screen (1), film (2).

corpus-complexity

Corpus VBR mode is a variant of standard VBR where the complexity distribution midpoint is passed in rather than calculated for a specific clip or chunk.

The valid range is [0, 10000]. 0 (default) uses standard VBR.

enable-tpl boolean

Enable temporal dependency model.

ref-frame-config

Using per-frame metadata, set members of the structure vpx_svc_ref_frame_config_t in vpx/vp8cx.h to fine-control referencing schemes and frame buffer management.
Use a :-separated list of key=value pairs. For example,

av_dict_set(&av_frame->metadata, "ref-frame-config", \
"rfc_update_buffer_slot=7:rfc_lst_fb_idx=0:rfc_gld_fb_idx=1:rfc_alt_fb_idx=2:rfc_reference_last=0:rfc_reference_golden=0:rfc_reference_alt_ref=0");
rfc_update_buffer_slot

Indicates the buffer slot number to update

rfc_update_last

Indicates whether to update the LAST frame

rfc_update_golden

Indicates whether to update GOLDEN frame

rfc_update_alt_ref

Indicates whether to update ALT_REF frame

rfc_lst_fb_idx

LAST frame buffer index

rfc_gld_fb_idx

GOLDEN frame buffer index

rfc_alt_fb_idx

ALT_REF frame buffer index

rfc_reference_last

Indicates whether to reference LAST frame

rfc_reference_golden

Indicates whether to reference GOLDEN frame

rfc_reference_alt_ref

Indicates whether to reference ALT_REF frame

rfc_reference_duration

Indicates frame duration

For more information about libvpx see: http://www.webmproject.org/

16.14 libvvenc

VVenC H.266/VVC encoder wrapper.

This encoder requires the presence of the libvvenc headers and library during configuration. You need to explicitly configure the build with --enable-libvvenc.

The VVenC project website is at https://github.com/fraunhoferhhi/vvenc.

16.14.1 Supported Pixel Formats

VVenC supports only 10-bit color spaces as input. But the internal (encoded) bit depth can be set to 8-bit or 10-bit at runtime.

16.14.2 Options

b

Sets target video bitrate.

g

Set the GOP size. Currently support for g=1 (Intra only) or default.

preset

Set the VVenC preset.

levelidc

Set level idc.

tier

Set vvc tier.

qp

Set constant quantization parameter.

subopt boolean

Set subjective (perceptually motivated) optimization. Default is 1 (on).

bitdepth8 boolean

Set 8bit coding mode instead of using 10bit. Default is 0 (off).

period

set (intra) refresh period in seconds.

vvenc-params

Set vvenc options using a list of key=value couples separated by ":". See vvencapp --fullhelp or vvencFFapp --fullhelp for a list of options.

For example, the options might be provided as:

intraperiod=64:decodingrefreshtype=idr:poc0idr=1:internalbitdepth=8

For example the encoding options might be provided with -vvenc-params:

ffmpeg -i input -c:v libvvenc -b 1M -vvenc-params intraperiod=64:decodingrefreshtype=idr:poc0idr=1:internalbitdepth=8 output.mp4

16.15 libwebp

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.

16.15.1 Pixel Format

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.

16.15.2 Options

-lossless boolean

Enables/Disables use of lossless mode. Default is 0.

-compression_level integer

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.

-quality float

For lossy encoding, this controls image quality. For lossless encoding, this controls the effort and time spent in compression. Range is 0 to 100. Default is 75.

-preset type

Configuration preset. This does some automatic settings based on the general type of the image.

none

Do not use a preset.

default

Use the encoder default.

picture

Digital picture, like portrait, inner shot

photo

Outdoor photograph, with natural lighting

drawing

Hand or line drawing, with high-contrast details

icon

Small-sized colorful images

text

Text-like

16.16 libx264, libx264rgb

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 one 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.

16.16.1 Supported Pixel Formats

x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264’s configure time.

16.16.2 Options

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 --fullhelp or consult the libx264 documentation.

In the list below, note that the x264 option name is shown in parentheses after the libavcodec corresponding name, in case there is a direct mapping.

b (bitrate)

Set bitrate in bits/s. Note that FFmpeg’s b option is expressed in bits/s, while x264’s bitrate is in kilobits/s.

bf (bframes)

Number of B-frames between I and P-frames

g (keyint)

Maximum GOP size

qmin (qpmin)

Minimum quantizer scale

qmax (qpmax)

Maximum quantizer scale

qdiff (qpstep)

Maximum difference between quantizer scales

qblur (qblur)

Quantizer curve blur

qcomp (qcomp)

Quantizer curve compression factor

refs (ref)

Number of reference frames each P-frame can use. The range is 0-16.

level (level)

Set the x264_param_t.i_level_idc value in case the value is positive, it is ignored otherwise.

This value can be set using the AVCodecContext API (e.g. by setting the AVCodecContext value directly), and is specified as an integer mapped on a corresponding level (e.g. the value 31 maps to H.264 level IDC "3.1", as defined in the x264_levels table). It is ignored when set to a non positive value.

Alternatively it can be set as a private option, overriding the value set in AVCodecContext, and in this case must be specified as the level IDC identifier (e.g. "3.1"), as defined by H.264 Annex A.

sc_threshold (scenecut)

Sets the threshold for the scene change detection.

trellis (trellis)

Performs Trellis quantization to increase efficiency. Enabled by default.

nr (nr)

Noise reduction

me_range (merange)

Maximum range of the motion search in pixels.

me_method (me)

Set motion estimation method. Possible values in the decreasing order of speed:

dia (dia)
epzs (dia)

Diamond search with radius 1 (fastest). ‘epzs’ is an alias for ‘dia’.

hex (hex)

Hexagonal search with radius 2.

umh (umh)

Uneven multi-hexagon search.

esa (esa)

Exhaustive search.

tesa (tesa)

Hadamard exhaustive search (slowest).

forced-idr

Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces it to choose an IDR-frame.

subq (subme)

Sub-pixel motion estimation method.

b_strategy (b-adapt)

Adaptive B-frame placement decision algorithm. Use only on first-pass.

keyint_min (min-keyint)

Minimum GOP size.

coder

Set entropy encoder. Possible values:

ac

Enable CABAC.

vlc

Enable CAVLC and disable CABAC. It generates the same effect as x264’s --no-cabac option.

cmp

Set full pixel motion estimation comparison algorithm. Possible values:

chroma

Enable chroma in motion estimation.

sad

Ignore chroma in motion estimation. It generates the same effect as x264’s --no-chroma-me option.

threads (threads)

Number of encoding threads.

thread_type

Set multithreading technique. Possible values:

slice

Slice-based multithreading. It generates the same effect as x264’s --sliced-threads option.

frame

Frame-based multithreading.

flags

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.

rc_init_occupancy (vbv-init)

Initial VBV buffer occupancy

preset (preset)

Set the encoding preset.

tune (tune)

Set tuning of the encoding params.

profile (profile)

Set profile restrictions.

fastfirstpass

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.

crf (crf)

Set the quality for constant quality mode.

crf_max (crf-max)

In CRF mode, prevents VBV from lowering quality beyond this point.

qp (qp)

Set constant quantization rate control method parameter.

aq-mode (aq-mode)

Set AQ method. Possible values:

none (0)

Disabled.

variance (1)

Variance AQ (complexity mask).

autovariance (2)

Auto-variance AQ (experimental).

aq-strength (aq-strength)

Set AQ strength, reduce blocking and blurring in flat and textured areas.

psy

Use psychovisual optimizations when set to 1. When set to 0, it has the same effect as x264’s --no-psy option.

psy-rd (psy-rd)

Set strength of psychovisual optimization, in psy-rd:psy-trellis format.

rc-lookahead (rc-lookahead)

Set number of frames to look ahead for frametype and ratecontrol.

weightb

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.

weightp (weightp)

Set weighted prediction method for P-frames. Possible values:

none (0)

Disabled

simple (1)

Enable only weighted refs

smart (2)

Enable both weighted refs and duplicates

ssim (ssim)

Enable calculation and printing SSIM stats after the encoding.

intra-refresh (intra-refresh)

Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.

avcintra-class (class)

Configure the encoder to generate AVC-Intra. Valid values are 50, 100 and 200

bluray-compat (bluray-compat)

Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting "bluray-compat=1 force-cfr=1".

b-bias (b-bias)

Set the influence on how often B-frames are used.

b-pyramid (b-pyramid)

Set method for keeping of some B-frames as references. Possible values:

none (none)

Disabled.

strict (strict)

Strictly hierarchical pyramid.

normal (normal)

Non-strict (not Blu-ray compatible).

mixed-refs

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.

8x8dct

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.

fast-pskip

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.

aud (aud)

Enable use of access unit delimiters when set to 1.

mbtree

Enable use macroblock tree ratecontrol when set to 1. When set to 0, it has the same effect as x264’s --no-mbtree option.

deblock (deblock)

Set loop filter parameters, in alpha:beta form.

cplxblur (cplxblur)

Set fluctuations reduction in QP (before curve compression).

partitions (partitions)

Set partitions to consider as a comma-separated list of values. Possible values in the list:

p8x8

8x8 P-frame partition.

p4x4

4x4 P-frame partition.

b8x8

4x4 B-frame partition.

i8x8

8x8 I-frame partition.

i4x4

4x4 I-frame partition. (Enabling ‘p4x4’ requires ‘p8x8’ to be enabled. Enabling ‘i8x8’ requires adaptive spatial transform (8x8dct option) to be enabled.)

none (none)

Do not consider any partitions.

all (all)

Consider every partition.

direct-pred (direct)

Set direct MV prediction mode. Possible values:

none (none)

Disable MV prediction.

spatial (spatial)

Enable spatial predicting.

temporal (temporal)

Enable temporal predicting.

auto (auto)

Automatically decided.

slice-max-size (slice-max-size)

Set the limit of the size of each slice in bytes. If not specified but RTP payload size (ps) is specified, that is used.

stats (stats)

Set the file name for multi-pass stats.

nal-hrd (nal-hrd)

Set signal HRD information (requires vbv-bufsize to be set). Possible values:

none (none)

Disable HRD information signaling.

vbr (vbr)

Variable bit rate.

cbr (cbr)

Constant bit rate (not allowed in MP4 container).

x264opts opts
x264-params opts

Override the x264 configuration using a :-separated list of key=value options.

The argument for both options is a list of key=value couples separated by ":". With x264opts the value can be omitted, and the value 1 is assumed in that case.

For filter and psy-rd options values 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, the options might be provided as:

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

For example to specify libx264 encoding options with ffmpeg:

ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv

To get the complete list of the libx264 options, invoke the command x264 --fullhelp or consult the libx264 documentation.

a53cc boolean

Import closed captions (which must be ATSC compatible format) into output. Only the mpeg2 and h264 decoders provide these. Default is 1 (on).

udu_sei boolean

Import user data unregistered SEI if available into output. Default is 0 (off).

mb_info boolean

Set mb_info data through AVFrameSideData, only useful when used from the API. Default is 0 (off).

Encoding ffpresets for common usages are provided so they can be used with the general presets system (e.g. passing the pre option).

16.17 libx265

x265 H.265/HEVC encoder wrapper.

This encoder requires the presence of the libx265 headers and library during configuration. You need to explicitly configure the build with --enable-libx265.

16.17.1 Options

b

Sets target video bitrate.

bf
g

Set the GOP size.

keyint_min

Minimum GOP size.

refs

Number of reference frames each P-frame can use. The range is from 1-16.

preset

Set the x265 preset.

tune

Set the x265 tune parameter.

profile

Set profile restrictions.

crf

Set the quality for constant quality mode.

qp

Set constant quantization rate control method parameter.

qmin

Minimum quantizer scale.

qmax

Maximum quantizer scale.

qdiff

Maximum difference between quantizer scales.

qblur

Quantizer curve blur

qcomp

Quantizer curve compression factor

i_qfactor
b_qfactor
forced-idr

Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces it to choose an IDR-frame.

udu_sei boolean

Import user data unregistered SEI if available into output. Default is 0 (off).

x265-params

Set x265 options using a list of key=value couples separated by ":". See x265 --help for a list of options.

For example to specify libx265 encoding options with -x265-params:

ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4

16.18 libxavs2

xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.

This encoder requires the presence of the libxavs2 headers and library during configuration. You need to explicitly configure the build with --enable-libxavs2.

The following standard libavcodec options are used:

  • b / bit_rate
  • g / gop_size
  • bf / max_b_frames

The encoder also has its own specific options:

16.18.1 Options

lcu_row_threads

Set the number of parallel threads for rows from 1 to 8 (default 5).

initial_qp

Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the initial qp for the first frame.

qp

Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the qp value under constant-QP mode.

max_qp

Set the max qp for rate control from 1 to 63 (default 55).

min_qp

Set the min qp for rate control from 1 to 63 (default 20).

speed_level

Set the Speed level from 0 to 9 (default 0). Higher is better but slower.

log_level

Set the log level from -1 to 3 (default 0). -1: none, 0: error, 1: warning, 2: info, 3: debug.

xavs2-params

Set xavs2 options using a list of key=value couples separated by ":".

For example to specify libxavs2 encoding options with -xavs2-params:

ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2

16.19 libxeve

eXtra-fast Essential Video Encoder (XEVE) MPEG-5 EVC encoder wrapper. The xeve-equivalent options or values are listed in parentheses for easy migration.

This encoder requires the presence of the libxeve headers and library during configuration. You need to explicitly configure the build with --enable-libxeve.

Many libxeve encoder options are mapped to FFmpeg global codec options, while unique encoder options are provided through private options. Additionally the xeve-params private options allows one to pass a list of key=value tuples as accepted by the libxeve parse_xeve_params function.

The xeve project website is at https://github.com/mpeg5/xeve.

16.19.1 Options

The following options are supported by the libxeve wrapper. The xeve-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 libxeve options, invoke the command xeve_app --help or consult the libxeve documentation.

b (bitrate)

Set target video bitrate in bits/s. Note that FFmpeg’s b option is expressed in bits/s, while xeve’s bitrate is in kilobits/s.

bf (bframes)

Set the maximum number of B frames (1,3,7,15).

g (keyint)

Set the GOP size (I-picture period).

preset (preset)

Set the xeve preset. Set the encoder preset value to determine encoding speed [fast, medium, slow, placebo]

tune (tune)

Set the encoder tune parameter [psnr, zerolatency]

profile (profile)

Set the encoder profile [0: baseline; 1: main]

crf (crf)

Set the quality for constant quality mode. Constant rate factor <10..49> [default: 32]

qp (qp)

Set constant quantization rate control method parameter. Quantization parameter qp <0..51> [default: 32]

threads (threads)

Force to use a specific number of threads

16.20 libxvid

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.

16.20.1 Options

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.

b
g
qmin
qmax
mpeg_quant
threads
bf
b_qfactor
b_qoffset
flags

Set specific encoding flags. Possible values:

mv4

Use four motion vector by macroblock.

aic

Enable high quality AC prediction.

gray

Only encode grayscale.

qpel

Enable quarter-pixel motion compensation.

cgop

Enable closed GOP.

global_header

Place global headers in extradata instead of every keyframe.

gmc

Enable the use of global motion compensation (GMC). Default is 0 (disabled).

me_quality

Set motion estimation quality level. Possible values in decreasing order of speed and increasing order of quality:

0

Use no motion estimation (default).

1, 2

Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks.

3, 4

Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks and half-pixel refinement for 8x8 blocks, also enable motion estimation on chroma planes for P and B-frames.

5, 6

Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.

mbd

Set macroblock decision algorithm. Possible values in the increasing order of quality:

simple

Use macroblock comparing function algorithm (default).

bits

Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.

rd

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.

lumi_aq

Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).

variance_aq

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.

trellis

Set rate-distortion optimal quantization.

ssim

Set structural similarity (SSIM) displaying method. Possible values:

off

Disable displaying of SSIM information.

avg

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).

frame

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).

ssim_acc

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.

16.21 MediaFoundation

This provides wrappers to encoders (both audio and video) in the MediaFoundation framework. It can access both SW and HW encoders. Video encoders can take input in either of nv12 or yuv420p form (some encoders support both, some support only either - in practice, nv12 is the safer choice, especially among HW encoders).

16.22 Microsoft RLE

Microsoft RLE aka MSRLE encoder. Only 8-bit palette mode supported. Compatible with Windows 3.1 and Windows 95.

16.22.1 Options

g integer

Keyframe interval. A keyframe is inserted at least every -g frames, sometimes sooner.

16.23 mpeg2

MPEG-2 video encoder.

16.23.1 Options

profile

Select the mpeg2 profile to encode:

422
high
ss

Spatially Scalable

snr

SNR Scalable

main
simple
level

Select the mpeg2 level to encode:

high
high1440
main
low
seq_disp_ext integer

Specifies if the encoder should write a sequence_display_extension to the output.

-1
auto

Decide automatically to write it or not (this is the default) by checking if the data to be written is different from the default or unspecified values.

0
never

Never write it.

1
always

Always write it.

video_format integer

Specifies the video_format written into the sequence display extension indicating the source of the video pictures. The default is ‘unspecified’, can be ‘component’, ‘pal’, ‘ntsc’, ‘secam’ or ‘mac’. For maximum compatibility, use ‘component’.

a53cc boolean

Import closed captions (which must be ATSC compatible format) into output. Default is 1 (on).

16.24 png

PNG image encoder.

16.24.1 Private options

dpi integer

Set physical density of pixels, in dots per inch, unset by default

dpm integer

Set physical density of pixels, in dots per meter, unset by default

16.25 ProRes

Apple ProRes encoder.

FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder. The used encoder can be chosen with the -vcodec option.

16.25.1 Private Options for prores-ks

profile integer

Select the ProRes profile to encode

proxy
lt
standard
hq
4444
4444xq
quant_mat integer

Select quantization matrix.

auto
default
proxy
lt
standard
hq

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.

bits_per_mb integer

How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits per macroblock, the maximum is 8000.

mbs_per_slice integer

Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all situations.

vendor string

Override the 4-byte vendor ID. A custom vendor ID like apl0 would claim the stream was produced by the Apple encoder.

alpha_bits integer

Specify number of bits for alpha component. Possible values are 0, 8 and 16. Use 0 to disable alpha plane coding.

16.25.2 Speed considerations

In the default mode of operation the encoder has to honor frame constraints (i.e. not produce 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.

16.26 QSV Encoders

The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG, VP9, AV1)

16.26.1 Ratecontrol Method

The ratecontrol method is selected as follows:

  • When global_quality is specified, a quality-based mode is used. Specifically this means either
    • - CQP - constant quantizer scale, when the qscale codec flag is also set (the -qscale ffmpeg option).
    • - LA_ICQ - intelligent constant quality with lookahead, when the look_ahead option is also set.
    • - ICQ – intelligent constant quality otherwise. For the ICQ modes, global quality range is 1 to 51, with 1 being the best quality.
  • Otherwise when the desired average bitrate is specified with the b option, a bitrate-based mode is used.
    • - LA - VBR with lookahead, when the look_ahead option is specified.
    • - VCM - video conferencing mode, when the vcm option is set.
    • - CBR - constant bitrate, when maxrate is specified and equal to the average bitrate.
    • - VBR - variable bitrate, when maxrate is specified, but is higher than the average bitrate.
    • - AVBR - average VBR mode, when maxrate is not specified, both avbr_accuracy and avbr_convergence are set to non-zero. This mode is available for H264 and HEVC on Windows.
  • Otherwise the default ratecontrol method CQP is used.

Note that depending on your system, a different mode than the one you specified may be selected by the encoder. Set the verbosity level to verbose or higher to see the actual settings used by the QSV runtime.

16.26.2 Global Options -> MSDK Options

Additional libavcodec global options are mapped to MSDK options as follows:

  • g/gop_size -> GopPicSize
  • bf/max_b_frames+1 -> GopRefDist
  • rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB
  • slices -> NumSlice
  • refs -> NumRefFrame
  • b_strategy/b_frame_strategy -> BRefType
  • cgop/CLOSED_GOP codec flag -> GopOptFlag
  • For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset set the difference between QPP and QPI, and QPP and QPB respectively.
  • Setting the coder option to the value vlc will make the H.264 encoder use CAVLC instead of CABAC.

16.26.3 Common Options

Following options are used by all qsv encoders.

async_depth

Specifies how many asynchronous operations an application performs before the application explicitly synchronizes the result. If zero, the value is not specified.

preset

This option itemizes a range of choices from veryfast (best speed) to veryslow (best quality).

veryfast
faster
fast
medium
slow
slower
veryslow
forced_idr

Forcing I frames as IDR frames.

low_power

For encoders set this flag to ON to reduce power consumption and GPU usage.

16.26.4 Runtime Options

Following options can be used durning qsv encoding.

global_quality
i_quant_factor
i_quant_offset
b_quant_factor
b_quant_offset

Supported in h264_qsv and hevc_qsv. Change these value to reset qsv codec’s qp configuration.

max_frame_size

Supported in h264_qsv and hevc_qsv. Change this value to reset qsv codec’s MaxFrameSize configuration.

gop_size

Change this value to reset qsv codec’s gop configuration.

int_ref_type
int_ref_cycle_size
int_ref_qp_delta
int_ref_cycle_dist

Supported in h264_qsv and hevc_qsv. Change these value to reset qsv codec’s Intra Refresh configuration.

qmax
qmin
max_qp_i
min_qp_i
max_qp_p
min_qp_p
max_qp_b
min_qp_b

Supported in h264_qsv. Change these value to reset qsv codec’s max/min qp configuration.

low_delay_brc

Supported in h264_qsv, hevc_qsv and av1_qsv. Change this value to reset qsv codec’s low_delay_brc configuration.

framerate

Change this value to reset qsv codec’s framerate configuration.

bit_rate
rc_buffer_size
rc_initial_buffer_occupancy
rc_max_rate

Change these value to reset qsv codec’s bitrate control configuration.

pic_timing_sei

Supported in h264_qsv and hevc_qsv. Change this value to reset qsv codec’s pic_timing_sei configuration.

qsv_params

Set QSV encoder parameters as a colon-separated list of key-value pairs.

The qsv_params should be formatted as key1=value1:key2=value2:....

These parameters are passed directly to the underlying Intel Quick Sync Video (QSV) encoder using the MFXSetParameter function.

Example:

ffmpeg -i input.mp4 -c:v h264_qsv -qsv_params "CodingOption1=1:CodingOption2=2" output.mp4

This option allows fine-grained control over various encoder-specific settings provided by the QSV encoder.

16.26.5 H264 options

These options are used by h264_qsv

extbrc

Extended bitrate control.

recovery_point_sei

Set this flag to insert the recovery point SEI message at the beginning of every intra refresh cycle.

rdo

Enable rate distortion optimization.

max_frame_size

Maximum encoded frame size in bytes.

max_frame_size_i

Maximum encoded frame size for I frames in bytes. If this value is set as larger than zero, then for I frames the value set by max_frame_size is ignored.

max_frame_size_p

Maximum encoded frame size for P frames in bytes. If this value is set as larger than zero, then for P frames the value set by max_frame_size is ignored.

max_slice_size

Maximum encoded slice size in bytes.

bitrate_limit

Toggle bitrate limitations. Modifies bitrate to be in the range imposed by the QSV encoder. Setting this flag off may lead to violation of HRD conformance. Mind that specifying bitrate below the QSV encoder range might significantly affect quality. If on this option takes effect in non CQP modes: if bitrate is not in the range imposed by the QSV encoder, it will be changed to be in the range.

mbbrc

Setting this flag enables macroblock level bitrate control that generally improves subjective visual quality. Enabling this flag may have negative impact on performance and objective visual quality metric.

low_delay_brc

Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides more accurate bitrate control to minimize the variance of bitstream size frame by frame. Value: -1-default 0-off 1-on

adaptive_i

This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of frame type from P and B to I.

adaptive_b

This flag controls changing of frame type from B to P.

p_strategy

Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0).

b_strategy

This option controls usage of B frames as reference.

dblk_idc

This option disable deblocking. It has value in range 0~2.

cavlc

If set, CAVLC is used; if unset, CABAC is used for encoding.

vcm

Video conferencing mode, please see ratecontrol method.

idr_interval

Distance (in I-frames) between IDR frames.

pic_timing_sei

Insert picture timing SEI with pic_struct_syntax element.

single_sei_nal_unit

Put all the SEI messages into one NALU.

max_dec_frame_buffering

Maximum number of frames buffered in the DPB.

look_ahead

Use VBR algorithm with look ahead.

look_ahead_depth

Depth of look ahead in number frames.

look_ahead_downsampling

Downscaling factor for the frames saved for the lookahead analysis.

unknown
auto
off
2x
4x
int_ref_type

Specifies intra refresh type. The major goal of intra refresh is improvement of error resilience without significant impact on encoded bitstream size caused by I frames. The SDK encoder achieves this by encoding part of each frame in refresh cycle using intra MBs. none means no refresh. vertical means vertical refresh, by column of MBs. horizontal means horizontal refresh, by rows of MBs. slice means horizontal refresh by slices without overlapping. In case of slice, in_ref_cycle_size is ignored. To enable intra refresh, B frame should be set to 0.

int_ref_cycle_size

Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are invalid values.

int_ref_qp_delta

Specifies QP difference for inserted intra MBs. This is signed value in [-51, 51] range if target encoding bit-depth for luma samples is 8 and this range is [-63, 63] for 10 bit-depth or [-75, 75] for 12 bit-depth respectively.

int_ref_cycle_dist

Distance between the beginnings of the intra-refresh cycles in frames.

profile
unknown
baseline
main
high
a53cc

Use A53 Closed Captions (if available).

aud

Insert the Access Unit Delimiter NAL.

mfmode

Multi-Frame Mode.

off
auto
repeat_pps

Repeat pps for every frame.

max_qp_i

Maximum video quantizer scale for I frame.

min_qp_i

Minimum video quantizer scale for I frame.

max_qp_p

Maximum video quantizer scale for P frame.

min_qp_p

Minimum video quantizer scale for P frame.

max_qp_b

Maximum video quantizer scale for B frame.

min_qp_b

Minimum video quantizer scale for B frame.

scenario

Provides a hint to encoder about the scenario for the encoding session.

unknown
displayremoting
videoconference
archive
livestreaming
cameracapture
videosurveillance
gamestreaming
remotegaming
avbr_accuracy

Accuracy of the AVBR ratecontrol (unit of tenth of percent).

avbr_convergence

Convergence of the AVBR ratecontrol (unit of 100 frames)

The parameters avbr_accuracy and avbr_convergence are for the average variable bitrate control (AVBR) algorithm. The algorithm focuses on overall encoding quality while meeting the specified bitrate, target_bitrate, within the accuracy range avbr_accuracy, after a avbr_Convergence period. This method does not follow HRD and the instant bitrate is not capped or padded.

skip_frame

Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option defines the usage of this metadata.

no_skip

Frame skipping is disabled.

insert_dummy

Encoder inserts into bitstream frame where all macroblocks are encoded as skipped.

insert_nothing

Similar to insert_dummy, but encoder inserts nothing into bitstream. The skipped frames are still used in brc. For example, gop still include skipped frames, and the frames after skipped frames will be larger in size.

brc_only

skip_frame metadata indicates the number of missed frames before the current frame.

16.26.6 HEVC Options

These options are used by hevc_qsv

extbrc

Extended bitrate control.

recovery_point_sei

Set this flag to insert the recovery point SEI message at the beginning of every intra refresh cycle.

rdo

Enable rate distortion optimization.

max_frame_size

Maximum encoded frame size in bytes.

max_frame_size_i

Maximum encoded frame size for I frames in bytes. If this value is set as larger than zero, then for I frames the value set by max_frame_size is ignored.

max_frame_size_p

Maximum encoded frame size for P frames in bytes. If this value is set as larger than zero, then for P frames the value set by max_frame_size is ignored.

max_slice_size

Maximum encoded slice size in bytes.

mbbrc

Setting this flag enables macroblock level bitrate control that generally improves subjective visual quality. Enabling this flag may have negative impact on performance and objective visual quality metric.

low_delay_brc

Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides more accurate bitrate control to minimize the variance of bitstream size frame by frame. Value: -1-default 0-off 1-on

adaptive_i

This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of frame type from P and B to I.

adaptive_b

This flag controls changing of frame type from B to P.

p_strategy

Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0).

b_strategy

This option controls usage of B frames as reference.

dblk_idc

This option disable deblocking. It has value in range 0~2.

idr_interval

Distance (in I-frames) between IDR frames.

begin_only

Output an IDR-frame only at the beginning of the stream.

load_plugin

A user plugin to load in an internal session.

none
hevc_sw
hevc_hw
load_plugins

A :-separate list of hexadecimal plugin UIDs to load in an internal session.

look_ahead_depth

Depth of look ahead in number frames, available when extbrc option is enabled.

profile

Set the encoding profile (scc requires libmfx >= 1.32).

unknown
main
main10
mainsp
rext
scc
tier

Set the encoding tier (only level >= 4 can support high tier). This option only takes effect when the level option is specified.

main
high
gpb

1: GPB (generalized P/B frame)

0: regular P frame.

tile_cols

Number of columns for tiled encoding.

tile_rows

Number of rows for tiled encoding.

aud

Insert the Access Unit Delimiter NAL.

pic_timing_sei

Insert picture timing SEI with pic_struct_syntax element.

transform_skip

Turn this option ON to enable transformskip. It is supported on platform equal or newer than ICL.

int_ref_type

Specifies intra refresh type. The major goal of intra refresh is improvement of error resilience without significant impact on encoded bitstream size caused by I frames. The SDK encoder achieves this by encoding part of each frame in refresh cycle using intra MBs. none means no refresh. vertical means vertical refresh, by column of MBs. horizontal means horizontal refresh, by rows of MBs. slice means horizontal refresh by slices without overlapping. In case of slice, in_ref_cycle_size is ignored. To enable intra refresh, B frame should be set to 0.

int_ref_cycle_size

Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are invalid values.

int_ref_qp_delta

Specifies QP difference for inserted intra MBs. This is signed value in [-51, 51] range if target encoding bit-depth for luma samples is 8 and this range is [-63, 63] for 10 bit-depth or [-75, 75] for 12 bit-depth respectively.

int_ref_cycle_dist

Distance between the beginnings of the intra-refresh cycles in frames.

max_qp_i

Maximum video quantizer scale for I frame.

min_qp_i

Minimum video quantizer scale for I frame.

max_qp_p

Maximum video quantizer scale for P frame.

min_qp_p

Minimum video quantizer scale for P frame.

max_qp_b

Maximum video quantizer scale for B frame.

min_qp_b

Minimum video quantizer scale for B frame.

scenario

Provides a hint to encoder about the scenario for the encoding session.

unknown
displayremoting
videoconference
archive
livestreaming
cameracapture
videosurveillance
gamestreaming
remotegaming
avbr_accuracy

Accuracy of the AVBR ratecontrol (unit of tenth of percent).

avbr_convergence

Convergence of the AVBR ratecontrol (unit of 100 frames)

The parameters avbr_accuracy and avbr_convergence are for the average variable bitrate control (AVBR) algorithm. The algorithm focuses on overall encoding quality while meeting the specified bitrate, target_bitrate, within the accuracy range avbr_accuracy, after a avbr_Convergence period. This method does not follow HRD and the instant bitrate is not capped or padded.

skip_frame

Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option defines the usage of this metadata.

no_skip

Frame skipping is disabled.

insert_dummy

Encoder inserts into bitstream frame where all macroblocks are encoded as skipped.

insert_nothing

Similar to insert_dummy, but encoder inserts nothing into bitstream. The skipped frames are still used in brc. For example, gop still include skipped frames, and the frames after skipped frames will be larger in size.

brc_only

skip_frame metadata indicates the number of missed frames before the current frame.

16.26.7 MPEG2 Options

These options are used by mpeg2_qsv

profile
unknown
simple
main
high

16.26.8 VP9 Options

These options are used by vp9_qsv

profile
unknown
profile0
profile1
profile2
profile3
tile_cols

Number of columns for tiled encoding (requires libmfx >= 1.29).

tile_rows

Number of rows for tiled encoding (requires libmfx >= 1.29).

16.26.9 AV1 Options

These options are used by av1_qsv (requires libvpl).

profile
unknown
main
tile_cols

Number of columns for tiled encoding.

tile_rows

Number of rows for tiled encoding.

adaptive_i

This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of frame type from P and B to I.

adaptive_b

This flag controls changing of frame type from B to P.

b_strategy

This option controls usage of B frames as reference.

extbrc

Extended bitrate control.

look_ahead_depth

Depth of look ahead in number frames, available when extbrc option is enabled.

low_delay_brc

Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides more accurate bitrate control to minimize the variance of bitstream size frame by frame. Value: -1-default 0-off 1-on

max_frame_size

Set the allowed max size in bytes for each frame. If the frame size exceeds the limitation, encoder will adjust the QP value to control the frame size. Invalid in CQP rate control mode.

max_frame_size_i

Maximum encoded frame size for I frames in bytes. If this value is set as larger than zero, then for I frames the value set by max_frame_size is ignored.

max_frame_size_p

Maximum encoded frame size for P frames in bytes. If this value is set as larger than zero, then for P frames the value set by max_frame_size is ignored.

16.27 snow

16.27.1 Options

iterative_dia_size

dia size for the iterative motion estimation

16.28 VAAPI encoders

Wrappers for hardware encoders accessible via VAAPI.

These encoders only accept input in VAAPI hardware surfaces. If you have input in software frames, use the hwupload filter to upload them to the GPU.

The following standard libavcodec options are used:

  • g / gop_size
  • bf / max_b_frames
  • profile

    If not set, this will be determined automatically from the format of the input frames and the profiles supported by the driver.

  • level
  • b / bit_rate
  • maxrate / rc_max_rate
  • bufsize / rc_buffer_size
  • rc_init_occupancy / rc_initial_buffer_occupancy
  • compression_level

    Speed / quality tradeoff: higher values are faster / worse quality.

  • q / global_quality

    Size / quality tradeoff: higher values are smaller / worse quality.

  • qmin
  • qmax
  • i_qfactor / i_quant_factor
  • i_qoffset / i_quant_offset
  • b_qfactor / b_quant_factor
  • b_qoffset / b_quant_offset
  • slices

All encoders support the following options:

low_power

Some drivers/platforms offer a second encoder for some codecs intended to use less power than the default encoder; setting this option will attempt to use that encoder. Note that it may support a reduced feature set, so some other options may not be available in this mode.

idr_interval

Set the number of normal intra frames between full-refresh (IDR) frames in open-GOP mode. The intra frames are still IRAPs, but will not include global headers and may have non-decodable leading pictures.

b_depth

Set the B-frame reference depth. When set to one (the default), all B-frames will refer only to P- or I-frames. When set to greater values multiple layers of B-frames will be present, frames in each layer only referring to frames in higher layers.

async_depth

Maximum processing parallelism. Increase this to improve single channel performance. This option doesn’t work if driver doesn’t implement vaSyncBuffer function. Please make sure there are enough hw_frames allocated if a large number of async_depth is used.

max_frame_size

Set the allowed max size in bytes for each frame. If the frame size exceeds the limitation, encoder will adjust the QP value to control the frame size. Invalid in CQP rate control mode.

rc_mode

Set the rate control mode to use. A given driver may only support a subset of modes.

Possible modes:

auto

Choose the mode automatically based on driver support and the other options. This is the default.

CQP

Constant-quality.

CBR

Constant-bitrate.

VBR

Variable-bitrate.

ICQ

Intelligent constant-quality.

QVBR

Quality-defined variable-bitrate.

AVBR

Average variable bitrate.

blbrc

Enable block level rate control, which assigns different bitrate block by block. Invalid for CQP mode.

Each encoder also has its own specific options:

av1_vaapi

profile sets the value of seq_profile. tier sets the value of seq_tier. level sets the value of seq_level_idx.

tiles

Set the number of tiles to encode the input video with, as columns x rows. (default is auto, which means use minimal tile column/row number).

tile_groups

Set tile groups number. All the tiles will be distributed as evenly as possible to each tile group. (default is 1).

h264_vaapi

profile sets the value of profile_idc and the constraint_set*_flags. level sets the value of level_idc.

coder

Set entropy encoder (default is cabac). Possible values:

ac
cabac

Use CABAC.

vlc
cavlc

Use CAVLC.

aud

Include access unit delimiters in the stream (not included by default).

sei

Set SEI message types to include. Some combination of the following values:

identifier

Include a user_data_unregistered message containing information about the encoder.

timing

Include picture timing parameters (buffering_period and pic_timing messages).

recovery_point

Include recovery points where appropriate (recovery_point messages).

hevc_vaapi

profile and level set the values of general_profile_idc and general_level_idc respectively.

aud

Include access unit delimiters in the stream (not included by default).

tier

Set general_tier_flag. This may affect the level chosen for the stream if it is not explicitly specified.

sei

Set SEI message types to include. Some combination of the following values:

hdr

Include HDR metadata if the input frames have it (mastering_display_colour_volume and content_light_level messages).

tiles

Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow greater parallelism in both encoding and decoding, but may decrease coding efficiency.

mjpeg_vaapi

Only baseline DCT encoding is supported. The encoder always uses the standard quantisation and huffman tables - global_quality scales the standard quantisation table (range 1-100).

For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported. RGB is also supported, and will create an RGB JPEG.

jfif

Include JFIF header in each frame (not included by default).

huffman

Include standard huffman tables (on by default). Turning this off will save a few hundred bytes in each output frame, but may lose compatibility with some JPEG decoders which don’t fully handle MJPEG.

mpeg2_vaapi

profile and level set the value of profile_and_level_indication.

vp8_vaapi

B-frames are not supported.

global_quality sets the q_idx used for non-key frames (range 0-127).

loop_filter_level
loop_filter_sharpness

Manually set the loop filter parameters.

vp9_vaapi

global_quality sets the q_idx used for P-frames (range 0-255).

loop_filter_level
loop_filter_sharpness

Manually set the loop filter parameters.

B-frames are supported, but the output stream is always in encode order rather than display order. If B-frames are enabled, it may be necessary to use the vp9_raw_reorder bitstream filter to modify the output stream to display frames in the correct order.

Only normal frames are produced - the vp9_superframe bitstream filter may be required to produce a stream usable with all decoders.

16.29 vbn

Vizrt Binary Image encoder.

This format is used by the broadcast vendor Vizrt for quick texture streaming. Advanced features of the format such as LZW compression of texture data or generation of mipmaps are not supported.

16.29.1 Options

format string

Sets the texture compression used by the VBN file. Can be dxt1, dxt5 or raw. Default is dxt5.

16.30 vc2

SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at professional broadcasting but since it supports yuv420, yuv422 and yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes it suitable for other tasks which require low overhead and low compression (like screen recording).

16.30.1 Options

b

Sets target video bitrate. Usually that’s around 1:6 of the uncompressed video bitrate (e.g. for 1920x1080 50fps yuv422p10 that’s around 400Mbps). Higher values (close to the uncompressed bitrate) turn on lossless compression mode.

field_order

Enables field coding when set (e.g. to tt - top field first) for interlaced inputs. Should increase compression with interlaced content as it splits the fields and encodes each separately.

wavelet_depth

Sets the total amount of wavelet transforms to apply, between 1 and 5 (default). Lower values reduce compression and quality. Less capable decoders may not be able to handle values of wavelet_depth over 3.

wavelet_type

Sets the transform type. Currently only 5_3 (LeGall) and 9_7 (Deslauriers-Dubuc) are implemented, with 9_7 being the one with better compression and thus is the default.

slice_width
slice_height

Sets the slice size for each slice. Larger values result in better compression. For compatibility with other more limited decoders use slice_width of 32 and slice_height of 8.

tolerance

Sets the undershoot tolerance of the rate control system in percent. This is to prevent an expensive search from being run.

qm

Sets the quantization matrix preset to use by default or when wavelet_depth is set to 5

  • - default Uses the default quantization matrix from the specifications, extended with values for the fifth level. This provides a good balance between keeping detail and omitting artifacts.
  • - flat Use a completely zeroed out quantization matrix. This increases PSNR but might reduce perception. Use in bogus benchmarks.
  • - color Reduces detail but attempts to preserve color at extremely low bitrates.

17 Subtitles Encoders

17.1 dvdsub

This codec encodes the bitmap subtitle format that is used in DVDs. Typically they are stored in VOBSUB file pairs (*.idx + *.sub), and they can also be used in Matroska files.

17.1.1 Options

palette

Specify the global palette used by the bitmaps.

The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix) separated by commas, for example 0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b.

even_rows_fix

When set to 1, enable a work-around that makes the number of pixel rows even in all subtitles. This fixes a problem with some players that cut off the bottom row if the number is odd. The work-around just adds a fully transparent row if needed. The overhead is low, typically one byte per subtitle on average.

By default, this work-around is disabled.

18 Bitstream Filters

When you configure your FFmpeg build, all the supported bitstream filters are enabled by default. You can list all available ones using the configure option --list-bsfs.

You can disable all the bitstream filters using the configure option --disable-bsfs, and selectively enable any bitstream filter using the option --enable-bsf=BSF, or you can disable a particular bitstream filter using the option --disable-bsf=BSF.

The option -bsfs of the ff* tools will display the list of all the supported bitstream filters included in your build.

The ff* tools have a -bsf option applied per stream, taking a comma-separated list of filters, whose parameters follow the filter name after a ’=’.

ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT

Below is a description of the currently available bitstream filters, with their parameters, if any.

18.1 aac_adtstoasc

Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration bitstream.

This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS header.

This filter is required for example when copying an AAC stream from a raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or to MOV/MP4 files and related formats such as 3GP or M4A. Please note that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.

18.2 av1_metadata

Modify metadata embedded in an AV1 stream.

td

Insert or remove temporal delimiter OBUs in all temporal units of the stream.

insert

Insert a TD at the beginning of every TU which does not already have one.

remove

Remove the TD from the beginning of every TU which has one.

color_primaries
transfer_characteristics
matrix_coefficients

Set the color description fields in the stream (see AV1 section 6.4.2).

color_range

Set the color range in the stream (see AV1 section 6.4.2; note that this cannot be set for streams using BT.709 primaries, sRGB transfer characteristic and identity (RGB) matrix coefficients).

tv

Limited range.

pc

Full range.

chroma_sample_position

Set the chroma sample location in the stream (see AV1 section 6.4.2). This can only be set for 4:2:0 streams.

vertical

Left position (matching the default in MPEG-2 and H.264).

colocated

Top-left position.

tick_rate

Set the tick rate (time_scale / num_units_in_display_tick) in the timing info in the sequence header.

num_ticks_per_picture

Set the number of ticks in each picture, to indicate that the stream has a fixed framerate. Ignored if tick_rate is not also set.

delete_padding

Deletes Padding OBUs.

18.3 chomp

Remove zero padding at the end of a packet.

18.4 dca_core

Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD.

18.5 dovi_rpu

Manipulate Dolby Vision metadata in a HEVC/AV1 bitstream, optionally enabling metadata compression.

strip

If enabled, strip all Dolby Vision metadata (configuration record + RPU data blocks) from the stream.

compression

Which compression level to enable.

none

No metadata compression.

limited

Limited metadata compression scheme. Should be compatible with most devices. This is the default.

extended

Extended metadata compression. Devices are not required to support this. Note that this level currently behaves the same as ‘limited’ in libavcodec.

18.6 dump_extra

Add extradata to the beginning of the filtered packets except when said packets already exactly begin with the extradata that is intended to be added.

freq

The additional argument specifies which packets should be filtered. It accepts the values:

k
keyframe

add extradata to all key packets

e
all

add extradata to all packets

If not specified it is assumed ‘k’.

For example the following ffmpeg command forces a global header (thus disabling individual packet headers) in the H.264 packets generated by the libx264 encoder, but corrects them by adding the header stored in extradata to the key packets:

ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts

18.7 dv_error_marker

Blocks in DV which are marked as damaged are replaced by blocks of the specified color.

color

The color to replace damaged blocks by

sta

A 16 bit mask which specifies which of the 16 possible error status values are to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0 error status values.

ok

No error, no concealment

err

Error, No concealment

res

Reserved

notok

Error or concealment

notres

Not reserved

Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru

The specific error status code

see page 44-46 or section 5.5 of http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf

18.8 eac3_core

Extract the core from a E-AC-3 stream, dropping extra channels.

18.9 extract_extradata

Extract the in-band extradata.

Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers, or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part of the bitstream containing the coded frames) or "out of band" (e.g. on the container level). This latter form is called "extradata" in FFmpeg terminology.

This bitstream filter detects the in-band headers and makes them available as extradata.

remove

When this option is enabled, the long-term headers are removed from the bitstream after extraction.

18.10 filter_units

Remove units with types in or not in a given set from the stream.

pass_types

List of unit types or ranges of unit types to pass through while removing all others. This is specified as a ’|’-separated list of unit type values or ranges of values with ’-’.

remove_types

Identical to pass_types, except the units in the given set removed and all others passed through.

The types used by pass_types and remove_types correspond to NAL unit types (nal_unit_type) in H.264, HEVC and H.266 (see Table 7-1 in the H.264 and HEVC specifications or Table 5 in the H.266 specification), to marker values for JPEG (without 0xFF prefix) and to start codes without start code prefix (i.e. the byte following the 0x000001) for MPEG-2. For VP8 and VP9, every unit has type zero.

Extradata is unchanged by this transformation, but note that if the stream contains inline parameter sets then the output may be unusable if they are removed.

For example, to remove all non-VCL NAL units from an H.264 stream:

ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT

To remove all AUDs, SEI and filler from an H.265 stream:

ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT

To remove all user data from a MPEG-2 stream, including Closed Captions:

ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=178' OUTPUT

To remove all SEI from a H264 stream, including Closed Captions:

ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=6' OUTPUT

To remove all prefix and suffix SEI from a HEVC stream, including Closed Captions and dynamic HDR:

ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=39|40' OUTPUT

18.11 hapqa_extract

Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.

texture

Specifies the texture to keep.

color
alpha

Convert HAPQA to HAPQ

ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov

Convert HAPQA to HAPAlphaOnly

ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov

18.12 h264_metadata

Modify metadata embedded in an H.264 stream.

aud

Insert or remove AUD NAL units in all access units of the stream.

pass
insert
remove

Default is pass.

sample_aspect_ratio

Set the sample aspect ratio of the stream in the VUI parameters. See H.264 table E-1.

overscan_appropriate_flag

Set whether the stream is suitable for display using overscan or not (see H.264 section E.2.1).

video_format
video_full_range_flag

Set the video format in the stream (see H.264 section E.2.1 and table E-2).

colour_primaries
transfer_characteristics
matrix_coefficients

Set the colour description in the stream (see H.264 section E.2.1 and tables E-3, E-4 and E-5).

chroma_sample_loc_type

Set the chroma sample location in the stream (see H.264 section E.2.1 and figure E-1).

tick_rate

Set the tick rate (time_scale / num_units_in_tick) in the VUI parameters. This is the smallest time unit representable in the stream, and in many cases represents the field rate of the stream (double the frame rate).

fixed_frame_rate_flag

Set whether the stream has fixed framerate - typically this indicates that the framerate is exactly half the tick rate, but the exact meaning is dependent on interlacing and the picture structure (see H.264 section E.2.1 and table E-6).

zero_new_constraint_set_flags

Zero constraint_set4_flag and constraint_set5_flag in the SPS. These bits were reserved in a previous version of the H.264 spec, and thus some hardware decoders require these to be zero. The result of zeroing this is still a valid bitstream.

crop_left
crop_right
crop_top
crop_bottom

Set the frame cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped.

These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled or the stream is interlaced (see H.264 section 7.4.2.1.1).

sei_user_data

Insert a string as SEI unregistered user data. The argument must be of the form UUID+string, where the UUID is as hex digits possibly separated by hyphens, and the string can be anything.

For example, ‘086f3693-b7b3-4f2c-9653-21492feee5b8+hello’ will insert the string “hello” associated with the given UUID.

delete_filler

Deletes both filler NAL units and filler SEI messages.

display_orientation

Insert, extract or remove Display orientation SEI messages. See H.264 section D.1.27 and D.2.27 for syntax and semantics.

pass
insert
remove
extract

Default is pass.

Insert mode works in conjunction with rotate and flip options. Any pre-existing Display orientation messages will be removed in insert or remove mode. Extract mode attaches the display matrix to the packet as side data.

rotate

Set rotation in display orientation SEI (anticlockwise angle in degrees). Range is -360 to +360. Default is NaN.

flip

Set flip in display orientation SEI.

horizontal
vertical

Default is unset.

level

Set the level in the SPS. Refer to H.264 section A.3 and tables A-1 to A-5.

The argument must be the name of a level (for example, ‘4.2’), a level_idc value (for example, ‘42’), or the special name ‘auto’ indicating that the filter should attempt to guess the level from the input stream properties.

18.13 h264_mp4toannexb

Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer mpegts).

For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg, you can use the command:

ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts

Please note that this filter is auto-inserted for MPEG-TS (muxer mpegts) and raw H.264 (muxer h264) output formats.

18.14 h264_redundant_pps

This applies a specific fixup to some Blu-ray streams which contain redundant PPSs modifying irrelevant parameters of the stream which confuse other transformations which require correct extradata.

18.15 hevc_metadata

Modify metadata embedded in an HEVC stream.

aud

Insert or remove AUD NAL units in all access units of the stream.

insert
remove
sample_aspect_ratio

Set the sample aspect ratio in the stream in the VUI parameters.

video_format
video_full_range_flag

Set the video format in the stream (see H.265 section E.3.1 and table E.2).

colour_primaries
transfer_characteristics
matrix_coefficients

Set the colour description in the stream (see H.265 section E.3.1 and tables E.3, E.4 and E.5).

chroma_sample_loc_type

Set the chroma sample location in the stream (see H.265 section E.3.1 and figure E.1).

tick_rate

Set the tick rate in the VPS and VUI parameters (time_scale / num_units_in_tick). Combined with num_ticks_poc_diff_one, this can set a constant framerate in the stream. Note that it is likely to be overridden by container parameters when the stream is in a container.

num_ticks_poc_diff_one

Set poc_proportional_to_timing_flag in VPS and VUI and use this value to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and E.3.1). Ignored if tick_rate is not also set.

crop_left
crop_right
crop_top
crop_bottom

Set the conformance window cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped.

These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled (H.265 section 7.4.3.2.1).

width
height

Set width and height after crop.

level

Set the level in the VPS and SPS. See H.265 section A.4 and tables A.6 and A.7.

The argument must be the name of a level (for example, ‘5.1’), a general_level_idc value (for example, ‘153’ for level 5.1), or the special name ‘auto’ indicating that the filter should attempt to guess the level from the input stream properties.

18.16 hevc_mp4toannexb

Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.265 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer mpegts).

For example to remux an MP4 file containing an HEVC stream to mpegts format with ffmpeg, you can use the command:

ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts

Please note that this filter is auto-inserted for MPEG-TS (muxer mpegts) and raw HEVC/H.265 (muxer h265 or hevc) output formats.

18.17 imxdump

Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This filter only applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7 and newer with the appropriate -tag:v.

For example, to remux 30 MB/sec NTSC IMX to MOV:

ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov

18.18 mjpeg2jpeg

Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by

ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg

Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml:

Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed – and *omitted* – Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won’t have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec."

This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images.

ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

18.19 mjpegadump

Add an MJPEG A header to the bitstream, to enable decoding by Quicktime.

18.20 mov2textsub

Extract a representable text file from MOV subtitles, stripping the metadata header from each subtitle packet.

See also the text2movsub filter.

18.21 mpeg2_metadata

Modify metadata embedded in an MPEG-2 stream.

display_aspect_ratio

Set the display aspect ratio in the stream.

The following fixed values are supported:

4/3
16/9
221/100

Any other value will result in square pixels being signalled instead (see H.262 section 6.3.3 and table 6-3).

frame_rate

Set the frame rate in the stream. This is constructed from a table of known values combined with a small multiplier and divisor - if the supplied value is not exactly representable, the nearest representable value will be used instead (see H.262 section 6.3.3 and table 6-4).

video_format

Set the video format in the stream (see H.262 section 6.3.6 and table 6-6).

colour_primaries
transfer_characteristics
matrix_coefficients

Set the colour description in the stream (see H.262 section 6.3.6 and tables 6-7, 6-8 and 6-9).

18.22 mpeg4_unpack_bframes

Unpack DivX-style packed B-frames.

DivX-style packed B-frames are not valid MPEG-4 and were only a workaround for the broken Video for Windows subsystem. They use more space, can cause minor AV sync issues, require more CPU power to decode (unless the player has some decoded picture queue to compensate the 2,0,2,0 frame per packet style) and cause trouble if copied into a standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may not be able to decode them, since they are not valid MPEG-4.

For example to fix an AVI file containing an MPEG-4 stream with DivX-style packed B-frames using ffmpeg, you can use the command:

ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi

18.23 noise

Damages the contents of packets or simply drops them without damaging the container. Can be used for fuzzing or testing error resilience/concealment.

Parameters:

amount

Accepts an expression whose evaluation per-packet determines how often bytes in that packet will be modified. A value below 0 will result in a variable frequency. Default is 0 which results in no modification. However, if neither amount nor drop is specified, amount will be set to -1. See below for accepted variables.

drop

Accepts an expression evaluated per-packet whose value determines whether that packet is dropped. Evaluation to a positive value results in the packet being dropped. Evaluation to a negative value results in a variable chance of it being dropped, roughly inverse in proportion to the magnitude of the value. Default is 0 which results in no drops. See below for accepted variables.

dropamount

Accepts a non-negative integer, which assigns a variable chance of it being dropped, roughly inverse in proportion to the value. Default is 0 which results in no drops. This option is kept for backwards compatibility and is equivalent to setting drop to a negative value with the same magnitude i.e. dropamount=4 is the same as drop=-4. Ignored if drop is also specified.

Both amount and drop accept expressions containing the following variables:

n

The index of the packet, starting from zero.

tb

The timebase for packet timestamps.

pts

Packet presentation timestamp.

dts

Packet decoding timestamp.

nopts

Constant representing AV_NOPTS_VALUE.

startpts

First non-AV_NOPTS_VALUE PTS seen in the stream.

startdts

First non-AV_NOPTS_VALUE DTS seen in the stream.

duration
d

Packet duration, in timebase units.

pos

Packet position in input; may be -1 when unknown or not set.

size

Packet size, in bytes.

key

Whether packet is marked as a keyframe.

state

A pseudo random integer, primarily derived from the content of packet payload.

18.23.1 Examples

Apply modification to every byte but don’t drop any packets.

ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv

Drop every video packet not marked as a keyframe after timestamp 30s but do not modify any of the remaining packets.

ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv

Drop one second of audio every 10 seconds and add some random noise to the rest.

ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv

18.24 null

This bitstream filter passes the packets through unchanged.

18.25 pcm_rechunk

Repacketize PCM audio to a fixed number of samples per packet or a fixed packet rate per second. This is similar to the (ffmpeg-filters)asetnsamples audio filter but works on audio packets instead of audio frames.

nb_out_samples, n

Set the number of samples per each output audio packet. The number is intended as the number of samples per each channel. Default value is 1024.

pad, p

If set to 1, the filter will pad the last audio packet with silence, so that it will contain the same number of samples (or roughly the same number of samples, see frame_rate) as the previous ones. Default value is 1.

frame_rate, r

This option makes the filter output a fixed number of packets per second instead of a fixed number of samples per packet. If the audio sample rate is not divisible by the frame rate then the number of samples will not be constant but will vary slightly so that each packet will start as close to the frame boundary as possible. Using this option has precedence over nb_out_samples.

You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio for NTSC frame rate using the frame_rate option.

ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -

18.26 pgs_frame_merge

Merge a sequence of PGS Subtitle segments ending with an "end of display set" segment into a single packet.

This is required by some containers that support PGS subtitles (muxer matroska).

18.27 prores_metadata

Modify color property metadata embedded in prores stream.

color_primaries

Set the color primaries. Available values are:

auto

Keep the same color primaries property (default).

unknown
bt709
bt470bg

BT601 625

smpte170m

BT601 525

bt2020
smpte431

DCI P3

smpte432

P3 D65

transfer_characteristics

Set the color transfer. Available values are:

auto

Keep the same transfer characteristics property (default).

unknown
bt709

BT 601, BT 709, BT 2020

smpte2084

SMPTE ST 2084

arib-std-b67

ARIB STD-B67

matrix_coefficients

Set the matrix coefficient. Available values are:

auto

Keep the same colorspace property (default).

unknown
bt709
smpte170m

BT 601

bt2020nc

Set Rec709 colorspace for each frame of the file

ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov

Set Hybrid Log-Gamma parameters for each frame of the file

ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov

18.28 remove_extra

Remove extradata from packets.

It accepts the following parameter:

freq

Set which frame types to remove extradata from.

k

Remove extradata from non-keyframes only.

keyframe

Remove extradata from keyframes only.

e, all

Remove extradata from all frames.

18.29 setts

Set PTS and DTS in packets.

It accepts the following parameters:

ts
pts
dts

Set expressions for PTS, DTS or both.

duration

Set expression for duration.

time_base

Set output time base.

The expressions are evaluated through the eval API and can contain the following constants:

N

The count of the input packet. Starting from 0.

TS

The demux timestamp in input in case of ts or dts option or presentation timestamp in case of pts option.

POS

The original position in the file of the packet, or undefined if undefined for the current packet

DTS

The demux timestamp in input.

PTS

The presentation timestamp in input.

DURATION

The duration in input.

STARTDTS

The DTS of the first packet.

STARTPTS

The PTS of the first packet.

PREV_INDTS

The previous input DTS.

PREV_INPTS

The previous input PTS.

PREV_INDURATION

The previous input duration.

PREV_OUTDTS

The previous output DTS.

PREV_OUTPTS

The previous output PTS.

PREV_OUTDURATION

The previous output duration.

NEXT_DTS

The next input DTS.

NEXT_PTS

The next input PTS.

NEXT_DURATION

The next input duration.

TB

The timebase of stream packet belongs.

TB_OUT

The output timebase.

SR

The sample rate of stream packet belongs.

NOPTS

The AV_NOPTS_VALUE constant.

For example, to set PTS equal to DTS (not recommended if B-frames are involved):

ffmpeg -i INPUT -c:a copy -bsf:a setts=pts=DTS out.mkv

18.30 showinfo

Log basic packet information. Mainly useful for testing, debugging, and development.

18.31 text2movsub

Convert text subtitles to MOV subtitles (as used by the mov_text codec) with metadata headers.

See also the mov2textsub filter.

18.32 trace_headers

Log trace output containing all syntax elements in the coded stream headers (everything above the level of individual coded blocks). This can be useful for debugging low-level stream issues.

Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending on the build only a subset of these may be available.

18.33 truehd_core

Extract the core from a TrueHD stream, dropping ATMOS data.

18.34 vp9_metadata

Modify metadata embedded in a VP9 stream.

color_space

Set the color space value in the frame header. Note that any frame set to RGB will be implicitly set to PC range and that RGB is incompatible with profiles 0 and 2.

unknown
bt601
bt709
smpte170
smpte240
bt2020
rgb
color_range

Set the color range value in the frame header. Note that any value imposed by the color space will take precedence over this value.

tv
pc

18.35 vp9_superframe

Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This fixes merging of split/segmented VP9 streams where the alt-ref frame was split from its visible counterpart.

18.36 vp9_superframe_split

Split VP9 superframes into single frames.

18.37 vp9_raw_reorder

Given a VP9 stream with correct timestamps but possibly out of order, insert additional show-existing-frame packets to correct the ordering.

19 Format Options

The libavformat library provides some generic global options, which can be set on all the muxers and demuxers. In addition each muxer or demuxer may support so-called private options, which are specific for that component.

Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in the AVFormatContext options or using the libavutil/opt.h API for programmatic use.

The list of supported options follows:

avioflags flags (input/output)

Possible values:

direct

Reduce buffering.

probesize integer (input)

Set probing size in bytes, i.e. the size of the data to analyze to get stream information. A higher value will enable detecting more information in case it is dispersed into the stream, but will increase latency. Must be an integer not lesser than 32. It is 5000000 by default.

max_probe_packets integer (input)

Set the maximum number of buffered packets when probing a codec. Default is 2500 packets.

packetsize integer (output)

Set packet size.

fflags flags

Set format flags. Some are implemented for a limited number of formats.

Possible values for input files:

discardcorrupt

Discard corrupted packets.

fastseek

Enable fast, but inaccurate seeks for some formats.

genpts

Generate missing PTS if DTS is present.

igndts

Ignore DTS if PTS is also set. In case the PTS is set, the DTS value is set to NOPTS. This is ignored when the nofillin flag is set.

ignidx

Ignore index.

nobuffer

Reduce the latency introduced by buffering during initial input streams analysis.

nofillin

Do not fill in missing values in packet fields that can be exactly calculated.

noparse

Disable AVParsers, this needs +nofillin too.

sortdts

Try to interleave output packets by DTS. At present, available only for AVIs with an index.

Possible values for output files:

autobsf

Automatically apply bitstream filters as required by the output format. Enabled by default.

bitexact

Only write platform-, build- and time-independent data. This ensures that file and data checksums are reproducible and match between platforms. Its primary use is for regression testing.

flush_packets

Write out packets immediately.

shortest

Stop muxing at the end of the shortest stream. It may be needed to increase max_interleave_delta to avoid flushing the longer streams before EOF.

seek2any integer (input)

Allow seeking to non-keyframes on demuxer level when supported if set to 1. Default is 0.

analyzeduration integer (input)

Specify how many microseconds are analyzed to probe the input. A higher value will enable detecting more accurate information, but will increase latency. It defaults to 5,000,000 microseconds = 5 seconds.

cryptokey hexadecimal string (input)

Set decryption key.

indexmem integer (input)

Set max memory used for timestamp index (per stream).

rtbufsize integer (input)

Set max memory used for buffering real-time frames.

fdebug flags (input/output)

Print specific debug info.

Possible values:

ts
max_delay integer (input/output)

Set maximum muxing or demuxing delay in microseconds.

fpsprobesize integer (input)

Set number of frames used to probe fps.

audio_preload integer (output)

Set microseconds by which audio packets should be interleaved earlier.

chunk_duration integer (output)

Set microseconds for each chunk.

chunk_size integer (output)

Set size in bytes for each chunk.

err_detect, f_err_detect flags (input)

Set error detection flags. f_err_detect is deprecated and should be used only via the ffmpeg tool.

Possible values:

crccheck

Verify embedded CRCs.

bitstream

Detect bitstream specification deviations.

buffer

Detect improper bitstream length.

explode

Abort decoding on minor error detection.

careful

Consider things that violate the spec and have not been seen in the wild as errors.

compliant

Consider all spec non compliancies as errors.

aggressive

Consider things that a sane encoder should not do as an error.

max_interleave_delta integer (output)

Set maximum buffering duration for interleaving. The duration is expressed in microseconds, and defaults to 10000000 (10 seconds).

To ensure all the streams are interleaved correctly, libavformat will wait until it has at least one packet for each stream before actually writing any packets to the output file. When some streams are "sparse" (i.e. there are large gaps between successive packets), this can result in excessive buffering.

This field specifies the maximum difference between the timestamps of the first and the last packet in the muxing queue, above which libavformat will output a packet regardless of whether it has queued a packet for all the streams.

If set to 0, libavformat will continue buffering packets until it has a packet for each stream, regardless of the maximum timestamp difference between the buffered packets.

use_wallclock_as_timestamps integer (input)

Use wallclock as timestamps if set to 1. Default is 0.

avoid_negative_ts integer (output)

Possible values:

make_non_negative

Shift timestamps to make them non-negative. Also note that this affects only leading negative timestamps, and not non-monotonic negative timestamps.

make_zero

Shift timestamps so that the first timestamp is 0.

auto (default)

Enables shifting when required by the target format.

disabled

Disables shifting of timestamp.

When shifting is enabled, all output timestamps are shifted by the same amount. Audio, video, and subtitles desynching and relative timestamp differences are preserved compared to how they would have been without shifting.

skip_initial_bytes integer (input)

Set number of bytes to skip before reading header and frames if set to 1. Default is 0.

correct_ts_overflow integer (input)

Correct single timestamp overflows if set to 1. Default is 1.

flush_packets integer (output)

Flush the underlying I/O stream after each packet. Default is -1 (auto), which means that the underlying protocol will decide, 1 enables it, and has the effect of reducing the latency, 0 disables it and may increase IO throughput in some cases.

output_ts_offset offset (output)

Set the output time offset.

offset must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.

The offset is added by the muxer to the output timestamps.

Specifying a positive offset means that the corresponding streams are delayed bt the time duration specified in offset. Default value is 0 (meaning that no offset is applied).

format_whitelist list (input)

"," separated list of allowed demuxers. By default all are allowed.

dump_separator string (input)

Separator used to separate the fields printed on the command line about the Stream parameters. For example, to separate the fields with newlines and indentation:

ffprobe -dump_separator "
                          "  -i ~/videos/matrixbench_mpeg2.mpg
max_streams integer (input)

Specifies the maximum number of streams. This can be used to reject files that would require too many resources due to a large number of streams.

skip_estimate_duration_from_pts bool (input)

Skip estimation of input duration if it requires an additional probing for PTS at end of file. At present, applicable for MPEG-PS and MPEG-TS.

duration_probesize integer (input)

Set probing size, in bytes, for input duration estimation when it actually requires an additional probing for PTS at end of file (at present: MPEG-PS and MPEG-TS). It is aimed at users interested in better durations probing for itself, or indirectly because using the concat demuxer, for example. The typical use case is an MPEG-TS CBR with a high bitrate, high video buffering and ending cleaning with similar PTS for video and audio: in such a scenario, the large physical gap between the last video packet and the last audio packet makes it necessary to read many bytes in order to get the video stream duration. Another use case is where the default probing behaviour only reaches a single video frame which is not the last one of the stream due to frame reordering, so the duration is not accurate. Setting this option has a performance impact even for small files because the probing size is fixed. Default behaviour is a general purpose trade-off, largely adaptive, but the probing size will not be extended to get streams durations at all costs. Must be an integer not lesser than 1, or 0 for default behaviour.

strict, f_strict integer (input/output)

Specify how strictly to follow the standards. f_strict is deprecated and should be used only via the ffmpeg tool.

Possible values:

very

strictly conform to an older more strict version of the spec or reference software

strict

strictly conform to all the things in the spec no matter what consequences

normal
unofficial

allow unofficial extensions

experimental

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.

19.1 Format stream specifiers

Format stream specifiers allow selection of one or more streams that match specific properties.

The exact semantics of stream specifiers is defined by the avformat_match_stream_specifier() function declared in the libavformat/avformat.h header and documented in the (ffmpeg)Stream specifiers section in the ffmpeg(1) manual.

20 Demuxers

Demuxers are configured elements in FFmpeg that can read the multimedia streams from a particular type of file.

When you configure your FFmpeg build, all the supported demuxers are enabled by default. You can list all available ones using the configure option --list-demuxers.

You can disable all the demuxers using the configure option --disable-demuxers, and selectively enable a single demuxer with the option --enable-demuxer=DEMUXER, or disable it with the option --disable-demuxer=DEMUXER.

The option -demuxers of the ff* tools will display the list of enabled demuxers. Use -formats to view a combined list of enabled demuxers and muxers.

The description of some of the currently available demuxers follows.

20.1 aa

Audible Format 2, 3, and 4 demuxer.

This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.

20.2 aac

Raw Audio Data Transport Stream AAC demuxer.

This demuxer is used to demux an ADTS input containing a single AAC stream alongwith any ID3v1/2 or APE tags in it.

20.3 apng

Animated Portable Network Graphics demuxer.

This demuxer is used to demux APNG files. All headers, but the PNG signature, up to (but not including) the first fcTL chunk are transmitted as extradata. Frames are then split as being all the chunks between two fcTL ones, or between the last fcTL and IEND chunks.

-ignore_loop bool

Ignore the loop variable in the file if set. Default is enabled.

-max_fps int

Maximum framerate in frames per second. Default of 0 imposes no limit.

-default_fps int

Default framerate in frames per second when none is specified in the file (0 meaning as fast as possible). Default is 15.

20.4 asf

Advanced Systems Format demuxer.

This demuxer is used to demux ASF files and MMS network streams.

-no_resync_search bool

Do not try to resynchronize by looking for a certain optional start code.

20.5 concat

Virtual concatenation script demuxer.

This demuxer reads a list of files and other directives from a text file and demuxes them one after the other, as if all their packets had been muxed together.

The timestamps in the files are adjusted so that the first file starts at 0 and each next file starts where the previous one finishes. Note that it is done globally and may cause gaps if all streams do not have exactly the same length.

All files must have the same streams (same codecs, same time base, etc.).

The duration of each file is used to adjust the timestamps of the next file: if the duration is incorrect (because it was computed using the bit-rate or because the file is truncated, for example), it can cause artifacts. The duration directive can be used to override the duration stored in each file.

20.5.1 Syntax

The script is a text file in extended-ASCII, with one directive per line. Empty lines, leading spaces and lines starting with ’#’ are ignored. The following directive is recognized:

file path

Path to a file to read; special characters and spaces must be escaped with backslash or single quotes.

All subsequent file-related directives apply to that file.

ffconcat version 1.0

Identify the script type and version.

To make FFmpeg recognize the format automatically, this directive must appear exactly as is (no extra space or byte-order-mark) on the very first line of the script.

duration dur

Duration of the file. This information can be specified from the file; specifying it here may be more efficient or help if the information from the file is not available or accurate.

If the duration is set for all files, then it is possible to seek in the whole concatenated video.

inpoint timestamp

In point of the file. When the demuxer opens the file it instantly seeks to the specified timestamp. Seeking is done so that all streams can be presented successfully at In point.

This directive works best with intra frame codecs, because for non-intra frame ones you will usually get extra packets before the actual In point and the decoded content will most likely contain frames before In point too.

For each file, packets before the file In point will have timestamps less than the calculated start timestamp of the file (negative in case of the first file), and the duration of the files (if not specified by the duration directive) will be reduced based on their specified In point.

Because of potential packets before the specified In point, packet timestamps may overlap between two concatenated files.

outpoint timestamp

Out point of the file. When the demuxer reaches the specified decoding timestamp in any of the streams, it handles it as an end of file condition and skips the current and all the remaining packets from all streams.

Out point is exclusive, which means that the demuxer will not output packets with a decoding timestamp greater or equal to Out point.

This directive works best with intra frame codecs and formats where all streams are tightly interleaved. For non-intra frame codecs you will usually get additional packets with presentation timestamp after Out point therefore the decoded content will most likely contain frames after Out point too. If your streams are not tightly interleaved you may not get all the packets from all streams before Out point and you may only will be able to decode the earliest stream until Out point.

The duration of the files (if not specified by the duration directive) will be reduced based on their specified Out point.

file_packet_metadata key=value

Metadata of the packets of the file. The specified metadata will be set for each file packet. You can specify this directive multiple times to add multiple metadata entries. This directive is deprecated, use file_packet_meta instead.

file_packet_meta key value

Metadata of the packets of the file. The specified metadata will be set for each file packet. You can specify this directive multiple times to add multiple metadata entries.

option key value

Option to access, open and probe the file. Can be present multiple times.

stream

Introduce a stream in the virtual file. All subsequent stream-related directives apply to the last introduced stream. Some streams properties must be set in order to allow identifying the matching streams in the subfiles. If no streams are defined in the script, the streams from the first file are copied.

exact_stream_id id

Set the id of the stream. If this directive is given, the string with the corresponding id in the subfiles will be used. This is especially useful for MPEG-PS (VOB) files, where the order of the streams is not reliable.

stream_meta key value

Metadata for the stream. Can be present multiple times.

stream_codec value

Codec for the stream.

stream_extradata hex_string

Extradata for the string, encoded in hexadecimal.

chapter id start end

Add a chapter. id is an unique identifier, possibly small and consecutive.

20.5.2 Options

This demuxer accepts the following option:

safe

If set to 1, reject unsafe file paths and directives. A file path is considered safe if it does not contain a protocol specification and is relative and all components only contain characters from the portable character set (letters, digits, period, underscore and hyphen) and have no period at the beginning of a component.

If set to 0, any file name is accepted.

The default is 1.

auto_convert

If set to 1, try to perform automatic conversions on packet data to make the streams concatenable. The default is 1.

Currently, the only conversion is adding the h264_mp4toannexb bitstream filter to H.264 streams in MP4 format. This is necessary in particular if there are resolution changes.

segment_time_metadata

If set to 1, every packet will contain the lavf.concat.start_time and the lavf.concat.duration packet metadata values which are the start_time and the duration of the respective file segments in the concatenated output expressed in microseconds. The duration metadata is only set if it is known based on the concat file. The default is 0.

20.5.3 Examples

  • Use absolute filenames and include some comments:
    # my first filename
    file /mnt/share/file-1.wav
    # my second filename including whitespace
    file '/mnt/share/file 2.wav'
    # my third filename including whitespace plus single quote
    file '/mnt/share/file 3'\''.wav'
    
  • Allow for input format auto-probing, use safe filenames and set the duration of the first file:
    ffconcat version 1.0
    
    file file-1.wav
    duration 20.0
    
    file subdir/file-2.wav
    

20.6 dash

Dynamic Adaptive Streaming over HTTP demuxer.

This demuxer presents all AVStreams found in the manifest. By setting the discard flags on AVStreams the caller can decide which streams to actually receive. Each stream mirrors the id and bandwidth properties from the <Representation> as metadata keys named "id" and "variant_bitrate" respectively.

20.6.1 Options

This demuxer accepts the following option:

cenc_decryption_key

16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

20.7 dvdvideo

DVD-Video demuxer, powered by libdvdnav and libdvdread.

Can directly ingest DVD titles, specifically sequential PGCs, into a conversion pipeline. Menu assets, such as background video or audio, can also be demuxed given the menu’s coordinates (at best effort). Seeking is not supported at this time.

Block devices (DVD drives), ISO files, and directory structures are accepted. Activate with -f dvdvideo in front of one of these inputs.

This demuxer does NOT have decryption code of any kind. You are on your own working with encrypted DVDs, and should not expect support on the matter.

Underlying playback is handled by libdvdnav, and structure parsing by libdvdread. FFmpeg must be built with GPL library support available as well as the configure switches --enable-libdvdnav and --enable-libdvdread.

You will need to provide either the desired "title number" or exact PGC/PG coordinates. Many open-source DVD players and tools can aid in providing this information. If not specified, the demuxer will default to title 1 which works for many discs. However, due to the flexibility of the format, it is recommended to check manually. There are many discs that are authored strangely or with invalid headers.

If the input is a real DVD drive, please note that there are some drives which may silently fail on reading bad sectors from the disc, returning random bits instead which is effectively corrupt data. This is especially prominent on aging or rotting discs. A second pass and integrity checks would be needed to detect the corruption. This is not an FFmpeg issue.

20.7.1 Background

DVD-Video is not a directly accessible, linear container format in the traditional sense. Instead, it allows for complex and programmatic playback of carefully muxed MPEG-PS streams that are stored in headerless VOB files. To the end-user, these streams are known simply as "titles", but the actual logical playback sequence is defined by one or more "PGCs", or Program Group Chains, within the title. The PGC is in turn comprised of multiple "PGs", or Programs", which are the actual video segments (and for a typical video feature, sequentially ordered). The PGC structure, along with stream layout and metadata, are stored in IFO files that need to be parsed. PGCs can be thought of as playlists in easier terms.

An actual DVD player relies on user GUI interaction via menus and an internal VM to drive the direction of demuxing. Generally, the user would either navigate (via menus) or automatically be redirected to the PGC of their choice. During this process and the subsequent playback, the DVD player’s internal VM also maintains a state and executes instructions that can create jumps to different sectors during playback. This is why libdvdnav is involved, as a linear read of the MPEG-PS blobs on the disc (VOBs) is not enough to produce the right sequence in many cases.

There are many other DVD structures (a long subject) that will not be discussed here. NAV packets, in particular, are handled by this demuxer to build accurate timing but not emitted as a stream. For a good high-level understanding, refer to: https://code.videolan.org/videolan/libdvdnav/-/blob/master/doc/dvd_structures

20.7.2 Options

This demuxer accepts the following options:

title int

The title number to play. Must be set if pgc and pg are not set. Not applicable to menus. Default is 0 (auto), which currently only selects the first available title (title 1) and notifies the user about the implications.

chapter_start int

The chapter, or PTT (part-of-title), number to start at. Not applicable to menus. Default is 1.

chapter_end int

The chapter, or PTT (part-of-title), number to end at. Not applicable to menus. Default is 0, which is a special value to signal end at the last possible chapter.

angle int

The video angle number, referring to what is essentially an additional video stream that is composed from alternate frames interleaved in the VOBs. Not applicable to menus. Default is 1.

region int

The region code to use for playback. Some discs may use this to default playback at a particular angle in different regions. This option will not affect the region code of a real DVD drive, if used as an input. Not applicable to menus. Default is 0, "world".

menu bool

Demux menu assets instead of navigating a title. Requires exact coordinates of the menu (menu_lu, menu_vts, pgc, pg). Default is false.

menu_lu int

The menu language to demux. In DVD, menus are grouped by language. Default is 0, the first language unit.

menu_vts int

The VTS where the menu lives, or 0 if it is a VMG menu (root-level). Default is 0, VMG menu.

pgc int

The entry PGC to start playback, in conjunction with pg. Alternative to setting title. Chapter markers are not supported at this time. Must be explicitly set for menus. Default is 0, automatically resolve from value of title.

pg int

The entry PG to start playback, in conjunction with pgc. Alternative to setting title. Chapter markers are not supported at this time. Default is 0, automatically resolve from value of title, or start from the beginning (PG 1) of the menu.

preindex bool

Enable this to have accurate chapter (PTT) markers and duration measurement, which requires a slow second pass read in order to index the chapter marker timestamps from NAV packets. This is non-ideal extra work for real optical drives. It is recommended and faster to use this option with a backup of the DVD structure stored on a hard drive. Not compatible with pgc and pg. Not applicable to menus. Default is 0, false.

trim bool

Skip padding cells (i.e. cells shorter than 1 second) from the beginning. There exist many discs with filler segments at the beginning of the PGC, often with junk data intended for controlling a real DVD player’s buffering speed and with no other material data value. Not applicable to menus. Default is 1, true.

20.7.3 Examples

  • Open title 3 from a given DVD structure:
    ffmpeg -f dvdvideo -title 3 -i <path to DVD> ...
    
  • Open chapters 3-6 from title 1 from a given DVD structure:
    ffmpeg -f dvdvideo -chapter_start 3 -chapter_end 6 -title 1 -i <path to DVD> ...
    
  • Open only chapter 5 from title 1 from a given DVD structure:
    ffmpeg -f dvdvideo -chapter_start 5 -chapter_end 5 -title 1 -i <path to DVD> ...
    
  • Demux menu with language 1 from VTS 1, PGC 1, starting at PG 1:
    ffmpeg -f dvdvideo -menu 1 -menu_lu 1 -menu_vts 1 -pgc 1 -pg 1 -i <path to DVD> ...
    

20.8 ea

Electronic Arts Multimedia format demuxer.

This format is used by various Electronic Arts games.

20.8.1 Options

merge_alpha bool

Normally the VP6 alpha channel (if exists) is returned as a secondary video stream, by setting this option you can make the demuxer return a single video stream which contains the alpha channel in addition to the ordinary video.

20.9 imf

Interoperable Master Format demuxer.

This demuxer presents audio and video streams found in an IMF Composition, as specified in SMPTE ST 2067-2.

ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...

If -assetmaps is not specified, the demuxer looks for a file called ASSETMAP.xml in the same directory as the CPL.

20.10 flv, live_flv, kux

Adobe Flash Video Format demuxer.

This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities. KUX is a flv variant used on the Youku platform.

ffmpeg -f flv -i myfile.flv ...
ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
-flv_metadata bool

Allocate the streams according to the onMetaData array content.

-flv_ignore_prevtag bool

Ignore the size of previous tag value.

-flv_full_metadata bool

Output all context of the onMetadata.

20.11 gif

Animated GIF demuxer.

It accepts the following options:

min_delay

Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 2.

max_gif_delay

Set the maximum valid delay between frames in hundredth of seconds. Range is 0 to 65535. Default value is 65535 (nearly eleven minutes), the maximum value allowed by the specification.

default_delay

Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 10.

ignore_loop

GIF files can contain information to loop a certain number of times (or infinitely). If ignore_loop is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set to 0, then looping will occur and will cycle the number of times according to the GIF. Default value is 1.

For example, with the overlay filter, place an infinitely looping GIF over another video:

ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv

Note that in the above example the shortest option for overlay filter is used to end the output video at the length of the shortest input file, which in this case is input.mp4 as the GIF in this example loops infinitely.

20.12 hls

HLS demuxer

Apple HTTP Live Streaming demuxer.

This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant index number. By setting the discard flags on AVStreams (by pressing ’a’ or ’v’ in ffplay), the caller can decide which variant streams to actually receive. The total bitrate of the variant that the stream belongs to is available in a metadata key named "variant_bitrate".

It accepts the following options:

live_start_index

segment index to start live streams at (negative values are from the end).

prefer_x_start

prefer to use #EXT-X-START if it’s in playlist instead of live_start_index.

allowed_extensions

’,’ separated list of file extensions that hls is allowed to access.

max_reload

Maximum number of times a insufficient list is attempted to be reloaded. Default value is 1000.

m3u8_hold_counters

The maximum number of times to load m3u8 when it refreshes without new segments. Default value is 1000.

http_persistent

Use persistent HTTP connections. Applicable only for HTTP streams. Enabled by default.

http_multiple

Use multiple HTTP connections for downloading HTTP segments. Enabled by default for HTTP/1.1 servers.

http_seekable

Use HTTP partial requests for downloading HTTP segments. 0 = disable, 1 = enable, -1 = auto, Default is auto.

seg_format_options

Set options for the demuxer of media segments using a list of key=value pairs separated by :.

seg_max_retry

Maximum number of times to reload a segment on error, useful when segment skip on network error is not desired. Default value is 0.

20.13 image2

Image file demuxer.

This demuxer reads from a list of image files specified by a pattern. The syntax and meaning of the pattern is specified by the option pattern_type.

The pattern may contain a suffix which is used to automatically determine the format of the images contained in the files.

The size, the pixel format, and the format of each image must be the same for all the files in the sequence.

This demuxer accepts the following options:

framerate

Set the frame rate for the video stream. It defaults to 25.

loop

If set to 1, loop over the input. Default value is 0.

pattern_type

Select the pattern type used to interpret the provided filename.

pattern_type accepts one of the following values.

none

Disable pattern matching, therefore the video will only contain the specified image. You should use this option if you do not want to create sequences from multiple images and your filenames may contain special pattern characters.

sequence

Select a sequence pattern type, used to specify a sequence of files indexed by sequential numbers.

A sequence pattern may contain the string "%d" or "%0Nd", which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form "%d0Nd" is used, the string representing the number in each filename is 0-padded and N is the total number of 0-padded digits representing the number. The literal character ’%’ can be specified in the pattern with the string "%%".

If the sequence pattern contains "%d" or "%0Nd", the first filename of the file list specified by the pattern must contain a number inclusively contained between start_number and start_number+start_number_range-1, and all the following numbers must be sequential.

For example the pattern "img-%03d.bmp" will match a sequence of filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.; the pattern "i%%m%%g-%d.jpg" will match a sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg, ..., i%m%g-10.jpg, etc.

Note that the pattern must not necessarily contain "%d" or "%0Nd", for example to convert a single image file img.jpeg you can employ the command:

ffmpeg -i img.jpeg img.png
glob

Select a glob wildcard pattern type.

The pattern is interpreted like a glob() pattern. This is only selectable if libavformat was compiled with globbing support.

glob_sequence (deprecated, will be removed)

Select a mixed glob wildcard/sequence pattern.

If your version of libavformat was compiled with globbing support, and the provided pattern contains at least one glob meta character among %*?[]{} that is preceded by an unescaped "%", the pattern is interpreted like a glob() pattern, otherwise it is interpreted like a sequence pattern.

All glob special characters %*?[]{} must be prefixed with "%". To escape a literal "%" you shall use "%%".

For example the pattern foo-%*.jpeg will match all the filenames prefixed by "foo-" and terminating with ".jpeg", and foo-%?%?%?.jpeg will match all the filenames prefixed with "foo-", followed by a sequence of three characters, and terminating with ".jpeg".

This pattern type is deprecated in favor of glob and sequence.

Default value is glob_sequence.

pixel_format

Set the pixel format of the images to read. If not specified the pixel format is guessed from the first image file in the sequence.

start_number

Set the index of the file matched by the image file pattern to start to read from. Default value is 0.

start_number_range

Set the index interval range to check when looking for the first image file in the sequence, starting from start_number. Default value is 5.

ts_from_file

If set to 1, will set frame timestamp to modification time of image file. Note that monotonity of timestamps is not provided: images go in the same order as without this option. Default value is 0. If set to 2, will set frame timestamp to the modification time of the image file in nanosecond precision.

video_size

Set the video size of the images to read. If not specified the video size is guessed from the first image file in the sequence.

export_path_metadata

If set to 1, will add two extra fields to the metadata found in input, making them also available for other filters (see drawtext filter for examples). Default value is 0. The extra fields are described below:

lavf.image2dec.source_path

Corresponds to the full path to the input file being read.

lavf.image2dec.source_basename

Corresponds to the name of the file being read.

20.13.1 Examples

  • Use ffmpeg for creating a video from the images in the file sequence img-001.jpeg, img-002.jpeg, ..., assuming an input frame rate of 10 frames per second:
    ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
    
  • As above, but start by reading from a file with index 100 in the sequence:
    ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
    
  • Read images matching the "*.png" glob pattern , that is all the files terminating with the ".png" suffix:
    ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
    

20.14 libgme

The Game Music Emu library is a collection of video game music file emulators.

See https://bitbucket.org/mpyne/game-music-emu/overview for more information.

It accepts the following options:

track_index

Set the index of which track to demux. The demuxer can only export one track. Track indexes start at 0. Default is to pick the first track. Number of tracks is exported as tracks metadata entry.

sample_rate

Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.

max_size (bytes)

The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Default is 50 MiB.

20.15 libmodplug

ModPlug based module demuxer

See https://github.com/Konstanty/libmodplug

It will export one 2-channel 16-bit 44.1 kHz audio stream. Optionally, a pal8 16-color video stream can be exported with or without printed metadata.

It accepts the following options:

noise_reduction

Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.

reverb_depth

Set amount of reverb. Range 0-100. Default is 0.

reverb_delay

Set delay in ms, clamped to 40-250 ms. Default is 0.

bass_amount

Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.

bass_range

Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.

surround_depth

Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.

surround_delay

Set surround delay in ms, clamped to 5-40 ms. Default is 0.

max_size

The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB. 0 removes buffer size limit (not recommended). Default is 5 MiB.

video_stream_expr

String which is evaluated using the eval API to assign colors to the generated video stream. Variables which can be used are x, y, w, h, t, speed, tempo, order, pattern and row.

video_stream

Generate video stream. Can be 1 (on) or 0 (off). Default is 0.

video_stream_w

Set video frame width in ’chars’ where one char indicates 8 pixels. Range is 20-512. Default is 30.

video_stream_h

Set video frame height in ’chars’ where one char indicates 8 pixels. Range is 20-512. Default is 30.

video_stream_ptxt

Print metadata on video stream. Includes speed, tempo, order, pattern, row and ts (time in ms). Can be 1 (on) or 0 (off). Default is 1.

20.16 libopenmpt

libopenmpt based module demuxer

See https://lib.openmpt.org/libopenmpt/ for more information.

Some files have multiple subsongs (tracks) this can be set with the subsong option.

It accepts the following options:

subsong

Set the subsong index. This can be either ’all’, ’auto’, or the index of the subsong. Subsong indexes start at 0. The default is ’auto’.

The default value is to let libopenmpt choose.

layout

Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO.

sample_rate

Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is 48000.

20.17 mov/mp4/3gp

Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).

Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v

20.17.1 Options

This demuxer accepts the following options:

enable_drefs

Enable loading of external tracks, disabled by default. Enabling this can theoretically leak information in some use cases.

use_absolute_path

Allows loading of external tracks via absolute paths, disabled by default. Enabling this poses a security risk. It should only be enabled if the source is known to be non-malicious.

seek_streams_individually

When seeking, identify the closest point in each stream individually and demux packets in that stream from identified point. This can lead to a different sequence of packets compared to demuxing linearly from the beginning. Default is true.

ignore_editlist

Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the timeline described by the edit list. Default is false.

advanced_editlist

Modify the stream index to reflect the timeline described by the edit list. ignore_editlist must be set to false for this option to be effective. If both ignore_editlist and this option are set to false, then only the start of the stream index is modified to reflect initial dwell time or starting timestamp described by the edit list. Default is true.

ignore_chapters

Don’t parse chapters. This includes GoPro ’HiLight’ tags/moments. Note that chapters are only parsed when input is seekable. Default is false.

use_mfra_for

For seekable fragmented input, set fragment’s starting timestamp from media fragment random access box, if present.

Following options are available:

auto

Auto-detect whether to set mfra timestamps as PTS or DTS (default)

dts

Set mfra timestamps as DTS

pts

Set mfra timestamps as PTS

0

Don’t use mfra box to set timestamps

use_tfdt

For fragmented input, set fragment’s starting timestamp to baseMediaDecodeTime from the tfdt box. Default is enabled, which will prefer to use the tfdt box to set DTS. Disable to use the earliest_presentation_time from the sidx box. In either case, the timestamp from the mfra box will be used if it’s available and use_mfra_for is set to pts or dts.

export_all

Export unrecognized boxes within the udta box as metadata entries. The first four characters of the box type are set as the key. Default is false.

export_xmp

Export entire contents of XMP_ box and uuid box as a string with key xmp. Note that if export_all is set and this option isn’t, the contents of XMP_ box are still exported but with key XMP_. Default is false.

activation_bytes

4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.

audible_fixed_key

Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to specify.

decryption_key

16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

max_stts_delta

Very high sample deltas written in a trak’s stts box may occasionally be intended but usually they are written in error or used to store a negative value for dts correction when treated as signed 32-bit integers. This option lets the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when cast to int32 are used to adjust onward dts.

Unit is the track time scale. Range is 0 to UINT_MAX. Default is UINT_MAX - 48000*10 which allows up to a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of uint32 range.

interleaved_read

Interleave packets from multiple tracks at demuxer level. For badly interleaved files, this prevents playback issues caused by large gaps between packets in different tracks, as MOV/MP4 do not have packet placement requirements. However, this can cause excessive seeking on very badly interleaved files, due to seeking between tracks, so disabling it may prevent I/O issues, at the expense of playback.

20.17.2 Audible AAX

Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.

ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4

20.18 mpegts

MPEG-2 transport stream demuxer.

This demuxer accepts the following options:

resync_size

Set size limit for looking up a new synchronization. Default value is 65536.

skip_unknown_pmt

Skip PMTs for programs not defined in the PAT. Default value is 0.

fix_teletext_pts

Override teletext packet PTS and DTS values with the timestamps calculated from the PCR of the first program which the teletext stream is part of and is not discarded. Default value is 1, set this option to 0 if you want your teletext packet PTS and DTS values untouched.

ts_packetsize

Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be set by the user.

scan_all_pmts

Scan and combine all PMTs. The value is an integer with value from -1 to 1 (-1 means automatic setting, 1 means enabled, 0 means disabled). Default value is -1.

merge_pmt_versions

Re-use existing streams when a PMT’s version is updated and elementary streams move to different PIDs. Default value is 0.

max_packet_size

Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.

20.19 mpjpeg

MJPEG encapsulated in multi-part MIME demuxer.

This demuxer allows reading of MJPEG, where each frame is represented as a part of multipart/x-mixed-replace stream.

strict_mime_boundary

Default implementation applies a relaxed standard to multi-part MIME boundary detection, to prevent regression with numerous existing endpoints not generating a proper MIME MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check of the boundary value.

20.20 rawvideo

Raw video demuxer.

This demuxer allows one to read raw video data. Since there is no header specifying the assumed video parameters, the user must specify them in order to be able to decode the data correctly.

This demuxer accepts the following options:

framerate

Set input video frame rate. Default value is 25.

pixel_format

Set the input video pixel format. Default value is yuv420p.

video_size

Set the input video size. This value must be specified explicitly.

For example to read a rawvideo file input.raw with ffplay, assuming a pixel format of rgb24, a video size of 320x240, and a frame rate of 10 images per second, use the command:

ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw

20.21 rcwt

RCWT (Raw Captions With Time) is a format native to ccextractor, a commonly used open source tool for processing 608/708 Closed Captions (CC) sources. For more information on the format, see (ffmpeg-formats)rcwtenc.

This demuxer implements the specification as of March 2024, which has been stable and unchanged since April 2014.

20.21.1 Examples

  • Render CC to ASS using the built-in decoder:
    ffmpeg -i CC.rcwt.bin CC.ass
    

    Note that if your output appears to be empty, you may have to manually set the decoder’s data_field option to pick the desired CC substream.

  • Convert an RCWT backup to Scenarist (SCC) format:
    ffmpeg -i CC.rcwt.bin -c:s copy CC.scc
    

    Note that the SCC format does not support all of the possible CC extensions that can be stored in RCWT (such as EIA-708).

20.22 sbg

SBaGen script demuxer.

This demuxer reads the script language used by SBaGen http://uazu.net/sbagen/ to generate binaural beats sessions. A SBG script looks like that:

-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW      == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00    off

A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps (including the script start time) or only relative ones, then its layout is fixed, and the conversion is straightforward. On the other hand, if the script mixes both kind of timestamps, then the NOW reference for relative timestamps will be taken from the current time of day at the time the script is read, and the script layout will be frozen according to that reference. That means that if the script is directly played, the actual times will match the absolute timestamps up to the sound controller’s clock accuracy, but if the user somehow pauses the playback or seeks, all times will be shifted accordingly.

20.23 tedcaptions

JSON captions used for TED Talks.

TED does not provide links to the captions, but they can be guessed from the page. The file tools/bookmarklets.html from the FFmpeg source tree contains a bookmarklet to expose them.

This demuxer accepts the following option:

start_time

Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is used to sync the captions with the downloadable videos, because they include a 15s intro.

Example: convert the captions to a format most players understand:

ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt

20.24 vapoursynth

Vapoursynth wrapper.

Due to security concerns, Vapoursynth scripts will not be autodetected so the input format has to be forced. For ff* CLI tools, add -f vapoursynth before the input -i yourscript.vpy.

This demuxer accepts the following option:

max_script_size

The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of scripts that can be read. Default is 1 MiB.

20.25 w64

Sony Wave64 Audio demuxer.

This demuxer accepts the following options:

max_size

See the same option for the wav demuxer.

20.26 wav

RIFF Wave Audio demuxer.

This demuxer accepts the following options:

max_size

Specify the maximum packet size in bytes for the demuxed packets. By default this is set to 0, which means that a sensible value is chosen based on the input format.

21 Muxers

Muxers are configured elements in FFmpeg which allow writing multimedia streams to a particular type of file.

When you configure your FFmpeg build, all the supported muxers are enabled by default. You can list all available muxers using the configure option --list-muxers.

You can disable all the muxers with the configure option --disable-muxers and selectively enable / disable single muxers with the options --enable-muxer=MUXER / --disable-muxer=MUXER.

The option -muxers of the ff* tools will display the list of enabled muxers. Use -formats to view a combined list of enabled demuxers and muxers.

A description of some of the currently available muxers follows.

21.1 Raw muxers

This section covers raw muxers. They accept a single stream matching the designated codec. They do not store timestamps or metadata. The recognized extension is the same as the muxer name unless indicated otherwise.

It comprises the following muxers. The media type and the eventual extensions used to automatically selects the muxer from the output extensions are also shown.

ac3 audio

Dolby Digital, also known as AC-3.

adx audio

CRI Middleware ADX audio.

This muxer will write out the total sample count near the start of the first packet when the output is seekable and the count can be stored in 32 bits.

aptx audio

aptX (Audio Processing Technology for Bluetooth)

aptx_hd audio (aptxdh)

aptX HD (Audio Processing Technology for Bluetooth) audio

avs2 video (avs, avs2)

AVS2-P2 (Audio Video Standard - Second generation - Part 2) / IEEE 1857.4 video

avs3 video (avs3)

AVS3-P2 (Audio Video Standard - Third generation - Part 2) / IEEE 1857.10 video

cavsvideo video (cavs)

Chinese AVS (Audio Video Standard - First generation)

codec2raw audio

Codec 2 audio.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool -f codec2raw.

data any

Generic data muxer.

This muxer accepts a single stream with any codec of any type. The input stream has to be selected using the -map option with the ffmpeg CLI tool.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool -f data.

dfpwm audio (dfpwm)

Raw DFPWM1a (Dynamic Filter Pulse With Modulation) audio muxer.

dirac video (drc, vc2)

BBC Dirac video.

The Dirac Pro codec is a subset and is standardized as SMPTE VC-2.

dnxhd video (dnxhd, dnxhr)

Avid DNxHD video.

It is standardized as SMPTE VC-3. Accepts DNxHR streams.

dts audio

DTS Coherent Acoustics (DCA) audio

eac3 audio

Dolby Digital Plus, also known as Enhanced AC-3

evc video (evc)

MPEG-5 Essential Video Coding (EVC) / EVC / MPEG-5 Part 1 EVC video

g722 audio

ITU-T G.722 audio

g723_1 audio (tco, rco)

ITU-T G.723.1 audio

g726 audio

ITU-T G.726 big-endian ("left-justified") audio.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool -f g726.

g726le audio

ITU-T G.726 little-endian ("right-justified") audio.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool -f g726le.

gsm audio

Global System for Mobile Communications audio

h261 video

ITU-T H.261 video

h263 video

ITU-T H.263 / H.263-1996, H.263+ / H.263-1998 / H.263 version 2 video

h264 video (h264, 264)

ITU-T H.264 / MPEG-4 Part 10 AVC video. Bitstream shall be converted to Annex B syntax if it’s in length-prefixed mode.

hevc video (hevc, h265, 265)

ITU-T H.265 / MPEG-H Part 2 HEVC video. Bitstream shall be converted to Annex B syntax if it’s in length-prefixed mode.

m4v video

MPEG-4 Part 2 video

mjpeg video (mjpg, mjpeg)

Motion JPEG video

mlp audio

Meridian Lossless Packing, also known as Packed PCM

mp2 audio (mp2, m2a, mpa)

MPEG-1 Audio Layer II audio

mpeg1video video (mpg, mpeg, m1v)

MPEG-1 Part 2 video.

mpeg2video video (m2v)

ITU-T H.262 / MPEG-2 Part 2 video

obu video

AV1 low overhead Open Bitstream Units muxer.

Temporal delimiter OBUs will be inserted in all temporal units of the stream.

rawvideo video (yuv, rgb)

Raw uncompressed video.

sbc audio (sbc, msbc)

Bluetooth SIG low-complexity subband codec audio

truehd audio (thd)

Dolby TrueHD audio

vc1 video

SMPTE 421M / VC-1 video

21.1.1 Examples

  • Store raw video frames with the ‘rawvideo’ muxer using ffmpeg:
    ffmpeg -f lavfi -i testsrc -t 10 -s hd1080p testsrc.yuv
    

    Since the rawvideo muxer do not store the information related to size and format, this information must be provided when demuxing the file:

    ffplay -video_size 1920x1080 -pixel_format rgb24 -f rawvideo testsrc.rgb
    

21.2 Raw PCM muxers

This section covers raw PCM (Pulse-Code Modulation) audio muxers.

They accept a single stream matching the designated codec. They do not store timestamps or metadata. The recognized extension is the same as the muxer name.

It comprises the following muxers. The optional additional extension used to automatically select the muxer from the output extension is also shown in parentheses.

alaw (al)

PCM A-law

f32be

PCM 32-bit floating-point big-endian

f32le

PCM 32-bit floating-point little-endian

f64be

PCM 64-bit floating-point big-endian

f64le

PCM 64-bit floating-point little-endian

mulaw (ul)

PCM mu-law

s16be

PCM signed 16-bit big-endian

s16le

PCM signed 16-bit little-endian

s24be

PCM signed 24-bit big-endian

s24le

PCM signed 24-bit little-endian

s32be

PCM signed 32-bit big-endian

s32le

PCM signed 32-bit little-endian

s8 (sb)

PCM signed 8-bit

u16be

PCM unsigned 16-bit big-endian

u16le

PCM unsigned 16-bit little-endian

u24be

PCM unsigned 24-bit big-endian

u24le

PCM unsigned 24-bit little-endian

u32be

PCM unsigned 32-bit big-endian

u32le

PCM unsigned 32-bit little-endian

u8 (ub)

PCM unsigned 8-bit

vidc

PCM Archimedes VIDC

21.3 MPEG-1/MPEG-2 program stream muxers

This section covers formats belonging to the MPEG-1 and MPEG-2 Systems family.

The MPEG-1 Systems format (also known as ISO/IEEC 11172-1 or MPEG-1 program stream) has been adopted for the format of media track stored in VCD (Video Compact Disc).

The MPEG-2 Systems standard (also known as ISO/IEEC 13818-1) covers two containers formats, one known as transport stream and one known as program stream; only the latter is covered here.

The MPEG-2 program stream format (also known as VOB due to the corresponding file extension) is an extension of MPEG-1 program stream: in addition to support different codecs for the audio and video streams, it also stores subtitles and navigation metadata. MPEG-2 program stream has been adopted for storing media streams in SVCD and DVD storage devices.

This section comprises the following muxers.

mpeg (mpg,mpeg)

MPEG-1 Systems / MPEG-1 program stream muxer.

vcd

MPEG-1 Systems / MPEG-1 program stream (VCD) muxer.

This muxer can be used to generate tracks in the format accepted by the VCD (Video Compact Disc) storage devices.

It is the same as the ‘mpeg’ muxer with a few differences.

vob

MPEG-2 program stream (VOB) muxer.

dvd

MPEG-2 program stream (DVD VOB) muxer.

This muxer can be used to generate tracks in the format accepted by the DVD (Digital Versatile Disc) storage devices.

This is the same as the ‘vob’ muxer with a few differences.

svcd (vob)

MPEG-2 program stream (SVCD VOB) muxer.

This muxer can be used to generate tracks in the format accepted by the SVCD (Super Video Compact Disc) storage devices.

This is the same as the ‘vob’ muxer with a few differences.

21.3.1 Options

muxrate rate

Set user-defined mux rate expressed as a number of bits/s. If not specied the automatically computed mux rate is employed. Default value is 0.

preload delay

Set initial demux-decode delay in microseconds. Default value is 500000.

21.4 MOV/MPEG-4/ISOMBFF muxers

This section covers formats belonging to the QuickTime / MOV family, including the MPEG-4 Part 14 format and ISO base media file format (ISOBMFF). These formats share a common structure based on the ISO base media file format (ISOBMFF).

The MOV format was originally developed for use with Apple QuickTime. It was later used as the basis for the MPEG-4 Part 1 (later Part 14) format, also known as ISO/IEC 14496-1. That format was then generalized into ISOBMFF, also named MPEG-4 Part 12 format, ISO/IEC 14496-12, or ISO/IEC 15444-12.

It comprises the following muxers.

3gp

Third Generation Partnership Project (3GPP) format for 3G UMTS multimedia services

3g2

Third Generation Partnership Project 2 (3GP2 or 3GPP2) format for 3G CDMA2000 multimedia services, similar to ‘3gp’ with extensions and limitations

f4v

Adobe Flash Video format

ipod

MPEG-4 audio file format, as MOV/MP4 but limited to contain only audio streams, typically played with the Apple ipod device

ismv

Microsoft IIS (Internet Information Services) Smooth Streaming Audio/Video (ISMV or ISMA) format. This is based on MPEG-4 Part 14 format with a few incompatible variants, used to stream media files for the Microsoft IIS server.

mov

QuickTime player format identified by the .mov extension

mp4

MP4 or MPEG-4 Part 14 format

psp

PlayStation Portable MP4/MPEG-4 Part 14 format variant. This is based on MPEG-4 Part 14 format with a few incompatible variants, used to play files on PlayStation devices.

21.4.1 Fragmentation

The ‘mov’, ‘mp4’, and ‘ismv’ muxers support fragmentation. Normally, a MOV/MP4 file has all the metadata about all packets stored in one location.

This data is usually written at the end of the file, but it can be moved to the start for better playback by adding +faststart to the -movflags, or using the qt-faststart tool).

A fragmented file consists of a number of fragments, where packets and metadata about these packets are stored together. Writing a fragmented file has the advantage that the file is decodable even if the writing is interrupted (while a normal MOV/MP4 is undecodable if it is not properly finished), and it requires less memory when writing very long files (since writing normal MOV/MP4 files stores info about every single packet in memory until the file is closed). The downside is that it is less compatible with other applications.

Fragmentation is enabled by setting one of the options that define how to cut the file into fragments:

frag_duration
frag_size
min_frag_duration
movflags +frag_keyframe
movflags +frag_custom

If more than one condition is specified, fragments are cut when one of the specified conditions is fulfilled. The exception to this is the option min_frag_duration, which has to be fulfilled for any of the other conditions to apply.

21.4.2 Options

brand brand_string

Override major brand.

empty_hdlr_name bool

Enable to skip writing the name inside a hdlr box. Default is false.

encryption_key key

set the media encryption key in hexadecimal format

encryption_kid kid

set the media encryption key identifier in hexadecimal format

encryption_scheme scheme

configure the encryption scheme, allowed values are ‘none’, and ‘cenc-aes-ctr

frag_duration duration

Create fragments that are duration microseconds long.

frag_interleave number

Interleave samples within fragments (max number of consecutive samples, lower is tighter interleaving, but with more overhead. It is set to 0 by default.

frag_size size

create fragments that contain up to size bytes of payload data

iods_audio_profile profile

specify iods number for the audio profile atom (from -1 to 255), default is -1

iods_video_profile profile

specify iods number for the video profile atom (from -1 to 255), default is -1

ism_lookahead num_entries

specify number of lookahead entries for ISM files (from 0 to 255), default is 0

min_frag_duration duration

do not create fragments that are shorter than duration microseconds long

moov_size bytes

Reserves space for the moov atom at the beginning of the file instead of placing the moov atom at the end. If the space reserved is insufficient, muxing will fail.

mov_gamma gamma

specify gamma value for gama atom (as a decimal number from 0 to 10), default is 0.0, must be set together with + movflags

movflags flags

Set various muxing switches. The following flags can be used:

cmaf

write CMAF (Common Media Application Format) compatible fragmented MP4 output

dash

write DASH (Dynamic Adaptive Streaming over HTTP) compatible fragmented MP4 output

default_base_moof

Similarly to the ‘omit_tfhd_offset’ flag, this flag avoids writing the absolute base_data_offset field in tfhd atoms, but does so by using the new default-base-is-moof flag instead. This flag is new from 14496-12:2012. This may make the fragments easier to parse in certain circumstances (avoiding basing track fragment location calculations on the implicit end of the previous track fragment).

delay_moov

delay writing the initial moov until the first fragment is cut, or until the first fragment flush

disable_chpl

Disable Nero chapter markers (chpl atom). Normally, both Nero chapters and a QuickTime chapter track are written to the file. With this option set, only the QuickTime chapter track will be written. Nero chapters can cause failures when the file is reprocessed with certain tagging programs, like mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well.

faststart

Run a second pass moving the index (moov atom) to the beginning of the file. This operation can take a while, and will not work in various situations such as fragmented output, thus it is not enabled by default.

frag_custom

Allow the caller to manually choose when to cut fragments, by calling av_write_frame(ctx, NULL) to write a fragment with the packets written so far. (This is only useful with other applications integrating libavformat, not from ffmpeg.)

frag_discont

signal that the next fragment is discontinuous from earlier ones

frag_every_frame

fragment at every frame

frag_keyframe

start a new fragment at each video keyframe

global_sidx

write a global sidx index at the start of the file

isml

create a live smooth streaming feed (for pushing to a publishing point)

negative_cts_offsets

Enables utilization of version 1 of the CTTS box, in which the CTS offsets can be negative. This enables the initial sample to have DTS/CTS of zero, and reduces the need for edit lists for some cases such as video tracks with B-frames. Additionally, eases conformance with the DASH-IF interoperability guidelines.

This option is implicitly set when writing ‘ismv’ (Smooth Streaming) files.

omit_tfhd_offset

Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute byte positions in the file/streams.

prefer_icc

If writing colr atom prioritise usage of ICC profile if it exists in stream packet side data.

rtphint

add RTP hinting tracks to the output file

separate_moof

Write a separate moof (movie fragment) atom for each track. Normally, packets for all tracks are written in a moof atom (which is slightly more efficient), but with this option set, the muxer writes one moof/mdat pair for each track, making it easier to separate tracks.

skip_sidx

Skip writing of sidx atom. When bitrate overhead due to sidx atom is high, this option could be used for cases where sidx atom is not mandatory. When the ‘global_sidx’ flag is enabled, this option is ignored.

skip_trailer

skip writing the mfra/tfra/mfro trailer for fragmented files

use_metadata_tags

use mdta atom for metadata

write_colr

write colr atom even if the color info is unspecified. This flag is experimental, may be renamed or changed, do not use from scripts.

write_gama

write deprecated gama atom

hybrid_fragmented

For recoverability - write the output file as a fragmented file. This allows the intermediate file to be read while being written (in particular, if the writing process is aborted uncleanly). When writing is finished, the file is converted to a regular, non-fragmented file, which is more compatible and allows easier and quicker seeking.

If writing is aborted, the intermediate file can manually be remuxed to get a regular, non-fragmented file of what had been written into the unfinished file.

movie_timescale scale

Set the timescale written in the movie header box (mvhd). Range is 1 to INT_MAX. Default is 1000.

rtpflags flags

Add RTP hinting tracks to the output file.

The following flags can be used:

h264_mode0

use mode 0 for H.264 in RTP

latm

use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC

rfc2190

use RFC 2190 packetization instead of RFC 4629 for H.263

send_bye

send RTCP BYE packets when finishing

skip_rtcp

do not send RTCP sender reports

skip_iods bool

skip writing iods atom (default value is true)

use_editlist bool

use edit list (default value is auto)

use_stream_ids_as_track_ids bool

use stream ids as track ids (default value is false)

video_track_timescale scale

Set the timescale used for video tracks. Range is 0 to INT_MAX. If set to 0, the timescale is automatically set based on the native stream time base. Default is 0.

write_btrt bool

Force or disable writing bitrate box inside stsd box of a track. The box contains decoding buffer size (in bytes), maximum bitrate and average bitrate for the track. The box will be skipped if none of these values can be computed. Default is -1 or auto, which will write the box only in MP4 mode.

write_prft option

Write producer time reference box (PRFT) with a specified time source for the NTP field in the PRFT box. Set value as ‘wallclock’ to specify timesource as wallclock time and ‘pts’ to specify timesource as input packets’ PTS values.

write_tmcd bool

Specify on to force writing a timecode track, off to disable it and auto to write a timecode track only for mov and mp4 output (default).

Setting value to ‘pts’ is applicable only for a live encoding use case, where PTS values are set as as wallclock time at the source. For example, an encoding use case with decklink capture source where video_pts and audio_pts are set to ‘abs_wallclock’.

21.4.3 Examples

  • Push Smooth Streaming content in real time to a publishing point on IIS with the ‘ismv’ muxer using ffmpeg:
    ffmpeg -re <normal input/transcoding options> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
    

21.5 a64

A64 Commodore 64 video muxer.

This muxer accepts a single a64_multi or a64_multi5 codec video stream.

21.6 ac4

Raw AC-4 audio muxer.

This muxer accepts a single ac4 audio stream.

21.6.1 Options

write_crc bool

when enabled, write a CRC checksum for each packet to the output, default is false

21.7 adts

Audio Data Transport Stream muxer.

It accepts a single AAC stream.

21.7.1 Options

write_id3v2 bool

Enable to write ID3v2.4 tags at the start of the stream. Default is disabled.

write_apetag bool

Enable to write APE tags at the end of the stream. Default is disabled.

write_mpeg2 bool

Enable to set MPEG version bit in the ADTS frame header to 1 which indicates MPEG-2. Default is 0, which indicates MPEG-4.

21.8 aea

MD STUDIO audio muxer.

This muxer accepts a single ATRAC1 audio stream with either one or two channels and a sample rate of 44100Hz.

As AEA supports storing the track title, this muxer will also write the title from stream’s metadata to the container.

21.9 aiff

Audio Interchange File Format muxer.

21.9.1 Options

write_id3v2 bool

Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).

id3v2_version bool

Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are supported. The default is version 4.

21.10 alp

High Voltage Software’s Lego Racers game audio muxer.

It accepts a single ADPCM_IMA_ALP stream with no more than 2 channels and a sample rate not greater than 44100 Hz.

Extensions: tun, pcm

21.10.1 Options

type type

Set file type.

type accepts the following values:

tun

Set file type as music. Must have a sample rate of 22050 Hz.

pcm

Set file type as sfx.

auto

Set file type as per output file extension. .pcm results in type pcm else type tun is set. (default)

21.11 amr

3GPP AMR (Adaptive Multi-Rate) audio muxer.

It accepts a single audio stream containing an AMR NB stream.

21.12 amv

AMV (Actions Media Video) format muxer.

21.13 apm

Ubisoft Rayman 2 APM audio muxer.

It accepts a single ADPCM IMA APM audio stream.

21.14 apng

Animated Portable Network Graphics muxer.

It accepts a single APNG video stream.

21.14.1 Options

final_delay delay

Force a delay expressed in seconds after the last frame of each repetition. Default value is 0.0.

plays repetitions

specify how many times to play the content, 0 causes an infinte loop, with 1 there is no loop

21.14.2 Examples

  • Use ffmpeg to generate an APNG output with 2 repetitions, and with a delay of half a second after the first repetition:
    ffmpeg -i INPUT -final_delay 0.5 -plays 2 out.apng
    

21.15 argo_asf

Argonaut Games ASF audio muxer.

It accepts a single ADPCM audio stream.

21.15.1 Options

version_major version

override file major version, specified as an integer, default value is 2

version_minor version

override file minor version, specified as an integer, default value is 1

name name

Embed file name into file, if not specified use the output file name. The name is truncated to 8 characters.

21.16 argo_cvg

Argonaut Games CVG audio muxer.

It accepts a single one-channel ADPCM 22050Hz audio stream.

The loop and reverb options set the corresponding flags in the header which can be later retrieved to process the audio stream accordingly.

21.16.1 Options

skip_rate_check bool

skip sample rate check (default is false)

loop bool

set loop flag (default is false)

reverb boolean

set reverb flag (default is true)

21.17 asf, asf_stream

Advanced / Active Systems (or Streaming) Format audio muxer.

The ‘asf_stream’ variant should be selected for streaming.

Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too.

21.17.1 Options

packet_size size

Set the muxer packet size as a number of bytes. By tuning this setting you may reduce data fragmentation or muxer overhead depending on your source. Default value is 3200, minimum is 100, maximum is 64Ki.

21.18 ass

ASS/SSA (SubStation Alpha) subtitles muxer.

It accepts a single ASS subtitles stream.

21.18.1 Options

ignore_readorder bool

Write dialogue events immediately, even if they are out-of-order, default is false, otherwise they are cached until the expected time event is found.

21.19 ast

AST (Audio Stream) muxer.

This format is used to play audio on some Nintendo Wii games.

It accepts a single audio stream.

The loopstart and loopend options can be used to define a section of the file to loop for players honoring such options.

21.19.1 Options

loopstart start

Specify loop start position expressesd in milliseconds, from -1 to INT_MAX, in case -1 is set then no loop is specified (default -1) and the loopend value is ignored.

loopend end

Specify loop end position expressed in milliseconds, from 0 to INT_MAX, default is 0, in case 0 is set it assumes the total stream duration.

21.20 au

SUN AU audio muxer.

It accepts a single audio stream.

21.21 avi

Audio Video Interleaved muxer.

AVI is a proprietary format developed by Microsoft, and later formally specified through the Open DML specification.

Because of differences in players implementations, it might be required to set some options to make sure that the generated output can be correctly played by the target player.

21.21.1 Options

flipped_raw_rgb bool

If set to true, store positive height for raw RGB bitmaps, which indicates bitmap is stored bottom-up. Note that this option does not flip the bitmap which has to be done manually beforehand, e.g. by using the ‘vflip’ filter. Default is false and indicates bitmap is stored top down.

reserve_index_space size

Reserve the specified amount of bytes for the OpenDML master index of each stream within the file header. By default additional master indexes are embedded within the data packets if there is no space left in the first master index and are linked together as a chain of indexes. This index structure can cause problems for some use cases, e.g. third-party software strictly relying on the OpenDML index specification or when file seeking is slow. Reserving enough index space in the file header avoids these problems.

The required index space depends on the output file size and should be about 16 bytes per gigabyte. When this option is omitted or set to zero the necessary index space is guessed.

Default value is 0.

write_channel_mask bool

Write the channel layout mask into the audio stream header.

This option is enabled by default. Disabling the channel mask can be useful in specific scenarios, e.g. when merging multiple audio streams into one for compatibility with software that only supports a single audio stream in AVI (see (ffmpeg-filters)the "amerge" section in the ffmpeg-filters manual).

21.22 avif

AV1 (Alliance for Open Media Video codec 1) image format muxer.

This muxers stores images encoded using the AV1 codec.

It accepts one or two video streams. In case two video streams are provided, the second one shall contain a single plane storing the alpha mask.

In case more than one image is provided, the generated output is considered an animated AVIF and the number of loops can be specified with the loop option.

This is based on the specification by Alliance for Open Media at url https://aomediacodec.github.io/av1-avif.

21.22.1 Options

loop count

number of times to loop an animated AVIF, 0 specify an infinite loop, default is 0

movie_timescale timescale

Set the timescale written in the movie header box (mvhd). Range is 1 to INT_MAX. Default is 1000.

21.23 avm2

ShockWave Flash (SWF) / ActionScript Virtual Machine 2 (AVM2) format muxer.

It accepts one audio stream, one video stream, or both.

21.24 bit

G.729 (.bit) file format muxer.

It accepts a single G.729 audio stream.

21.25 caf

Apple CAF (Core Audio Format) muxer.

It accepts a single audio stream.

21.26 codec2

Codec2 audio audio muxer.

It accepts a single codec2 audio stream.

21.27 chromaprint

Chromaprint fingerprinter muxers.

To enable compilation of this filter you need to configure FFmpeg with --enable-chromaprint.

This muxer feeds audio data to the Chromaprint library, which generates a fingerprint for the provided audio data. See: https://acoustid.org/chromaprint

It takes a single signed native-endian 16-bit raw audio stream of at most 2 channels.

21.27.1 Options

algorithm version

Select version of algorithm to fingerprint with. Range is 0 to 4. Version 3 enables silence detection. Default is 1.

fp_format format

Format to output the fingerprint as. Accepts the following options:

base64

Base64 compressed fingerprint (default)

compressed

Binary compressed fingerprint

raw

Binary raw fingerprint

silence_threshold threshold

Threshold for detecting silence. Range is from -1 to 32767, where -1 disables silence detection. Silence detection can only be used with version 3 of the algorithm.

Silence detection must be disabled for use with the AcoustID service. Default is -1.

21.28 crc

CRC (Cyclic Redundancy Check) muxer.

This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.

The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames.

See also the framecrc muxer.

21.28.1 Examples

  • Use ffmpeg to compute the CRC of the input, and store it in the file out.crc:
    ffmpeg -i INPUT -f crc out.crc
    
  • Use ffmpeg to print the CRC to stdout with the command:
    ffmpeg -i INPUT -f crc -
    
  • You can select the output format of each frame with ffmpeg by specifying the audio and video codec and format. For example, to compute the CRC of the input audio converted to PCM unsigned 8-bit and the input video converted to MPEG-2 video, use the command:
    ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
    

21.29 dash

Dynamic Adaptive Streaming over HTTP (DASH) muxer.

This muxer creates segments and manifest files according to the MPEG-DASH standard ISO/IEC 23009-1:2014 and following standard updates.

For more information see:

This muxer creates an MPD (Media Presentation Description) manifest file and segment files for each stream. Segment files are placed in the same directory of the MPD manifest file.

The segment filename might contain pre-defined identifiers used in the manifest SegmentTemplate section as defined in section 5.3.9.4.4 of the standard.

Available identifiers are $RepresentationID$, $Number$, $Bandwidth$, and $Time$. In addition to the standard identifiers, an ffmpeg-specific $ext$ identifier is also supported. When specified, ffmpeg will replace $ext$ in the file name with muxing format’s extensions such as mp4, webm etc.

21.29.1 Options

adaptation_sets adaptation_sets

Assign streams to adaptation sets, specified in the MPD manifest AdaptationSets section.

An adaptation set contains a set of one or more streams accessed as a single subset, e.g. corresponding streams encoded at different size selectable by the user depending on the available bandwidth, or to different audio streams with a different language.

Each adaptation set is specified with the syntax:

id=index,streams=streams

where index must be a numerical index, and streams is a sequence of ,-separated stream indices. Multiple adaptation sets can be specified, separated by spaces.

To map all video (or audio) streams to an adaptation set, v (or a) can be used as stream identifier instead of IDs.

When no assignment is defined, this defaults to an adaptation set for each stream.

The following optional fields can also be specified:

descriptor

Define the descriptor as defined by ISO/IEC 23009-1:2014/Amd.2:2015.

For example:

<SupplementalProperty schemeIdUri=\"urn:mpeg:dash:srd:2014\" value=\"0,0,0,1,1,2,2\"/>

The descriptor string should be a self-closing XML tag.

frag_duration

Override the global fragment duration specified with the frag_duration option.

frag_type

Override the global fragment type specified with the frag_type option.

seg_duration

Override the global segment duration specified with the seg_duration option.

trick_id

Mark an adaptation set as containing streams meant to be used for Trick Mode for the referenced adaptation set.

A few examples of possible values for the adaptation_sets option follow:

id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a
id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1
dash_segment_type type

Set DASH segment files type.

Possible values:

auto

The dash segment files format will be selected based on the stream codec. This is the default mode.

mp4

the dash segment files will be in ISOBMFF/MP4 format

webm

the dash segment files will be in WebM format

extra_window_size size

Set the maximum number of segments kept outside of the manifest before removing from disk.

format_options options_list

Set container format (mp4/webm) options using a :-separated list of key=value parameters. Values containing : special characters must be escaped.

frag_duration duration

Set the length in seconds of fragments within segments, fractional value can also be set.

frag_type type

Set the type of interval for fragmentation.

Possible values:

auto

set one fragment per segment

every_frame

fragment at every frame

duration

fragment at specific time intervals

pframes

fragment at keyframes and following P-Frame reordering (Video only, experimental)

global_sidx bool

Write global SIDX atom. Applicable only for single file, mp4 output, non-streaming mode.

hls_master_name file_name

HLS master playlist name. Default is master.m3u8.

hls_playlist bool

Generate HLS playlist files. The master playlist is generated with filename specified by the hls_master_name option. One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.

http_opts http_opts

Specify a list of :-separated key=value options to pass to the underlying HTTP protocol. Applicable only for HTTP output.

http_persistent bool

Use persistent HTTP connections. Applicable only for HTTP output.

http_user_agent user_agent

Override User-Agent field in HTTP header. Applicable only for HTTP output.

ignore_io_errors bool

Ignore IO errors during open and write. Useful for long-duration runs with network output. This is disabled by default.

index_correction bool

Enable or disable segment index correction logic. Applicable only when use_template is enabled and use_timeline is disabled. This is disabled by default.

When enabled, the logic monitors the flow of segment indexes. If a streams’s segment index value is not at the expected real time position, then the logic corrects that index value.

Typically this logic is needed in live streaming use cases. The network bandwidth fluctuations are common during long run streaming. Each fluctuation can cause the segment indexes fall behind the expected real time position.

init_seg_name init_name

DASH-templated name to use for the initialization segment. Default is init-stream$RepresentationID$.$ext$. $ext$ is replaced with the file name extension specific for the segment format.

ldash bool

Enable Low-latency Dash by constraining the presence and values of some elements. This is disabled by default.

lhls bool

Enable Low-latency HLS (LHLS). Add #EXT-X-PREFETCH tag with current segment’s URI. hls.js player folks are trying to standardize an open LHLS spec. The draft spec is available at https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md.

This option tries to comply with the above open spec. It enables streaming and hls_playlist options automatically. This is an experimental feature.

Note: This is not Apple’s version LHLS. See https://datatracker.ietf.org/doc/html/draft-pantos-hls-rfc8216bis

master_m3u8_publish_rate segment_intervals_count

Publish master playlist repeatedly every after specified number of segment intervals.

max_playback_rate rate

Set the maximum playback rate indicated as appropriate for the purposes of automatically adjusting playback latency and buffer occupancy during normal playback by clients.

media_seg_name segment_name

DASH-templated name to use for the media segments. Default is chunk-stream$RepresentationID$-$Number%05d$.$ext$. $ext$ is replaced with the file name extension specific for the segment format.

method method

Use the given HTTP method to create output files. Generally set to PUT or POST.

min_playback_rate rate

Set the minimum playback rate indicated as appropriate for the purposes of automatically adjusting playback latency and buffer occupancy during normal playback by clients.

mpd_profile flags

Set one or more MPD manifest profiles.

Possible values:

dash

MPEG-DASH ISO Base media file format live profile

dvb_dash

DVB-DASH profile

Default value is dash.

remove_at_exit bool

Enable or disable removal of all segments when finished. This is disabled by default.

seg_duration duration

Set the segment length in seconds (fractional value can be set). The value is treated as average segment duration when the use_template option is enabled and the use_timeline option is disabled and as minimum segment duration for all the other use cases.

Default value is 5.

single_file bool

Enable or disable storing all segments in one file, accessed using byte ranges. This is disabled by default.

The name of the single file can be specified with the single_file_name option, if not specified assume the basename of the manifest file with the output format extension.

single_file_name file_name

DASH-templated name to use for the manifest baseURL element. Imply that the single_file option is set to true. In the template, $ext$ is replaced with the file name extension specific for the segment format.

streaming bool

Enable or disable chunk streaming mode of output. In chunk streaming mode, each frame will be a moof fragment which forms a chunk. This is disabled by default.

target_latency target_latency

Set an intended target latency in seconds for serving (fractional value can be set). Applicable only when the streaming and write_prft options are enabled. This is an informative fields clients can use to measure the latency of the service.

timeout timeout

Set timeout for socket I/O operations expressed in seconds (fractional value can be set). Applicable only for HTTP output.

update_period period

Set the MPD update period, for dynamic content. The unit is second. If set to 0, the period is automatically computed.

Default value is 0.

use_template bool

Enable or disable use of SegmentTemplate instead of SegmentList in the manifest. This is enabled by default.

use_timeline bool

Enable or disable use of SegmentTimeline within the SegmentTemplate manifest section. This is enabled by default.

utc_timing_url url

URL of the page that will return the UTC timestamp in ISO format, for example https://time.akamai.com/?iso

window_size size

Set the maximum number of segments kept in the manifest, discard the oldest one. This is useful for live streaming.

If the value is 0, all segments are kept in the manifest. Default value is 0.

write_prft write_prft

Write Producer Reference Time elements on supported streams. This also enables writing prft boxes in the underlying muxer. Applicable only when the utc_url option is enabled. It is set to auto by default, in which case the muxer will attempt to enable it only in modes that require it.

21.29.2 Example

Generate a DASH output reading from an input source in realtime using ffmpeg.

Two multimedia streams are generated from the input file, both containing a video stream encoded through ‘libx264’, and an audio stream encoded with ‘libfdk_aac’. The first multimedia stream contains video with a bitrate of 800k and audio at the default rate, the second with video scaled to 320x170 pixels at 300k and audio resampled at 22005 Hz.

The window_size option keeps only the latest 5 segments with the default duration of 5 seconds.

ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
-b:v:0 800k -profile:v:0 main \
-b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline -ar:a:1 22050 \
-bf 1 -keyint_min 120 -g 120 -sc_threshold 0 -b_strategy 0 \
-use_timeline 1 -use_template 1 -window_size 5 \
-adaptation_sets "id=0,streams=v id=1,streams=a" \
-f dash /path/to/out.mpd

21.30 daud

D-Cinema audio muxer.

It accepts a single 6-channels audio stream resampled at 96000 Hz encoded with the ‘pcm_24daud’ codec.

21.30.1 Example

Use ffmpeg to mux input audio to a ‘5.1’ channel layout resampled at 96000Hz:

ffmpeg -i INPUT -af aresample=96000,pan=5.1 slow.302

For ffmpeg versions before 7.0 you might have to use the ‘asetnsamples’ filter to limit the muxed packet size, because this format does not support muxing packets larger than 65535 bytes (3640 samples). For newer ffmpeg versions audio is automatically packetized to 36000 byte (2000 sample) packets.

21.31 dv

DV (Digital Video) muxer.

It accepts exactly one ‘dvvideo’ video stream and at most two ‘pcm_s16’ audio streams. More constraints are defined by the property of the video, which must correspond to a DV video supported profile, and on the framerate.

21.31.1 Example

Use ffmpeg to convert the input:

ffmpeg -i INPUT -s:v 720x480 -pix_fmt yuv411p -r 29.97 -ac 2 -ar 48000 -y out.dv

21.32 ffmetadata

FFmpeg metadata muxer.

This muxer writes the streams metadata in the ‘ffmetadata’ format.

See (ffmpeg-formats)the Metadata chapter for information about the format.

21.32.1 Example

Use ffmpeg to extract metadata from an input file to a metadata.ffmeta file in ‘ffmetadata’ format:

ffmpeg -i INPUT -f ffmetadata metadata.ffmeta

21.33 fifo

FIFO (First-In First-Out) muxer.

The ‘fifo’ pseudo-muxer allows the separation of encoding and muxing by using a first-in-first-out queue and running the actual muxer in a separate thread.

This is especially useful in combination with the tee muxer and can be used to send data to several destinations with different reliability/writing speed/latency.

The target muxer is either selected from the output name or specified through the fifo_format option.

The behavior of the ‘fifo’ muxer if the queue fills up or if the output fails (e.g. if a packet cannot be written to the output) is selectable:

  • Output can be transparently restarted with configurable delay between retries based on real time or time of the processed stream.
  • Encoding can be blocked during temporary failure, or continue transparently dropping packets in case the FIFO queue fills up.

API users should be aware that callback functions (interrupt_callback, io_open and io_close) used within its AVFormatContext must be thread-safe.

21.33.1 Options

attempt_recovery bool

If failure occurs, attempt to recover the output. This is especially useful when used with network output, since it makes it possible to restart streaming transparently. By default this option is set to false.

drop_pkts_on_overflow bool

If set to true, in case the fifo queue fills up, packets will be dropped rather than blocking the encoder. This makes it possible to continue streaming without delaying the input, at the cost of omitting part of the stream. By default this option is set to false, so in such cases the encoder will be blocked until the muxer processes some of the packets and none of them is lost.

fifo_format format_name

Specify the format name. Useful if it cannot be guessed from the output name suffix.

format_opts options

Specify format options for the underlying muxer. Muxer options can be specified as a list of key=value pairs separated by ’:’.

max_recovery_attempts count

Set maximum number of successive unsuccessful recovery attempts after which the output fails permanently. By default this option is set to 0 (unlimited).

queue_size size

Specify size of the queue as a number of packets. Default value is 60.

recover_any_error bool

If set to true, recovery will be attempted regardless of type of the error causing the failure. By default this option is set to false and in case of certain (usually permanent) errors the recovery is not attempted even when the attempt_recovery option is set to true.

recovery_wait_streamtime bool

If set to false, the real time is used when waiting for the recovery attempt (i.e. the recovery will be attempted after the time specified by the recovery_wait_time option).

If set to true, the time of the processed stream is taken into account instead (i.e. the recovery will be attempted after discarding the packets corresponding to the recovery_wait_time option).

By default this option is set to false.

recovery_wait_time duration

Specify waiting time in seconds before the next recovery attempt after previous unsuccessful recovery attempt. Default value is 5.

restart_with_keyframe bool

Specify whether to wait for the keyframe after recovering from queue overflow or failure. This option is set to false by default.

timeshift duration

Buffer the specified amount of packets and delay writing the output. Note that the value of the queue_size option must be big enough to store the packets for timeshift. At the end of the input the fifo buffer is flushed at realtime speed.

21.33.2 Example

Use ffmpeg to stream to an RTMP server, continue processing the stream at real-time rate even in case of temporary failure (network outage) and attempt to recover streaming every second indefinitely:

ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv \
  -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 \
  -map 0:v -map 0:a rtmp://example.com/live/stream_name

21.34 film_cpk

Sega film (.cpk) muxer.

This format was used as internal format for several Sega games.

For more information regarding the Sega film file format, visit http://wiki.multimedia.cx/index.php?title=Sega_FILM.

It accepts at maximum one ‘cinepak’ or raw video stream, and at maximum one audio stream.

21.35 filmstrip

Adobe Filmstrip muxer.

This format is used by several Adobe tools to store a generated filmstrip export. It accepts a single raw video stream.

21.36 fits

Flexible Image Transport System (FITS) muxer.

This image format is used to store astronomical data.

For more information regarding the format, visit https://fits.gsfc.nasa.gov.

21.37 flac

Raw FLAC audio muxer.

This muxer accepts exactly one FLAC audio stream. Additionally, it is possible to add images with disposition ‘attached_pic’.

21.37.1 Options

write_header bool

write the file header if set to true, default is true

21.37.2 Example

Use ffmpeg to store the audio stream from an input file, together with several pictures used with ‘attached_pic’ disposition:

ffmpeg -i INPUT -i pic1.png -i pic2.jpg -map 0:a -map 1 -map 2 -disposition:v attached_pic OUTPUT

21.38 flv

Adobe Flash Video Format muxer.

21.38.1 Options

flvflags flags

Possible values:

aac_seq_header_detect

Place AAC sequence header based on audio stream data.

no_sequence_end

Disable sequence end tag.

no_metadata

Disable metadata tag.

no_duration_filesize

Disable duration and filesize in metadata when they are equal to zero at the end of stream. (Be used to non-seekable living stream).

add_keyframe_index

Used to facilitate seeking; particularly for HTTP pseudo streaming.

21.39 framecrc

Per-packet CRC (Cyclic Redundancy Check) testing format.

This muxer computes and prints the Adler-32 CRC for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.

The output of the muxer consists of a line for each audio and video packet of the form:

stream_index, packet_dts, packet_pts, packet_duration, packet_size, 0xCRC

CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.

21.39.1 Examples

For example to compute the CRC of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.crc:

ffmpeg -i INPUT -f framecrc out.crc

To print the information to stdout, use the command:

ffmpeg -i INPUT -f framecrc -

With ffmpeg, you can select the output format to which the audio and video frames are encoded before computing the CRC for each packet by specifying the audio and video codec. For example, to compute the CRC of each decoded input audio frame converted to PCM unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the command:

ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -

See also the crc muxer.

21.40 framehash

Per-packet hash testing format.

This muxer computes and prints a cryptographic hash for each audio and video packet. This can be used for packet-by-packet equality checks without having to individually do a binary comparison on each.

By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms.

The output of the muxer consists of a line for each audio and video packet of the form:

stream_index, packet_dts, packet_pts, packet_duration, packet_size, hash

hash is a hexadecimal number representing the computed hash for the packet.

hash algorithm

Use the cryptographic hash function specified by the string algorithm. Supported values include MD5, murmur3, RIPEMD128, RIPEMD160, RIPEMD256, RIPEMD320, SHA160, SHA224, SHA256 (default), SHA512/224, SHA512/256, SHA384, SHA512, CRC32 and adler32.

21.40.1 Examples

To compute the SHA-256 hash of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.sha256:

ffmpeg -i INPUT -f framehash out.sha256

To print the information to stdout, using the MD5 hash function, use the command:

ffmpeg -i INPUT -f framehash -hash md5 -

See also the hash muxer.

21.41 framemd5

Per-packet MD5 testing format.

This is a variant of the framehash muxer. Unlike that muxer, it defaults to using the MD5 hash function.

21.41.1 Examples

To compute the MD5 hash of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.md5:

ffmpeg -i INPUT -f framemd5 out.md5

To print the information to stdout, use the command:

ffmpeg -i INPUT -f framemd5 -

See also the framehash and md5 muxers.

21.42 gif

Animated GIF muxer.

Note that the GIF format has a very large time base: the delay between two frames can therefore not be smaller than one centi second.

21.42.1 Options

loop bool

Set the number of times to loop the output. Use -1 for no loop, 0 for looping indefinitely (default).

final_delay delay

Force the delay (expressed in centiseconds) after the last frame. Each frame ends with a delay until the next frame. The default is -1, which is a special value to tell the muxer to re-use the previous delay. In case of a loop, you might want to customize this value to mark a pause for instance.

21.42.2 Example

Encode a gif looping 10 times, with a 5 seconds delay between the loops:

ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif

Note 1: if you wish to extract the frames into separate GIF files, you need to force the image2 muxer:

ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"

21.43 gxf

General eXchange Format (GXF) muxer.

GXF was developed by Grass Valley Group, then standardized by SMPTE as SMPTE 360M and was extended in SMPTE RDD 14-2007 to include high-definition video resolutions.

It accepts at most one video stream with codec ‘mjpeg’, or ‘mpeg1video’, or ‘mpeg2video’, or ‘dvvideo’ with resolution ‘512x480’ or ‘608x576’, and several audio streams with rate 48000Hz and codec ‘pcm16_le’.

21.44 hash

Hash testing format.

This muxer computes and prints a cryptographic hash of all the input audio and video frames. This can be used for equality checks without having to do a complete binary comparison.

By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. Timestamps are ignored. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms.

The output of the muxer consists of a single line of the form: algo=hash, where algo is a short string representing the hash function used, and hash is a hexadecimal number representing the computed hash.

hash algorithm

Use the cryptographic hash function specified by the string algorithm. Supported values include MD5, murmur3, RIPEMD128, RIPEMD160, RIPEMD256, RIPEMD320, SHA160, SHA224, SHA256 (default), SHA512/224, SHA512/256, SHA384, SHA512, CRC32 and adler32.

21.44.1 Examples

To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file out.sha256:

ffmpeg -i INPUT -f hash out.sha256

To print an MD5 hash to stdout use the command:

ffmpeg -i INPUT -f hash -hash md5 -

See also the framehash muxer.

21.45 hds

HTTP Dynamic Streaming (HDS) muxer.

HTTP dynamic streaming, or HDS, is an adaptive bitrate streaming method developed by Adobe. HDS delivers MP4 video content over HTTP connections. HDS can be used for on-demand streaming or live streaming.

This muxer creates an .f4m (Adobe Flash Media Manifest File) manifest, an .abst (Adobe Bootstrap File) for each stream, and segment files in a directory specified as the output.

These needs to be accessed by an HDS player throuhg HTTPS for it to be able to perform playback on the generated stream.

21.45.1 Options

extra_window_size int

number of fragments kept outside of the manifest before removing from disk

min_frag_duration microseconds

minimum fragment duration (in microseconds), default value is 1 second (10000000)

remove_at_exit bool

remove all fragments when finished when set to true

window_size int

number of fragments kept in the manifest, if set to a value different from 0. By default all segments are kept in the output directory.

21.45.2 Example

Use ffmpeg to generate HDS files to the output.hds directory in real-time rate:

ffmpeg -re -i INPUT -f hds -b:v 200k output.hds

21.46 hls

Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming (HLS) specification.

It creates a playlist file, and one or more segment files. The output filename specifies the playlist filename.

By default, the muxer creates a file for each segment produced. These files have the same name as the playlist, followed by a sequential number and a .ts extension.

Make sure to require a closed GOP when encoding and to set the GOP size to fit your segment time constraint.

For example, to convert an input file with ffmpeg:

ffmpeg -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 out.m3u8

This example will produce the playlist, out.m3u8, and segment files: out0.ts, out1.ts, out2.ts, etc.

See also the segment muxer, which provides a more generic and flexible implementation of a segmenter, and can be used to perform HLS segmentation.

21.46.1 Options

hls_init_time duration

Set the initial target segment length. Default value is 0.

duration must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.

Segment will be cut on the next key frame after this time has passed on the first m3u8 list. After the initial playlist is filled, ffmpeg will cut segments at duration equal to hls_time.

hls_time duration

Set the target segment length. Default value is 2.

duration must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual. Segment will be cut on the next key frame after this time has passed.

hls_list_size size

Set the maximum number of playlist entries. If set to 0 the list file will contain all the segments. Default value is 5.

hls_delete_threshold size

Set the number of unreferenced segments to keep on disk before hls_flags delete_segments deletes them. Increase this to allow continue clients to download segments which were recently referenced in the playlist. Default value is 1, meaning segments older than hls_list_size+1 will be deleted.

hls_start_number_source source

Start the playlist sequence number (#EXT-X-MEDIA-SEQUENCE) according to the specified source. Unless hls_flags single_file is set, it also specifies source of starting sequence numbers of segment and subtitle filenames. In any case, if hls_flags append_list is set and read playlist sequence number is greater than the specified start sequence number, then that value will be used as start value.

It accepts the following values:

generic (default)

Set the start numbers according to the start_number option value.

epoch

Set the start number as the seconds since epoch (1970-01-01 00:00:00).

epoch_us

Set the start number as the microseconds since epoch (1970-01-01 00:00:00).

datetime

Set the start number based on the current date/time as YYYYmmddHHMMSS. e.g. 20161231235759.

start_number number

Start the playlist sequence number (#EXT-X-MEDIA-SEQUENCE) from the specified number when hls_start_number_source value is generic. (This is the default case.) Unless hls_flags single_file is set, it also specifies starting sequence numbers of segment and subtitle filenames. Default value is 0.

hls_allow_cache bool

Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments.

hls_base_url baseurl

Append baseurl to every entry in the playlist. Useful to generate playlists with absolute paths.

Note that the playlist sequence number must be unique for each segment and it is not to be confused with the segment filename sequence number which can be cyclic, for example if the wrap option is specified.

hls_segment_filename filename

Set the segment filename. Unless the hls_flags option is set with ‘single_file’, filename is used as a string format with the segment number appended.

For example:

ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8

will produce the playlist, out.m3u8, and segment files: file000.ts, file001.ts, file002.ts, etc.

filename may contain a full path or relative path specification, but only the file name part without any path will be contained in the m3u8 segment list. Should a relative path be specified, the path of the created segment files will be relative to the current working directory. When strftime_mkdir is set, the whole expanded value of filename will be written into the m3u8 segment list.

When var_stream_map is set with two or more variant streams, the filename pattern must contain the string "%v", and this string will be expanded to the position of variant stream index in the generated segment file names.

For example:

ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
  -hls_segment_filename 'file_%v_%03d.ts' out_%v.m3u8

will produce the playlists segment file sets: file_0_000.ts, file_0_001.ts, file_0_002.ts, etc. and file_1_000.ts, file_1_001.ts, file_1_002.ts, etc.

The string "%v" may be present in the filename or in the last directory name containing the file, but only in one of them. (Additionally, %v may appear multiple times in the last sub-directory or filename.) If the string %v is present in the directory name, then sub-directories are created after expanding the directory name pattern. This enables creation of segments corresponding to different variant streams in subdirectories.

For example:

ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
  -hls_segment_filename 'vs%v/file_%03d.ts' vs%v/out.m3u8

will produce the playlists segment file sets: vs0/file_000.ts, vs0/file_001.ts, vs0/file_002.ts, etc. and vs1/file_000.ts, vs1/file_001.ts, vs1/file_002.ts, etc.

strftime bool

Use strftime() on filename to expand the segment filename with localtime. The segment number is also available in this mode, but to use it, you need to set ‘second_level_segment_index’ in the hls_flag and %%d will be the specifier.

For example:

ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8

will produce the playlist, out.m3u8, and segment files: file-20160215-1455569023.ts, file-20160215-1455569024.ts, etc. Note: On some systems/environments, the %s specifier is not available. See strftime() documentation.

For example:

ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8

will produce the playlist, out.m3u8, and segment files: file-20160215-0001.ts, file-20160215-0002.ts, etc.

strftime_mkdir bool

Used together with strftime, it will create all subdirectories which are present in the expanded values of option hls_segment_filename.

For example:

ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8

will create a directory 201560215 (if it does not exist), and then produce the playlist, out.m3u8, and segment files: 20160215/file-20160215-1455569023.ts, 20160215/file-20160215-1455569024.ts, etc.

For example:

ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8

will create a directory hierarchy 2016/02/15 (if any of them do not exist), and then produce the playlist, out.m3u8, and segment files: 2016/02/15/file-20160215-1455569023.ts, 2016/02/15/file-20160215-1455569024.ts, etc.

hls_segment_options options_list

Set output format options using a :-separated list of key=value parameters. Values containing : special characters must be escaped.

hls_key_info_file key_info_file

Use the information in key_info_file for segment encryption. The first line of key_info_file specifies the key URI written to the playlist. The key URL is used to access the encryption key during playback. The second line specifies the path to the key file used to obtain the key during the encryption process. The key file is read as a single packed array of 16 octets in binary format. The optional third line specifies the initialization vector (IV) as a hexadecimal string to be used instead of the segment sequence number (default) for encryption. Changes to key_info_file will result in segment encryption with the new key/IV and an entry in the playlist for the new key URI/IV if hls_flags periodic_rekey is enabled.

Key info file format:

key URI
key file path
IV (optional)

Example key URIs:

http://server/file.key
/path/to/file.key
file.key

Example key file paths:

file.key
/path/to/file.key

Example IV:

0123456789ABCDEF0123456789ABCDEF

Key info file example:

http://server/file.key
/path/to/file.key
0123456789ABCDEF0123456789ABCDEF

Example shell script:

#!/bin/sh
BASE_URL=${1:-'.'}
openssl rand 16 > file.key
echo $BASE_URL/file.key > file.keyinfo
echo file.key >> file.keyinfo
echo $(openssl rand -hex 16) >> file.keyinfo
ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
  -hls_key_info_file file.keyinfo out.m3u8
hls_enc bool

Enable (1) or disable (0) the AES128 encryption. When enabled every segment generated is encrypted and the encryption key is saved as playlist name.key.

hls_enc_key key

Specify a 16-octet key to encrypt the segments, by default it is randomly generated.

hls_enc_key_url keyurl

If set, keyurl is prepended instead of baseurl to the key filename in the playlist.

hls_enc_iv iv

Specify the 16-octet initialization vector for every segment instead of the autogenerated ones.

hls_segment_type flags

Possible values:

mpegts

Output segment files in MPEG-2 Transport Stream format. This is compatible with all HLS versions.

fmp4

Output segment files in fragmented MP4 format, similar to MPEG-DASH. fmp4 files may be used in HLS version 7 and above.

hls_fmp4_init_filename filename

Set filename for the fragment files header file, default filename is init.mp4.

When strftime is enabled, filename is expanded to the segment filename with localtime.

For example:

ffmpeg -i in.nut -hls_segment_type fmp4 -strftime 1 -hls_fmp4_init_filename "%s_init.mp4" out.m3u8

will produce init like this 1602678741_init.mp4.

hls_fmp4_init_resend bool

Resend init file after m3u8 file refresh every time, default is 0.

When var_stream_map is set with two or more variant streams, the filename pattern must contain the string "%v", this string specifies the position of variant stream index in the generated init file names. The string "%v" may be present in the filename or in the last directory name containing the file. If the string is present in the directory name, then sub-directories are created after expanding the directory name pattern. This enables creation of init files corresponding to different variant streams in subdirectories.

hls_flags flags

Possible values:

single_file

If this flag is set, the muxer will store all segments in a single MPEG-TS file, and will use byte ranges in the playlist. HLS playlists generated with this way will have the version number 4.

For example:

ffmpeg -i in.nut -hls_flags single_file out.m3u8

will produce the playlist, out.m3u8, and a single segment file, out.ts.

delete_segments

Segment files removed from the playlist are deleted after a period of time equal to the duration of the segment plus the duration of the playlist.

append_list

Append new segments into the end of old segment list, and remove the #EXT-X-ENDLIST from the old segment list.

round_durations

Round the duration info in the playlist file segment info to integer values, instead of using floating point. If there are no other features requiring higher HLS versions be used, then this will allow ffmpeg to output a HLS version 2 m3u8.

discont_start

Add the #EXT-X-DISCONTINUITY tag to the playlist, before the first segment’s information.

omit_endlist

Do not append the EXT-X-ENDLIST tag at the end of the playlist.

periodic_rekey

The file specified by hls_key_info_file will be checked periodically and detect updates to the encryption info. Be sure to replace this file atomically, including the file containing the AES encryption key.

independent_segments

Add the #EXT-X-INDEPENDENT-SEGMENTS tag to playlists that has video segments and when all the segments of that playlist are guaranteed to start with a key frame.

iframes_only

Add the #EXT-X-I-FRAMES-ONLY tag to playlists that has video segments and can play only I-frames in the #EXT-X-BYTERANGE mode.

split_by_time

Allow segments to start on frames other than key frames. This improves behavior on some players when the time between key frames is inconsistent, but may make things worse on others, and can cause some oddities during seeking. This flag should be used with the hls_time option.

program_date_time

Generate EXT-X-PROGRAM-DATE-TIME tags.

second_level_segment_index

Make it possible to use segment indexes as %%d in the hls_segment_filename option expression besides date/time values when strftime option is on. To get fixed width numbers with trailing zeroes, %%0xd format is available where x is the required width.

second_level_segment_size

Make it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename option expression besides date/time values when strftime is on. To get fixed width numbers with trailing zeroes, %%0xs format is available where x is the required width.

second_level_segment_duration

Make it possible to use segment duration (calculated in microseconds) as %%t in hls_segment_filename option expression besides date/time values when strftime is on. To get fixed width numbers with trailing zeroes, %%0xt format is available where x is the required width.

For example:

ffmpeg -i sample.mpeg \
   -f hls -hls_time 3 -hls_list_size 5 \
   -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
   -strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8

will produce segments like this: segment_20170102194334_0003_00122200_0000003000000.ts, segment_20170102194334_0004_00120072_0000003000000.ts etc.

temp_file

Write segment data to filename.tmp and rename to filename only once the segment is complete.

A webserver serving up segments can be configured to reject requests to *.tmp to prevent access to in-progress segments before they have been added to the m3u8 playlist.

This flag also affects how m3u8 playlist files are created. If this flag is set, all playlist files will be written into a temporary file and renamed after they are complete, similarly as segments are handled. But playlists with file protocol and with hls_playlist_type type other than ‘vod’ are always written into a temporary file regardless of this flag.

Master playlist files specified with master_pl_name, if any, with file protocol, are always written into temporary file regardless of this flag if master_pl_publish_rate value is other than zero.

hls_playlist_type type

If type is ‘event’, emit #EXT-X-PLAYLIST-TYPE:EVENT in the m3u8 header. This forces hls_list_size to 0; the playlist can only be appended to.

If type is ‘vod’, emit #EXT-X-PLAYLIST-TYPE:VOD in the m3u8 header. This forces hls_list_size to 0; the playlist must not change.

method method

Use the given HTTP method to create the hls files.

For example:

ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8

will upload all the mpegts segment files to the HTTP server using the HTTP PUT method, and update the m3u8 files every refresh times using the same method. Note that the HTTP server must support the given method for uploading files.

http_user_agent agent

Override User-Agent field in HTTP header. Applicable only for HTTP output.

var_stream_map stream_map

Specify a map string defining how to group the audio, video and subtitle streams into different variant streams. The variant stream groups are separated by space.

Expected string format is like this "a:0,v:0 a:1,v:1 ....". Here a:, v:, s: are the keys to specify audio, video and subtitle streams respectively. Allowed values are 0 to 9 (limited just based on practical usage).

When there are two or more variant streams, the output filename pattern must contain the string "%v": this string specifies the position of variant stream index in the output media playlist filenames. The string "%v" may be present in the filename or in the last directory name containing the file. If the string is present in the directory name, then sub-directories are created after expanding the directory name pattern. This enables creation of variant streams in subdirectories.

A few examples follow.

  • Create two hls variant streams. The first variant stream will contain video stream of bitrate 1000k and audio stream of bitrate 64k and the second variant stream will contain video stream of bitrate 256k and audio stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and out_1.m3u8 will be created.
    ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
      -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
      http://example.com/live/out_%v.m3u8
    
  • If you want something meaningful text instead of indexes in result names, you may specify names for each or some of the variants. The following example will create two hls variant streams as in the previous one. But here, the two media playlist with file names out_my_hd.m3u8 and out_my_sd.m3u8 will be created.
    ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
      -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0,name:my_hd v:1,a:1,name:my_sd" \
      http://example.com/live/out_%v.m3u8
    
  • Create three hls variant streams. The first variant stream will be a video only stream with video bitrate 1000k, the second variant stream will be an audio only stream with bitrate 64k and the third variant stream will be a video only stream with bitrate 256k. Here, three media playlist with file names out_0.m3u8, out_1.m3u8 and out_2.m3u8 will be created.
    ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \
      -map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \
      http://example.com/live/out_%v.m3u8
    
  • Create the variant streams in subdirectories. Here, the first media playlist is created at http://example.com/live/vs_0/out.m3u8 and the second one at http://example.com/live/vs_1/out.m3u8.
    ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
      -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
      http://example.com/live/vs_%v/out.m3u8
    
  • Create two audio only and two video only variant streams. In addition to the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, the #EXT-X-MEDIA tag is also added for the two audio only variant streams and they are mapped to the two video only variant streams with audio group names ’aud_low’ and ’aud_high’. By default, a single hls variant containing all the encoded streams is created.
    ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k -b:v:1 3000k  \
      -map 0:a -map 0:a -map 0:v -map 0:v -f hls \
      -var_stream_map "a:0,agroup:aud_low a:1,agroup:aud_high v:0,agroup:aud_low v:1,agroup:aud_high" \
      -master_pl_name master.m3u8 \
      http://example.com/live/out_%v.m3u8
    
  • Create two audio only and one video only variant streams. In addition to the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, the #EXT-X-MEDIA tag is also added for the two audio only variant streams and they are mapped to the one video only variant streams with audio group name ’aud_low’, and the audio group have default stat is NO or YES. By default, a single hls variant containing all the encoded streams is created.
    ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
      -map 0:a -map 0:a -map 0:v -f hls \
      -var_stream_map "a:0,agroup:aud_low,default:yes a:1,agroup:aud_low v:0,agroup:aud_low" \
      -master_pl_name master.m3u8 \
      http://example.com/live/out_%v.m3u8
    
  • Create two audio only and one video only variant streams. In addition to the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, the #EXT-X-MEDIA tag is also added for the two audio only variant streams and they are mapped to the one video only variant streams with audio group name ’aud_low’, and the audio group have default stat is NO or YES, and one audio have and language is named ENG, the other audio language is named CHN. By default, a single hls variant containing all the encoded streams is created.
    ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
      -map 0:a -map 0:a -map 0:v -f hls \
      -var_stream_map "a:0,agroup:aud_low,default:yes,language:ENG a:1,agroup:aud_low,language:CHN v:0,agroup:aud_low" \
      -master_pl_name master.m3u8 \
      http://example.com/live/out_%v.m3u8
    
  • Create a single variant stream. Add the #EXT-X-MEDIA tag with TYPE=SUBTITLES in the master playlist with webvtt subtitle group name ’subtitle’ and optional subtitle name, e.g. ’English’. Make sure the input file has one text subtitle stream at least.
    ffmpeg -y -i input_with_subtitle.mkv \
     -b:v:0 5250k -c:v h264 -pix_fmt yuv420p -profile:v main -level 4.1 \
     -b:a:0 256k \
     -c:s webvtt -c:a mp2 -ar 48000 -ac 2 -map 0:v -map 0:a:0 -map 0:s:0 \
     -f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle,sname:English" \
     -master_pl_name master.m3u8 -t 300 -hls_time 10 -hls_init_time 4 -hls_list_size \
     10 -master_pl_publish_rate 10 -hls_flags \
     delete_segments+discont_start+split_by_time ./tmp/video.m3u8
    
cc_stream_map cc_stream_map

Map string which specifies different closed captions groups and their attributes. The closed captions stream groups are separated by space.

Expected string format is like this "ccgroup:<group name>,instreamid:<INSTREAM-ID>,language:<language code> ....". ’ccgroup’ and ’instreamid’ are mandatory attributes. ’language’ is an optional attribute.

The closed captions groups configured using this option are mapped to different variant streams by providing the same ’ccgroup’ name in the var_stream_map string.

For example:

ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
  -a53cc:0 1 -a53cc:1 1 \
  -map 0:v -map 0:a -map 0:v -map 0:a -f hls \
  -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en ccgroup:cc,instreamid:CC2,language:sp" \
  -var_stream_map "v:0,a:0,ccgroup:cc v:1,a:1,ccgroup:cc" \
  -master_pl_name master.m3u8 \
  http://example.com/live/out_%v.m3u8

will add two #EXT-X-MEDIA tags with TYPE=CLOSED-CAPTIONS in the master playlist for the INSTREAM-IDs ’CC1’ and ’CC2’. Also, it will add CLOSED-CAPTIONS attribute with group name ’cc’ for the two output variant streams.

If var_stream_map is not set, then the first available ccgroup in cc_stream_map is mapped to the output variant stream.

For example:

ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \
  -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en" \
  -master_pl_name master.m3u8 \
  http://example.com/live/out.m3u8

this will add #EXT-X-MEDIA tag with TYPE=CLOSED-CAPTIONS in the master playlist with group name ’cc’, language ’en’ (english) and INSTREAM-ID ’CC1’. Also, it will add CLOSED-CAPTIONS attribute with group name ’cc’ for the output variant stream.

master_pl_name name

Create HLS master playlist with the given name.

For example:

ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8

creates an HLS master playlist with name master.m3u8 which is published at http://example.com/live/.

master_pl_publish_rate count

Publish master play list repeatedly every after specified number of segment intervals.

For example:

ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 \
-hls_time 2 -master_pl_publish_rate 30 http://example.com/live/out.m3u8

creates an HLS master playlist with name master.m3u8 and keeps publishing it repeatedly every after 30 segments i.e. every after 60s.

http_persistent bool

Use persistent HTTP connections. Applicable only for HTTP output.

timeout timeout

Set timeout for socket I/O operations. Applicable only for HTTP output.

ignore_io_errors bool

Ignore IO errors during open, write and delete. Useful for long-duration runs with network output.

headers headers

Set custom HTTP headers, can override built in default headers. Applicable only for HTTP output.

21.47 iamf

Immersive Audio Model and Formats (IAMF) muxer.

IAMF is used to provide immersive audio content for presentation on a wide range of devices in both streaming and offline applications. These applications include internet audio streaming, multicasting/broadcasting services, file download, gaming, communication, virtual and augmented reality, and others. In these applications, audio may be played back on a wide range of devices, e.g., headphones, mobile phones, tablets, TVs, sound bars, home theater systems, and big screens.

This format was promoted and desgined by Alliance for Open Media.

For more information about this format, see https://aomedia.org/iamf/.

21.48 ico

ICO file muxer.

Microsoft’s icon file format (ICO) has some strict limitations that should be noted:

  • Size cannot exceed 256 pixels in any dimension
  • Only BMP and PNG images can be stored
  • If a BMP image is used, it must be one of the following pixel formats:
    BMP Bit Depth      FFmpeg Pixel Format
    1bit               pal8
    4bit               pal8
    8bit               pal8
    16bit              rgb555le
    24bit              bgr24
    32bit              bgra
    
  • If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
  • If a PNG image is used, it must use the rgba pixel format

21.49 ilbc

Internet Low Bitrate Codec (iLBC) raw muxer.

It accepts a single ‘ilbc’ audio stream.

21.50 image2, image2pipe

Image file muxer.

The ‘image2’ muxer writes video frames to image files.

The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string "%d" or "%0Nd", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0Nd" is used, the string representing the number in each filename is 0-padded to N digits. The literal character ’%’ can be specified in the pattern with the string "%%".

If the pattern contains "%d" or "%0Nd", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential.

The pattern may contain a suffix which is used to automatically determine the format of the image files to write.

For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc. The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form img%-1.jpg, img%-2.jpg, ..., img%-10.jpg, etc.

The image muxer supports the .Y.U.V image file format. This format is special in that each image frame consists of three files, for each of the YUV420P components. To read or write this image file format, specify the name of the ’.Y’ file. The muxer will automatically open the ’.U’ and ’.V’ files as required.

The ‘image2pipe’ muxer accepts the same options as the ‘image2’ muxer, but ignores the pattern verification and expansion, as it is supposed to write to the command output rather than to an actual stored file.

21.50.1 Options

frame_pts bool

If set to 1, expand the filename with the packet PTS (presentation time stamp). Default value is 0.

start_number count

Start the sequence from the specified number. Default value is 1.

update bool

If set to 1, the filename will always be interpreted as just a filename, not a pattern, and the corresponding file will be continuously overwritten with new images. Default value is 0.

strftime bool

If set to 1, expand the filename with date and time information from strftime(). Default value is 0.

atomic_writing bool

Write output to a temporary file, which is renamed to target filename once writing is completed. Default is disabled.

protocol_opts options_list

Set protocol options as a :-separated list of key=value parameters. Values containing the : special character must be escaped.

21.50.2 Examples

  • Use ffmpeg for creating a sequence of files img-001.jpeg, img-002.jpeg, ..., taking one image every second from the input video:
    ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg'
    

    Note that with ffmpeg, if the format is not specified with the -f option and the output filename specifies an image file format, the image2 muxer is automatically selected, so the previous command can be written as:

    ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg'
    

    Note also that the pattern must not necessarily contain "%d" or "%0Nd", for example to create a single image file img.jpeg from the start of the input video you can employ the command:

    ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
    
  • The strftime option allows you to expand the filename with date and time information. Check the documentation of the strftime() function for the syntax.

    To generate image files from the strftime() "%Y-%m-%d_%H-%M-%S" pattern, the following ffmpeg command can be used:

    ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
    
  • Set the file name with current frame’s PTS:
    ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg
    
  • Publish contents of your desktop directly to a WebDAV server every second:
    ffmpeg -f x11grab -framerate 1 -i :0.0 -q:v 6 -update 1 -protocol_opts method=PUT http://example.com/desktop.jpg
    

21.51 ircam

Berkeley / IRCAM / CARL Sound Filesystem (BICSF) format muxer.

The Berkeley/IRCAM/CARL Sound Format, developed in the 1980s, is a result of the merging of several different earlier sound file formats and systems including the csound system developed by Dr Gareth Loy at the Computer Audio Research Lab (CARL) at UC San Diego, the IRCAM sound file system developed by Rob Gross and Dan Timis at the Institut de Recherche et Coordination Acoustique / Musique in Paris and the Berkeley Fast Filesystem.

It was developed initially as part of the Berkeley/IRCAM/CARL Sound Filesystem, a suite of programs designed to implement a filesystem for audio applications running under Berkeley UNIX. It was particularly popular in academic music research centres, and was used a number of times in the creation of early computer-generated compositions.

This muxer accepts a single audio stream containing PCM data.

21.52 ivf

On2 IVF muxer.

IVF was developed by On2 Technologies (formerly known as Duck Corporation), to store internally developed codecs.

This muxer accepts a single ‘vp8’, ‘vp9’, or ‘av1’ video stream.

21.53 jacosub

JACOsub subtitle format muxer.

This muxer accepts a single ‘jacosub’ subtitles stream.

For more information about the format, see http://unicorn.us.com/jacosub/jscripts.html.

21.54 kvag

Simon & Schuster Interactive VAG muxer.

This custom VAG container is used by some Simon & Schuster Interactive games such as "Real War", and "Real War: Rogue States".

This muxer accepts a single ‘adpcm_ima_ssi’ audio stream.

21.55 lc3

Bluetooth SIG Low Complexity Communication Codec audio (LC3), or ETSI TS 103 634 Low Complexity Communication Codec plus (LC3plus).

This muxer accepts a single ‘lc3’ audio stream.

21.56 lrc

LRC lyrics file format muxer.

LRC (short for LyRiCs) is a computer file format that synchronizes song lyrics with an audio file, such as MP3, Vorbis, or MIDI.

This muxer accepts a single ‘subrip’ or ‘text’ subtitles stream.

21.56.1 Metadata

The following metadata tags are converted to the format corresponding metadata:

title
album
artist
author
creator
encoder
encoder_version

If ‘encoder_version’ is not explicitly set, it is automatically set to the libavformat version.

21.57 matroska

Matroska container muxer.

This muxer implements the matroska and webm container specs.

21.57.1 Metadata

The recognized metadata settings in this muxer are:

title

Set title name provided to a single track. This gets mapped to the FileDescription element for a stream written as attachment.

language

Specify the language of the track in the Matroska languages form.

The language can be either the 3 letters bibliographic ISO-639-2 (ISO 639-2/B) form (like "fre" for French), or a language code mixed with a country code for specialities in languages (like "fre-ca" for Canadian French).

stereo_mode

Set stereo 3D video layout of two views in a single video track.

The following values are recognized:

mono

video is not stereo

left_right

Both views are arranged side by side, Left-eye view is on the left

bottom_top

Both views are arranged in top-bottom orientation, Left-eye view is at bottom

top_bottom

Both views are arranged in top-bottom orientation, Left-eye view is on top

checkerboard_rl

Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first

checkerboard_lr

Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first

row_interleaved_rl

Each view is constituted by a row based interleaving, Right-eye view is first row

row_interleaved_lr

Each view is constituted by a row based interleaving, Left-eye view is first row

col_interleaved_rl

Both views are arranged in a column based interleaving manner, Right-eye view is first column

col_interleaved_lr

Both views are arranged in a column based interleaving manner, Left-eye view is first column

anaglyph_cyan_red

All frames are in anaglyph format viewable through red-cyan filters

right_left

Both views are arranged side by side, Right-eye view is on the left

anaglyph_green_magenta

All frames are in anaglyph format viewable through green-magenta filters

block_lr

Both eyes laced in one Block, Left-eye view is first

block_rl

Both eyes laced in one Block, Right-eye view is first

For example a 3D WebM clip can be created using the following command line:

ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm

21.57.2 Options

reserve_index_space size

By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space to leave for the index at the beginning of the file. However for some use cases – e.g. streaming where seeking is possible but slow – it is useful to put the index at the beginning of the file.

If this option is set to a non-zero value, the muxer will reserve size bytes of space in the file header and then try to write the cues there when the muxing finishes. If the reserved space does not suffice, no Cues will be written, the file will be finalized and writing the trailer will return an error. A safe size for most use cases should be about 50kB per hour of video.

Note that cues are only written if the output is seekable and this option will have no effect if it is not.

cues_to_front bool

If set, the muxer will write the index at the beginning of the file by shifting the main data if necessary. This can be combined with reserve_index_space in which case the data is only shifted if the initially reserved space turns out to be insufficient.

This option is ignored if the output is unseekable.

cluster_size_limit size

Store at most the provided amount of bytes in a cluster.

If not specified, the limit is set automatically to a sensible hardcoded fixed value.

cluster_time_limit duration

Store at most the provided number of milliseconds in a cluster.

If not specified, the limit is set automatically to a sensible hardcoded fixed value.

dash bool

Create a WebM file conforming to WebM DASH specification. By default it is set to false.

dash_track_number index

Track number for the DASH stream. By default it is set to 1.

live bool

Write files assuming it is a live stream. By default it is set to false.

allow_raw_vfw bool

Allow raw VFW mode. By default it is set to false.

flipped_raw_rgb bool

If set to true, store positive height for raw RGB bitmaps, which indicates bitmap is stored bottom-up. Note that this option does not flip the bitmap which has to be done manually beforehand, e.g. by using the ‘vflip’ filter. Default is false and indicates bitmap is stored top down.

write_crc32 bool

Write a CRC32 element inside every Level 1 element. By default it is set to true. This option is ignored for WebM.

default_mode mode

Control how the FlagDefault of the output tracks will be set. It influences which tracks players should play by default. The default mode is ‘passthrough’.

infer

Every track with disposition default will have the FlagDefault set. Additionally, for each type of track (audio, video or subtitle), if no track with disposition default of this type exists, then the first track of this type will be marked as default (if existing). This ensures that the default flag is set in a sensible way even if the input originated from containers that lack the concept of default tracks.

infer_no_subs

This mode is the same as infer except that if no subtitle track with disposition default exists, no subtitle track will be marked as default.

passthrough

In this mode the FlagDefault is set if and only if the AV_DISPOSITION_DEFAULT flag is set in the disposition of the corresponding stream.

21.58 md5

MD5 testing format.

This is a variant of the hash muxer. Unlike that muxer, it defaults to using the MD5 hash function.

See also the hash and framemd5 muxers.

21.58.1 Examples

  • To compute the MD5 hash of the input converted to raw audio and video, and store it in the file out.md5:
    ffmpeg -i INPUT -f md5 out.md5
    
  • To print the MD5 hash to stdout:
    ffmpeg -i INPUT -f md5 -
    

21.59 microdvd

MicroDVD subtitle format muxer.

This muxer accepts a single ‘microdvd’ subtitles stream.

21.60 mmf

Synthetic music Mobile Application Format (SMAF) format muxer.

SMAF is a music data format specified by Yamaha for portable electronic devices, such as mobile phones and personal digital assistants.

This muxer accepts a single ‘adpcm_yamaha’ audio stream.

21.61 mp3

The MP3 muxer writes a raw MP3 stream with the following optional features:

  • An ID3v2 metadata header at the beginning (enabled by default). Versions 2.3 and 2.4 are supported, the id3v2_version private option controls which one is used (3 or 4). Setting id3v2_version to 0 disables the ID3v2 header completely.

    The muxer supports writing attached pictures (APIC frames) to the ID3v2 header. The pictures are supplied to the muxer in form of a video stream with a single packet. There can be any number of those streams, each will correspond to a single APIC frame. The stream metadata tags title and comment map to APIC description and picture type respectively. See http://id3.org/id3v2.4.0-frames for allowed picture types.

    Note that the APIC frames must be written at the beginning, so the muxer will buffer the audio frames until it gets all the pictures. It is therefore advised to provide the pictures as soon as possible to avoid excessive buffering.

  • A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by default, but will be written only if the output is seekable. The write_xing private option can be used to disable it. The frame contains various information that may be useful to the decoder, like the audio duration or encoder delay.
  • A legacy ID3v1 tag at the end of the file (disabled by default). It may be enabled with the write_id3v1 private option, but as its capabilities are very limited, its usage is not recommended.

Examples:

Write an mp3 with an ID3v2.3 header and an ID3v1 footer:

ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3

To attach a picture to an mp3 file select both the audio and the picture stream with map:

ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

Write a "clean" MP3 without any extra features:

ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3

21.62 mpegts

MPEG transport stream muxer.

This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

The recognized metadata settings in mpegts muxer are service_provider and service_name. If they are not set the default for service_provider is ‘FFmpeg’ and the default for service_name is ‘Service01’.

21.62.1 Options

The muxer options are:

mpegts_transport_stream_id integer

Set the ‘transport_stream_id’. This identifies a transponder in DVB. Default is 0x0001.

mpegts_original_network_id integer

Set the ‘original_network_id’. This is unique identifier of a network in DVB. Its main use is in the unique identification of a service through the path ‘Original_Network_ID, Transport_Stream_ID’. Default is 0x0001.

mpegts_service_id integer

Set the ‘service_id’, also known as program in DVB. Default is 0x0001.

mpegts_service_type integer

Set the program ‘service_type’. Default is digital_tv. Accepts the following options:

hex_value

Any hexadecimal value between 0x01 and 0xff as defined in ETSI 300 468.

digital_tv

Digital TV service.

digital_radio

Digital Radio service.

teletext

Teletext service.

advanced_codec_digital_radio

Advanced Codec Digital Radio service.

mpeg2_digital_hdtv

MPEG2 Digital HDTV service.

advanced_codec_digital_sdtv

Advanced Codec Digital SDTV service.

advanced_codec_digital_hdtv

Advanced Codec Digital HDTV service.

mpegts_pmt_start_pid integer

Set the first PID for PMTs. Default is 0x1000, minimum is 0x0020, maximum is 0x1ffa. This option has no effect in m2ts mode where the PMT PID is fixed 0x0100.

mpegts_start_pid integer

Set the first PID for elementary streams. Default is 0x0100, minimum is 0x0020, maximum is 0x1ffa. This option has no effect in m2ts mode where the elementary stream PIDs are fixed.

mpegts_m2ts_mode boolean

Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.

muxrate integer

Set a constant muxrate. Default is VBR.

pes_payload_size integer

Set minimum PES packet payload in bytes. Default is 2930.

mpegts_flags flags

Set mpegts flags. Accepts the following options:

resend_headers

Reemit PAT/PMT before writing the next packet.

latm

Use LATM packetization for AAC.

pat_pmt_at_frames

Reemit PAT and PMT at each video frame.

system_b

Conform to System B (DVB) instead of System A (ATSC).

initial_discontinuity

Mark the initial packet of each stream as discontinuity.

nit

Emit NIT table.

omit_rai

Disable writing of random access indicator.

mpegts_copyts boolean

Preserve original timestamps, if value is set to 1. Default value is -1, which results in shifting timestamps so that they start from 0.

omit_video_pes_length boolean

Omit the PES packet length for video packets. Default is 1 (true).

pcr_period integer

Override the default PCR retransmission time in milliseconds. Default is -1 which means that the PCR interval will be determined automatically: 20 ms is used for CBR streams, the highest multiple of the frame duration which is less than 100 ms is used for VBR streams.

pat_period duration

Maximum time in seconds between PAT/PMT tables. Default is 0.1.

sdt_period duration

Maximum time in seconds between SDT tables. Default is 0.5.

nit_period duration

Maximum time in seconds between NIT tables. Default is 0.5.

tables_version integer

Set PAT, PMT, SDT and NIT version (default 0, valid values are from 0 to 31, inclusively). This option allows updating stream structure so that standard consumer may detect the change. To do so, reopen output AVFormatContext (in case of API usage) or restart ffmpeg instance, cyclically changing tables_version value:

ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...
ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...

21.62.2 Example

ffmpeg -i file.mpg -c copy \
     -mpegts_original_network_id 0x1122 \
     -mpegts_transport_stream_id 0x3344 \
     -mpegts_service_id 0x5566 \
     -mpegts_pmt_start_pid 0x1500 \
     -mpegts_start_pid 0x150 \
     -metadata service_provider="Some provider" \
     -metadata service_name="Some Channel" \
     out.ts

21.63 mxf, mxf_d10, mxf_opatom

MXF muxer.

21.63.1 Options

The muxer options are:

store_user_comments bool

Set if user comments should be stored if available or never. IRT D-10 does not allow user comments. The default is thus to write them for mxf and mxf_opatom but not for mxf_d10

21.64 null

Null muxer.

This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes.

For example to benchmark decoding with ffmpeg you can use the command:

ffmpeg -benchmark -i INPUT -f null out.null

Note that the above command does not read or write the out.null file, but specifying the output file is required by the ffmpeg syntax.

Alternatively you can write the command as:

ffmpeg -benchmark -i INPUT -f null -

21.65 nut

-syncpoints flags

Change the syncpoint usage in nut:

default use the normal low-overhead seeking aids.
none do not use the syncpoints at all, reducing the overhead but making the stream non-seekable;

Use of this option is not recommended, as the resulting files are very damage sensitive and seeking is not possible. Also in general the overhead from syncpoints is negligible. Note, -write_index 0 can be used to disable all growing data tables, allowing to mux endless streams with limited memory and without these disadvantages.

timestamped extend the syncpoint with a wallclock field.

The none and timestamped flags are experimental.

-write_index bool

Write index at the end, the default is to write an index.

ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor

21.66 ogg

Ogg container muxer.

-page_duration duration

Preferred page duration, in microseconds. The muxer will attempt to create pages that are approximately duration microseconds long. This allows the user to compromise between seek granularity and container overhead. The default is 1 second. A value of 0 will fill all segments, making pages as large as possible. A value of 1 will effectively use 1 packet-per-page in most situations, giving a small seek granularity at the cost of additional container overhead.

-serial_offset value

Serial value from which to set the streams serial number. Setting it to different and sufficiently large values ensures that the produced ogg files can be safely chained.

21.67 rcwt

RCWT (Raw Captions With Time) is a format native to ccextractor, a commonly used open source tool for processing 608/708 Closed Captions (CC) sources. It can be used to archive the original extracted CC bitstream and to produce a source file for later processing or conversion. The format allows for interoperability between ccextractor and FFmpeg, is simple to parse, and can be used to create a backup of the CC presentation.

This muxer implements the specification as of March 2024, which has been stable and unchanged since April 2014.

This muxer will have some nuances from the way that ccextractor muxes RCWT. No compatibility issues when processing the output with ccextractor have been observed as a result of this so far, but mileage may vary and outputs will not be a bit-exact match.

A free specification of RCWT can be found here: https://github.com/CCExtractor/ccextractor/blob/master/docs/BINARY_FILE_FORMAT.TXT

21.67.1 Examples

  • Extract Closed Captions to RCWT using lavfi:
    ffmpeg -f lavfi -i "movie=INPUT.mkv[out+subcc]" -map 0:s:0 -c:s copy -f rcwt CC.rcwt.bin
    

21.68 segment, stream_segment, ssegment

Basic stream segmenter.

This muxer outputs streams to a number of separate files of nearly fixed duration. Output filename pattern can be set in a fashion similar to image2, or by using a strftime template if the strftime option is enabled.

stream_segment is a variant of the muxer used to write to streaming output formats, i.e. which do not require global headers, and is recommended for outputting e.g. to MPEG transport stream segments. ssegment is a shorter alias for stream_segment.

Every segment starts with a keyframe of the selected reference stream, which is set through the reference_stream option.

Note that if you want accurate splitting for a video file, you need to make the input key frames correspond to the exact splitting times expected by the segmenter, or the segment muxer will start the new segment with the key frame found next after the specified start time.

The segment muxer works best with a single constant frame rate video.

Optionally it can generate a list of the created segments, by setting the option segment_list. The list type is specified by the segment_list_type option. The entry filenames in the segment list are set by default to the basename of the corresponding segment files.

See also the hls muxer, which provides a more specific implementation for HLS segmentation.

21.68.1 Options

The segment muxer supports the following options:

increment_tc 1|0

if set to 1, increment timecode between each segment If this is selected, the input need to have a timecode in the first video stream. Default value is 0.

reference_stream specifier

Set the reference stream, as specified by the string specifier. If specifier is set to auto, the reference is chosen automatically. Otherwise it must be a stream specifier (see the “Stream specifiers” chapter in the ffmpeg manual) which specifies the reference stream. The default value is auto.

segment_format format

Override the inner container format, by default it is guessed by the filename extension.

segment_format_options options_list

Set output format options using a :-separated list of key=value parameters. Values containing the : special character must be escaped.

segment_list name

Generate also a listfile named name. If not specified no listfile is generated.

segment_list_flags flags

Set flags affecting the segment list generation.

It currently supports the following flags:

cache

Allow caching (only affects M3U8 list files).

live

Allow live-friendly file generation.

segment_list_size size

Update the list file so that it contains at most size segments. If 0 the list file will contain all the segments. Default value is 0.

segment_list_entry_prefix prefix

Prepend prefix to each entry. Useful to generate absolute paths. By default no prefix is applied.

segment_list_type type

Select the listing format.

The following values are recognized:

flat

Generate a flat list for the created segments, one segment per line.

csv, ext

Generate a list for the created segments, one segment per line, each line matching the format (comma-separated values):

segment_filename,segment_start_time,segment_end_time

segment_filename is the name of the output file generated by the muxer according to the provided pattern. CSV escaping (according to RFC4180) is applied if required.

segment_start_time and segment_end_time specify the segment start and end time expressed in seconds.

A list file with the suffix ".csv" or ".ext" will auto-select this format.

ext’ is deprecated in favor or ‘csv’.

ffconcat

Generate an ffconcat file for the created segments. The resulting file can be read using the FFmpeg concat demuxer.

A list file with the suffix ".ffcat" or ".ffconcat" will auto-select this format.

m3u8

Generate an extended M3U8 file, version 3, compliant with http://tools.ietf.org/id/draft-pantos-http-live-streaming.

A list file with the suffix ".m3u8" will auto-select this format.

If not specified the type is guessed from the list file name suffix.

segment_time time

Set segment duration to time, the value must be a duration specification. Default value is "2". See also the segment_times option.

Note that splitting may not be accurate, unless you force the reference stream key-frames at the given time. See the introductory notice and the examples below.

min_seg_duration time

Set minimum segment duration to time, the value must be a duration specification. This prevents the muxer ending segments at a duration below this value. Only effective with segment_time. Default value is "0".

segment_atclocktime 1|0

If set to "1" split at regular clock time intervals starting from 00:00 o’clock. The time value specified in segment_time is used for setting the length of the splitting interval.

For example with segment_time set to "900" this makes it possible to create files at 12:00 o’clock, 12:15, 12:30, etc.

Default value is "0".

segment_clocktime_offset duration

Delay the segment splitting times with the specified duration when using segment_atclocktime.

For example with segment_time set to "900" and segment_clocktime_offset set to "300" this makes it possible to create files at 12:05, 12:20, 12:35, etc.

Default value is "0".

segment_clocktime_wrap_duration duration

Force the segmenter to only start a new segment if a packet reaches the muxer within the specified duration after the segmenting clock time. This way you can make the segmenter more resilient to backward local time jumps, such as leap seconds or transition to standard time from daylight savings time.

Default is the maximum possible duration which means starting a new segment regardless of the elapsed time since the last clock time.

segment_time_delta delta

Specify the accuracy time when selecting the start time for a segment, expressed as a duration specification. Default value is "0".

When delta is specified a key-frame will start a new segment if its PTS satisfies the relation:

PTS >= start_time - time_delta

This option is useful when splitting video content, which is always split at GOP boundaries, in case a key frame is found just before the specified split time.

In particular may be used in combination with the ffmpeg option force_key_frames. The key frame times specified by force_key_frames may not be set accurately because of rounding issues, with the consequence that a key frame time may result set just before the specified time. For constant frame rate videos a value of 1/(2*frame_rate) should address the worst case mismatch between the specified time and the time set by force_key_frames.

segment_times times

Specify a list of split points. times contains a list of comma separated duration specifications, in increasing order. See also the segment_time option.

segment_frames frames

Specify a list of split video frame numbers. frames contains a list of comma separated integer numbers, in increasing order.

This option specifies to start a new segment whenever a reference stream key frame is found and the sequential number (starting from 0) of the frame is greater or equal to the next value in the list.

segment_wrap limit

Wrap around segment index once it reaches limit.

segment_start_number number

Set the sequence number of the first segment. Defaults to 0.

strftime 1|0

Use the strftime function to define the name of the new segments to write. If this is selected, the output segment name must contain a strftime function template. Default value is 0.

break_non_keyframes 1|0

If enabled, allow segments to start on frames other than keyframes. This improves behavior on some players when the time between keyframes is inconsistent, but may make things worse on others, and can cause some oddities during seeking. Defaults to 0.

reset_timestamps 1|0

Reset timestamps at the beginning of each segment, so that each segment will start with near-zero timestamps. It is meant to ease the playback of the generated segments. May not work with some combinations of muxers/codecs. It is set to 0 by default.

initial_offset offset

Specify timestamp offset to apply to the output packet timestamps. The argument must be a time duration specification, and defaults to 0.

write_empty_segments 1|0

If enabled, write an empty segment if there are no packets during the period a segment would usually span. Otherwise, the segment will be filled with the next packet written. Defaults to 0.

Make sure to require a closed GOP when encoding and to set the GOP size to fit your segment time constraint.

21.68.2 Examples

  • Remux the content of file in.mkv to a list of segments out-000.nut, out-001.nut, etc., and write the list of generated segments to out.list:
    ffmpeg -i in.mkv -codec hevc -flags +cgop -g 60 -map 0 -f segment -segment_list out.list out%03d.nut
    
  • Segment input and set output format options for the output segments:
    ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
    
  • Segment the input file according to the split points specified by the segment_times option:
    ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
    
  • Use the ffmpeg force_key_frames option to force key frames in the input at the specified location, together with the segment option segment_time_delta to account for possible roundings operated when setting key frame times.
    ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
    -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut
    

    In order to force key frames on the input file, transcoding is required.

  • Segment the input file by splitting the input file according to the frame numbers sequence specified with the segment_frames option:
    ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
    
  • Convert the in.mkv to TS segments using the libx264 and aac encoders:
    ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts
    
  • Segment the input file, and create an M3U8 live playlist (can be used as live HLS source):
    ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
    -segment_list_flags +live -segment_time 10 out%03d.mkv
    

21.69 smoothstreaming

Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server.

window_size

Specify the number of fragments kept in the manifest. Default 0 (keep all).

extra_window_size

Specify the number of fragments kept outside of the manifest before removing from disk. Default 5.

lookahead_count

Specify the number of lookahead fragments. Default 2.

min_frag_duration

Specify the minimum fragment duration (in microseconds). Default 5000000.

remove_at_exit

Specify whether to remove all fragments when finished. Default 0 (do not remove).

21.70 streamhash

Per stream hash testing format.

This muxer computes and prints a cryptographic hash of all the input frames, on a per-stream basis. This can be used for equality checks without having to do a complete binary comparison.

By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. Timestamps are ignored. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms.

The output of the muxer consists of one line per stream of the form: streamindex,streamtype,algo=hash, where streamindex is the index of the mapped stream, streamtype is a single character indicating the type of stream, algo is a short string representing the hash function used, and hash is a hexadecimal number representing the computed hash.

hash algorithm

Use the cryptographic hash function specified by the string algorithm. Supported values include MD5, murmur3, RIPEMD128, RIPEMD160, RIPEMD256, RIPEMD320, SHA160, SHA224, SHA256 (default), SHA512/224, SHA512/256, SHA384, SHA512, CRC32 and adler32.

21.70.1 Examples

To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file out.sha256:

ffmpeg -i INPUT -f streamhash out.sha256

To print an MD5 hash to stdout use the command:

ffmpeg -i INPUT -f streamhash -hash md5 -

See also the hash and framehash muxers.

21.71 tee

The tee muxer can be used to write the same data to several outputs, such as files or streams. It can be used, for example, to stream a video over a network and save it to disk at the same time.

It is different from specifying several outputs to the ffmpeg command-line tool. With the tee muxer, the audio and video data will be encoded only once. With conventional multiple outputs, multiple encoding operations in parallel are initiated, which can be a very expensive process. The tee muxer is not useful when using the libavformat API directly because it is then possible to feed the same packets to several muxers directly.

Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-select output streams. So all streams intended for output must be specified using -map. See the examples below.

Some encoders may need different options depending on the output format; the auto-detection of this can not work with the tee muxer, so they need to be explicitly specified. The main example is the global_header flag.

The slave outputs are specified in the file name given to the muxer, separated by ’|’. If any of the slave name contains the ’|’ separator, leading or trailing spaces or any special character, those must be escaped (see (ffmpeg-utils)the "Quoting and escaping" section in the ffmpeg-utils(1) manual).

21.71.1 Options

use_fifo bool

If set to 1, slave outputs will be processed in separate threads using the fifo muxer. This allows to compensate for different speed/latency/reliability of outputs and setup transparent recovery. By default this feature is turned off.

fifo_options

Options to pass to fifo pseudo-muxer instances. See fifo.

Muxer options can be specified for each slave by prepending them as a list of key=value pairs separated by ’:’, between square brackets. If the options values contain a special character or the ’:’ separator, they must be escaped; note that this is a second level escaping.

The following special options are also recognized:

f

Specify the format name. Required if it cannot be guessed from the output URL.

bsfs[/spec]

Specify a list of bitstream filters to apply to the specified output.

It is possible to specify to which streams a given bitstream filter applies, by appending a stream specifier to the option separated by /. spec must be a stream specifier (see Format stream specifiers).

If the stream specifier is not specified, the bitstream filters will be applied to all streams in the output. This will cause that output operation to fail if the output contains streams to which the bitstream filter cannot be applied e.g. h264_mp4toannexb being applied to an output containing an audio stream.

Options for a bitstream filter must be specified in the form of opt=value.

Several bitstream filters can be specified, separated by ",".

use_fifo bool

This allows to override tee muxer use_fifo option for individual slave muxer.

fifo_options

This allows to override tee muxer fifo_options for individual slave muxer. See fifo.

select

Select the streams that should be mapped to the slave output, specified by a stream specifier. If not specified, this defaults to all the mapped streams. This will cause that output operation to fail if the output format does not accept all mapped streams.

You may use multiple stream specifiers separated by commas (,) e.g.: a:0,v

onfail

Specify behaviour on output failure. This can be set to either abort (which is default) or ignore. abort will cause whole process to fail in case of failure on this slave output. ignore will ignore failure on this output, so other outputs will continue without being affected.

21.71.2 Examples

  • Encode something and both archive it in a WebM file and stream it as MPEG-TS over UDP:
    ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
      "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
    
  • As above, but continue streaming even if output to local file fails (for example local drive fills up):
    ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
      "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
    
  • Use ffmpeg to encode the input, and send the output to three different destinations. The dump_extra bitstream filter is used to add extradata information to all the output video keyframes packets, as requested by the MPEG-TS format. The select option is applied to out.aac in order to make it contain only audio packets.
    ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
           -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
    
  • As above, but select only stream a:1 for the audio output. Note that a second level escaping must be performed, as ":" is a special character used to separate options.
    ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
           -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
    

21.72 webm_chunk

WebM Live Chunk Muxer.

This muxer writes out WebM headers and chunks as separate files which can be consumed by clients that support WebM Live streams via DASH.

21.72.1 Options

This muxer supports the following options:

chunk_start_index

Index of the first chunk (defaults to 0).

header

Filename of the header where the initialization data will be written.

audio_chunk_duration

Duration of each audio chunk in milliseconds (defaults to 5000).

21.72.2 Example

ffmpeg -f v4l2 -i /dev/video0 \
       -f alsa -i hw:0 \
       -map 0:0 \
       -c:v libvpx-vp9 \
       -s 640x360 -keyint_min 30 -g 30 \
       -f webm_chunk \
       -header webm_live_video_360.hdr \
       -chunk_start_index 1 \
       webm_live_video_360_%d.chk \
       -map 1:0 \
       -c:a libvorbis \
       -b:a 128k \
       -f webm_chunk \
       -header webm_live_audio_128.hdr \
       -chunk_start_index 1 \
       -audio_chunk_duration 1000 \
       webm_live_audio_128_%d.chk

21.73 webm_dash_manifest

WebM DASH Manifest muxer.

This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML. It also supports manifest generation for DASH live streams.

For more information see:

21.73.1 Options

This muxer supports the following options:

adaptation_sets

This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding audio and video streams. Any number of adaptation sets can be added using this option.

live

Set this to 1 to create a live stream DASH Manifest. Default: 0.

chunk_start_index

Start index of the first chunk. This will go in the ‘startNumber’ attribute of the ‘SegmentTemplate’ element in the manifest. Default: 0.

chunk_duration_ms

Duration of each chunk in milliseconds. This will go in the ‘duration’ attribute of the ‘SegmentTemplate’ element in the manifest. Default: 1000.

utc_timing_url

URL of the page that will return the UTC timestamp in ISO format. This will go in the ‘value’ attribute of the ‘UTCTiming’ element in the manifest. Default: None.

time_shift_buffer_depth

Smallest time (in seconds) shifting buffer for which any Representation is guaranteed to be available. This will go in the ‘timeShiftBufferDepth’ attribute of the ‘MPD’ element. Default: 60.

minimum_update_period

Minimum update period (in seconds) of the manifest. This will go in the ‘minimumUpdatePeriod’ attribute of the ‘MPD’ element. Default: 0.

21.73.2 Example

ffmpeg -f webm_dash_manifest -i video1.webm \
       -f webm_dash_manifest -i video2.webm \
       -f webm_dash_manifest -i audio1.webm \
       -f webm_dash_manifest -i audio2.webm \
       -map 0 -map 1 -map 2 -map 3 \
       -c copy \
       -f webm_dash_manifest \
       -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
       manifest.xml

22 Metadata

FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then load it back using the metadata muxer/demuxer.

The file format is as follows:

  1. A file consists of a header and a number of metadata tags divided into sections, each on its own line.
  2. The header is a ‘;FFMETADATA’ string, followed by a version number (now 1).
  3. Metadata tags are of the form ‘key=value
  4. Immediately after header follows global metadata
  5. After global metadata there may be sections with per-stream/per-chapter metadata.
  6. A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in brackets (‘[’, ‘]’) and ends with next section or end of file.
  7. At the beginning of a chapter section there may be an optional timebase to be used for start/end values. It must be in form ‘TIMEBASE=num/den’, where num and den are integers. If the timebase is missing then start/end times are assumed to be in nanoseconds.

    Next a chapter section must contain chapter start and end times in form ‘START=num’, ‘END=num’, where num is a positive integer.

  8. Empty lines and lines starting with ‘;’ or ‘#’ are ignored.
  9. Metadata keys or values containing special characters (‘=’, ‘;’, ‘#’, ‘\’ and a newline) must be escaped with a backslash ‘\’.
  10. Note that whitespace in metadata (e.g. ‘foo = bar’) is considered to be a part of the tag (in the example above key is ‘foo ’, value is ‘ bar’).

A ffmetadata file might look like this:

;FFMETADATA1
title=bike\\shed
;this is a comment
artist=FFmpeg troll team

[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line

By using the ffmetadata muxer and demuxer it is possible to extract metadata from an input file to an ffmetadata file, and then transcode the file into an output file with the edited ffmetadata file.

Extracting an ffmetadata file with ffmpeg goes as follows:

ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

Reinserting edited metadata information from the FFMETADATAFILE file can be done as:

ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT

23 Protocol Options

The libavformat library provides some generic global options, which can be set on all the protocols. In addition each protocol may support so-called private options, which are specific for that component.

Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in the AVFormatContext options or using the libavutil/opt.h API for programmatic use.

The list of supported options follows:

protocol_whitelist list (input)

Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols prefixed by "-" are disabled. All protocols are allowed by default but protocols used by an another protocol (nested protocols) are restricted to a per protocol subset.

24 Protocols

Protocols are configured elements in FFmpeg that enable access to resources that require specific protocols.

When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option "–list-protocols".

You can disable all the protocols using the configure option "–disable-protocols", and selectively enable a protocol using the option "–enable-protocol=PROTOCOL", or you can disable a particular protocol using the option "–disable-protocol=PROTOCOL".

The option "-protocols" of the ff* tools will display the list of supported protocols.

All protocols accept the following options:

rw_timeout

Maximum time to wait for (network) read/write operations to complete, in microseconds.

A description of the currently available protocols follows.

24.1 amqp

Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based publish-subscribe communication protocol.

FFmpeg must be compiled with –enable-librabbitmq to support AMQP. A separate AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.

After starting the broker, an FFmpeg client may stream data to the broker using the command:

ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]

Where hostname and port (default is 5672) is the address of the broker. The client may also set a user/password for authentication. The default for both fields is "guest". Name of virtual host on broker can be set with vhost. The default value is "/".

Muliple subscribers may stream from the broker using the command:

ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]

In RabbitMQ all data published to the broker flows through a specific exchange, and each subscribing client has an assigned queue/buffer. When a packet arrives at an exchange, it may be copied to a client’s queue depending on the exchange and routing_key fields.

The following options are supported:

exchange

Sets the exchange to use on the broker. RabbitMQ has several predefined exchanges: "amq.direct" is the default exchange, where the publisher and subscriber must have a matching routing_key; "amq.fanout" is the same as a broadcast operation (i.e. the data is forwarded to all queues on the fanout exchange independent of the routing_key); and "amq.topic" is similar to "amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ documentation).

routing_key

Sets the routing key. The default value is "amqp". The routing key is used on the "amq.direct" and "amq.topic" exchanges to decide whether packets are written to the queue of a subscriber.

pkt_size

Maximum size of each packet sent/received to the broker. Default is 131072. Minimum is 4096 and max is any large value (representable by an int). When receiving packets, this sets an internal buffer size in FFmpeg. It should be equal to or greater than the size of the published packets to the broker. Otherwise the received message may be truncated causing decoding errors.

connection_timeout

The timeout in seconds during the initial connection to the broker. The default value is rw_timeout, or 5 seconds if rw_timeout is not set.

delivery_mode mode

Sets the delivery mode of each message sent to broker. The following values are accepted:

persistent

Delivery mode set to "persistent" (2). This is the default value. Messages may be written to the broker’s disk depending on its setup.

non-persistent

Delivery mode set to "non-persistent" (1). Messages will stay in broker’s memory unless the broker is under memory pressure.

24.2 async

Asynchronous data filling wrapper for input stream.

Fill data in a background thread, to decouple I/O operation from demux thread.

async:URL
async:http://host/resource
async:cache:http://host/resource

24.3 bluray

Read BluRay playlist.

The accepted options are:

angle

BluRay angle

chapter

Start chapter (1...N)

playlist

Playlist to read (BDMV/PLAYLIST/?????.mpls)

Examples:

Read longest playlist from BluRay mounted to /mnt/bluray:

bluray:/mnt/bluray

Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:

-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

24.4 cache

Caching wrapper for input stream.

Cache the input stream to temporary file. It brings seeking capability to live streams.

The accepted options are:

read_ahead_limit

Amount in bytes that may be read ahead when seeking isn’t supported. Range is -1 to INT_MAX. -1 for unlimited. Default is 65536.

URL Syntax is

cache:URL

24.5 concat

Physical concatenation protocol.

Read and seek from many resources in sequence as if they were a unique resource.

A URL accepted by this protocol has the syntax:

concat:URL1|URL2|...|URLN

where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol.

For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay use the command:

ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

Note that you may need to escape the character "|" which is special for many shells.

24.6 concatf

Physical concatenation protocol using a line break delimited list of resources.

Read and seek from many resources in sequence as if they were a unique resource.

A URL accepted by this protocol has the syntax:

concatf:URL

where URL is the url containing a line break delimited list of resources to be concatenated, each one possibly specifying a distinct protocol. Special characters must be escaped with backslash or single quotes. See (ffmpeg-utils)the "Quoting and escaping" section in the ffmpeg-utils(1) manual.

For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg listed in separate lines within a file split.txt with ffplay use the command:

ffplay concatf:split.txt

Where split.txt contains the lines:

split1.mpeg
split2.mpeg
split3.mpeg

24.7 crypto

AES-encrypted stream reading protocol.

The accepted options are:

key

Set the AES decryption key binary block from given hexadecimal representation.

iv

Set the AES decryption initialization vector binary block from given hexadecimal representation.

Accepted URL formats:

crypto:URL
crypto+URL

24.8 data

Data in-line in the URI. See http://en.wikipedia.org/wiki/Data_URI_scheme.

For example, to convert a GIF file given inline with ffmpeg:

ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

24.9 fd

File descriptor access protocol.

The accepted syntax is:

fd: -fd file_descriptor

If fd is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has seek support if it corresponding to a regular file. fd protocol doesn’t support pass file descriptor via URL for security.

This protocol accepts the following options:

blocksize

Set I/O operation maximum block size, in bytes. Default value is INT_MAX, which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable if data transmission is slow.

fd

Set file descriptor.

24.10 file

File access protocol.

Read from or write to a file.

A file URL can have the form:

file:filename

where filename is the path of the file to read.

An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on the build, an URL that looks like a Windows path with the drive letter at the beginning will also be assumed to be a file URL (usually not the case in builds for unix-like systems).

For example to read from a file input.mpeg with ffmpeg use the command:

ffmpeg -i file:input.mpeg output.mpeg

This protocol accepts the following options:

truncate

Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.

blocksize

Set I/O operation maximum block size, in bytes. Default value is INT_MAX, which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable for files on slow medium.

follow

If set to 1, the protocol will retry reading at the end of the file, allowing reading files that still are being written. In order for this to terminate, you either need to use the rw_timeout option, or use the interrupt callback (for API users).

seekable

Controls if seekability is advertised on the file. 0 means non-seekable, -1 means auto (seekable for normal files, non-seekable for named pipes).

Many demuxers handle seekable and non-seekable resources differently, overriding this might speed up opening certain files at the cost of losing some features (e.g. accurate seeking).

24.11 ftp

FTP (File Transfer Protocol).

Read from or write to remote resources using FTP protocol.

Following syntax is required.

ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout

Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.

ftp-user

Set a user to be used for authenticating to the FTP server. This is overridden by the user in the FTP URL.

ftp-password

Set a password to be used for authenticating to the FTP server. This is overridden by the password in the FTP URL, or by ftp-anonymous-password if no user is set.

ftp-anonymous-password

Password used when login as anonymous user. Typically an e-mail address should be used.

ftp-write-seekable

Control seekability of connection during encoding. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value is 0.

NOTE: Protocol can be used as output, but it is recommended to not do it, unless special care is taken (tests, customized server configuration etc.). Different FTP servers behave in different way during seek operation. ff* tools may produce incomplete content due to server limitations.

24.12 gopher

Gopher protocol.

24.13 gophers

Gophers protocol.

The Gopher protocol with TLS encapsulation.

24.14 hls

Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+proto" after the hls URI scheme name, where proto is either "file" or "http".

hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8

Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files.

24.15 http

HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options:

seekable

Control seekability of connection. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable, if set to -1 it will try to autodetect if it is seekable. Default value is -1.

chunked_post

If set to 1 use chunked Transfer-Encoding for posts, default is 1.

http_proxy

set HTTP proxy to tunnel through e.g. http://example.com:1234

headers

Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers.

content_type

Set a specific content type for the POST messages or for listen mode.

user_agent

Override the User-Agent header. If not specified the protocol will use a string describing the libavformat build. ("Lavf/<version>")

referer

Set the Referer header. Include ’Referer: URL’ header in HTTP request.

multiple_requests

Use persistent connections if set to 1, default is 0.

post_data

Set custom HTTP post data.

mime_type

Export the MIME type.

http_version

Exports the HTTP response version number. Usually "1.0" or "1.1".

cookies

Set the cookies to be sent in future requests. The format of each cookie is the same as the value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by a newline character.

icy

If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the metadata has to be retrieved by the application by reading the icy_metadata_headers and icy_metadata_packet options. The default is 1.

icy_metadata_headers

If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters.

icy_metadata_packet

If the server supports ICY metadata, and icy was set to 1, this contains the last non-empty metadata packet sent by the server. It should be polled in regular intervals by applications interested in mid-stream metadata updates.

metadata

Set an exported dictionary containing Icecast metadata from the bitstream, if present. Only useful with the C API.

auth_type

Set HTTP authentication type. No option for Digest, since this method requires getting nonce parameters from the server first and can’t be used straight away like Basic.

none

Choose the HTTP authentication type automatically. This is the default.

basic

Choose the HTTP basic authentication.

Basic authentication sends a Base64-encoded string that contains a user name and password for the client. Base64 is not a form of encryption and should be considered the same as sending the user name and password in clear text (Base64 is a reversible encoding). If a resource needs to be protected, strongly consider using an authentication scheme other than basic authentication. HTTPS/TLS should be used with basic authentication. Without these additional security enhancements, basic authentication should not be used to protect sensitive or valuable information.

send_expect_100

Send an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it won’t, if set to -1 it will try to send if it is applicable. Default value is -1.

location

An exported dictionary containing the content location. Only useful with the C API.

offset

Set initial byte offset.

end_offset

Try to limit the request to bytes preceding this offset.

method

When used as a client option it sets the HTTP method for the request.

When used as a server option it sets the HTTP method that is going to be expected from the client(s). If the expected and the received HTTP method do not match the client will be given a Bad Request response. When unset the HTTP method is not checked for now. This will be replaced by autodetection in the future.

reconnect

Reconnect automatically when disconnected before EOF is hit.

reconnect_at_eof

If set then eof is treated like an error and causes reconnection, this is useful for live / endless streams.

reconnect_on_network_error

Reconnect automatically in case of TCP/TLS errors during connect.

reconnect_on_http_error

A comma separated list of HTTP status codes to reconnect on. The list can include specific status codes (e.g. ’503’) or the strings ’4xx’ / ’5xx’.

reconnect_streamed

If set then even streamed/non seekable streams will be reconnected on errors.

reconnect_delay_max

Set the maximum delay in seconds after which to give up reconnecting.

reconnect_max_retries

Set the maximum number of times to retry a connection. Default unset.

reconnect_delay_total_max

Set the maximum total delay in seconds after which to give up reconnecting.

respect_retry_after

If enabled, and a Retry-After header is encountered, its requested reconnection delay will be honored, rather than using exponential backoff. Useful for 429 and 503 errors. Default enabled.

listen

If set to 1 enables experimental HTTP server. This can be used to send data when used as an output option, or read data from a client with HTTP POST when used as an input option. If set to 2 enables experimental multi-client HTTP server. This is not yet implemented in ffmpeg.c and thus must not be used as a command line option.

# Server side (sending):
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://server:port

# Client side (receiving):
ffmpeg -i http://server:port -c copy somefile.ogg

# Client can also be done with wget:
wget http://server:port -O somefile.ogg

# Server side (receiving):
ffmpeg -listen 1 -i http://server:port -c copy somefile.ogg

# Client side (sending):
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://server:port

# Client can also be done with wget:
wget --post-file=somefile.ogg http://server:port
resource

The resource requested by a client, when the experimental HTTP server is in use.

reply_code

The HTTP code returned to the client, when the experimental HTTP server is in use.

short_seek_size

Set the threshold, in bytes, for when a readahead should be prefered over a seek and new HTTP request. This is useful, for example, to make sure the same connection is used for reading large video packets with small audio packets in between.

24.15.1 HTTP Cookies

Some HTTP requests will be denied unless cookie values are passed in with the request. The cookies option allows these cookies to be specified. At the very least, each cookie must specify a value along with a path and domain. HTTP requests that match both the domain and path will automatically include the cookie value in the HTTP Cookie header field. Multiple cookies can be delimited by a newline.

The required syntax to play a stream specifying a cookie is:

ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8

24.16 Icecast

Icecast protocol (stream to Icecast servers)

This protocol accepts the following options:

ice_genre

Set the stream genre.

ice_name

Set the stream name.

ice_description

Set the stream description.

ice_url

Set the stream website URL.

ice_public

Set if the stream should be public. The default is 0 (not public).

user_agent

Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.

password

Set the Icecast mountpoint password.

content_type

Set the stream content type. This must be set if it is different from audio/mpeg.

legacy_icecast

This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method.

tls

Establish a TLS (HTTPS) connection to Icecast.

icecast://[username[:password]@]server:port/mountpoint

24.17 ipfs

InterPlanetary File System (IPFS) protocol support. One can access files stored on the IPFS network through so-called gateways. These are http(s) endpoints. This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent to such a gateway. Users can (and should) host their own node which means this protocol will use one’s local gateway to access files on the IPFS network.

This protocol accepts the following options:

gateway

Defines the gateway to use. When not set, the protocol will first try locating the local gateway by looking at $IPFS_GATEWAY, $IPFS_PATH and $HOME/.ipfs/, in that order.

One can use this protocol in 2 ways. Using IPFS:

ffplay ipfs://<hash>

Or the IPNS protocol (IPNS is mutable IPFS):

ffplay ipns://<hash>

24.18 mmst

MMS (Microsoft Media Server) protocol over TCP.

24.19 mmsh

MMS (Microsoft Media Server) protocol over HTTP.

The required syntax is:

mmsh://server[:port][/app][/playpath]

24.20 md5

MD5 output protocol.

Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file.

Some examples follow.

# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
ffmpeg -i input.flv -f avi -y md5:output.avi.md5

# Write the MD5 hash of the encoded AVI file to stdout.
ffmpeg -i input.flv -f avi -y md5:

Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol.

24.21 pipe

UNIX pipe access protocol.

Read and write from UNIX pipes.

The accepted syntax is:

pipe:[number]

If fd isn’t specified, number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading.

For example to read from stdin with ffmpeg:

cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:

For writing to stdout with ffmpeg:

ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
ffmpeg -i test.wav -f avi pipe: | cat > test.avi

This protocol accepts the following options:

blocksize

Set I/O operation maximum block size, in bytes. Default value is INT_MAX, which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable if data transmission is slow.

fd

Set file descriptor.

Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol.

24.22 prompeg

Pro-MPEG Code of Practice #3 Release 2 FEC protocol.

The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2 Transport Streams sent over RTP.

This protocol must be used in conjunction with the rtp_mpegts muxer and the rtp protocol.

The required syntax is:

-f rtp_mpegts -fec prompeg=option=val... rtp://hostname:port

The destination UDP ports are port + 2 for the column FEC stream and port + 4 for the row FEC stream.

This protocol accepts the following options:

l=n

The number of columns (4-20, LxD <= 100)

d=n

The number of rows (4-20, LxD <= 100)

Example usage:

-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://hostname:port

24.23 rist

Reliable Internet Streaming Transport protocol

The accepted options are:

rist_profile

Supported values:

simple
main

This one is default.

advanced
buffer_size

Set internal RIST buffer size in milliseconds for retransmission of data. Default value is 0 which means the librist default (1 sec). Maximum value is 30 seconds.

fifo_size

Size of the librist receiver output fifo in number of packets. This must be a power of 2. Defaults to 8192 (vs the librist default of 1024).

overrun_nonfatal=1|0

Survive in case of librist fifo buffer overrun. Default value is 0.

pkt_size

Set maximum packet size for sending data. 1316 by default.

log_level

Set loglevel for RIST logging messages. You only need to set this if you explicitly want to enable debug level messages or packet loss simulation, otherwise the regular loglevel is respected.

secret

Set override of encryption secret, by default is unset.

encryption

Set encryption type, by default is disabled. Acceptable values are 128 and 256.

24.24 rtmp

Real-Time Messaging Protocol.

The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network.

The required syntax is:

rtmp://[username:password@]server[:port][/app][/instance][/playpath]

The accepted parameters are:

username

An optional username (mostly for publishing).

password

An optional password (mostly for publishing).

server

The address of the RTMP server.

port

The number of the TCP port to use (by default is 1935).

app

It is the name of the application to access. It usually corresponds to the path where the application is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.). You can override the value parsed from the URI through the rtmp_app option, too.

playpath

It is the path or name of the resource to play with reference to the application specified in app, may be prefixed by "mp4:". You can override the value parsed from the URI through the rtmp_playpath option, too.

listen

Act as a server, listening for an incoming connection.

timeout

Maximum time to wait for the incoming connection. Implies listen.

Additionally, the following parameters can be set via command line options (or in code via AVOptions):

rtmp_app

Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI.

rtmp_buffer

Set the client buffer time in milliseconds. The default is 3000.

rtmp_conn

Extra arbitrary AMF connection parameters, parsed from a string, e.g. like B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0. Each value is prefixed by a single character denoting the type, B for Boolean, N for number, S for string, O for object, or Z for null, followed by a colon. For Booleans the data must be either 0 or 1 for FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or 1 to end or begin an object, respectively. Data items in subobjects may be named, by prefixing the type with ’N’ and specifying the name before the value (i.e. NB:myFlag:1). This option may be used multiple times to construct arbitrary AMF sequences.

rtmp_enhanced_codecs

Specify the list of codecs the client advertises to support in an enhanced RTMP stream. This option should be set to a comma separated list of fourcc values, like hvc1,av01,vp09 for multiple codecs or hvc1 for only one codec. The specified list will be presented in the "fourCcLive" property of the Connect Command Message.

rtmp_flashver

Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)

rtmp_flush_interval

Number of packets flushed in the same request (RTMPT only). The default is 10.

rtmp_live

Specify that the media is a live stream. No resuming or seeking in live streams is possible. The default value is any, which means the subscriber first tries to play the live stream specified in the playpath. If a live stream of that name is not found, it plays the recorded stream. The other possible values are live and recorded.

rtmp_pageurl

URL of the web page in which the media was embedded. By default no value will be sent.

rtmp_playpath

Stream identifier to play or to publish. This option overrides the parameter specified in the URI.

rtmp_subscribe

Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live.

rtmp_swfhash

SHA256 hash of the decompressed SWF file (32 bytes).

rtmp_swfsize

Size of the decompressed SWF file, required for SWFVerification.

rtmp_swfurl

URL of the SWF player for the media. By default no value will be sent.

rtmp_swfverify

URL to player swf file, compute hash/size automatically.

rtmp_tcurl

URL of the target stream. Defaults to proto://host[:port]/app.

tcp_nodelay=1|0

Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0.

Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.

For example to read with ffplay a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver":

ffplay rtmp://myserver/vod/sample

To publish to a password protected server, passing the playpath and app names separately:

ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

24.25 rtmpe

Encrypted Real-Time Messaging Protocol.

The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and HMACSHA256, generating a pair of RC4 keys.

24.26 rtmps

Real-Time Messaging Protocol over a secure SSL connection.

The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an encrypted connection.

24.27 rtmpt

Real-Time Messaging Protocol tunneled through HTTP.

The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming multimedia content within HTTP requests to traverse firewalls.

24.28 rtmpte

Encrypted Real-Time Messaging Protocol tunneled through HTTP.

The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for streaming multimedia content within HTTP requests to traverse firewalls.

24.29 rtmpts

Real-Time Messaging Protocol tunneled through HTTPS.

The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls.

24.30 libsmbclient

libsmbclient permits one to manipulate CIFS/SMB network resources.

Following syntax is required.

smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

This protocol accepts the following options.

timeout

Set timeout in milliseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.

truncate

Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.

workgroup

Set the workgroup used for making connections. By default workgroup is not specified.

For more information see: http://www.samba.org/.

24.31 libssh

Secure File Transfer Protocol via libssh

Read from or write to remote resources using SFTP protocol.

Following syntax is required.

sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout

Set timeout of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.

truncate

Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.

private_key

Specify the path of the file containing private key to use during authorization. By default libssh searches for keys in the ~/.ssh/ directory.

Example: Play a file stored on remote server.

ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

24.32 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte

Real-Time Messaging Protocol and its variants supported through librtmp.

Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with "–enable-librtmp". If enabled this will replace the native RTMP protocol.

This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).

The required syntax is:

rtmp_proto://server[:port][/app][/playpath] options

where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space-separated options of the form key=val.

See the librtmp manual page (man 3 librtmp) for more information.

For example, to stream a file in real-time to an RTMP server using ffmpeg:

ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

To play the same stream using ffplay:

ffplay "rtmp://myserver/live/mystream live=1"

24.33 rtp

Real-time Transport Protocol.

The required syntax for an RTP URL is: rtp://hostname[:port][?option=val...]

port specifies the RTP port to use.

The following URL options are supported:

ttl=n

Set the TTL (Time-To-Live) value (for multicast only).

rtcpport=n

Set the remote RTCP port to n.

localrtpport=n

Set the local RTP port to n.

localrtcpport=n'

Set the local RTCP port to n.

pkt_size=n

Set max packet size (in bytes) to n.

buffer_size=size

Set the maximum UDP socket buffer size in bytes.

connect=0|1

Do a connect() on the UDP socket (if set to 1) or not (if set to 0).

sources=ip[,ip]

List allowed source IP addresses.

block=ip[,ip]

List disallowed (blocked) source IP addresses.

write_to_source=0|1

Send packets to the source address of the latest received packet (if set to 1) or to a default remote address (if set to 0).

localport=n

Set the local RTP port to n.

localaddr=addr

Local IP address of a network interface used for sending packets or joining multicast groups.

timeout=n

Set timeout (in microseconds) of socket I/O operations to n.

This is a deprecated option. Instead, localrtpport should be used.

Important notes:

  1. If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.
  2. If localrtpport (the local RTP port) is not set any available port will be used for the local RTP and RTCP ports.
  3. If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port value plus 1.

24.34 rtsp

Real-Time Streaming Protocol.

RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with data transferred over RDT).

The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock’s RTSP server).

The required syntax for a RTSP url is:

rtsp://hostname[:port]/path

Options can be set on the ffmpeg/ffplay command line, or set in code via AVOptions or in avformat_open_input.

24.34.1 Muxer

The following options are supported.

rtsp_transport

Set RTSP transport protocols.

It accepts the following values:

udp

Use UDP as lower transport protocol.

tcp

Use TCP (interleaving within the RTSP control channel) as lower transport protocol.

Default value is ‘0’.

rtsp_flags

Set RTSP flags.

The following values are accepted:

latm

Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.

rfc2190

Use RFC 2190 packetization instead of RFC 4629 for H.263.

skip_rtcp

Don’t send RTCP sender reports.

h264_mode0

Use mode 0 for H.264 in RTP.

send_bye

Send RTCP BYE packets when finishing.

Default value is ‘0’.

min_port

Set minimum local UDP port. Default value is 5000.

max_port

Set maximum local UDP port. Default value is 65000.

buffer_size

Set the maximum socket buffer size in bytes.

pkt_size

Set max send packet size (in bytes). Default value is 1472.

24.34.2 Demuxer

The following options are supported.

initial_pause

Do not start playing the stream immediately if set to 1. Default value is 0.

rtsp_transport

Set RTSP transport protocols.

It accepts the following values:

udp

Use UDP as lower transport protocol.

tcp

Use TCP (interleaving within the RTSP control channel) as lower transport protocol.

udp_multicast

Use UDP multicast as lower transport protocol.

http

Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.

https

Use HTTPs tunneling as lower transport protocol, which is useful for passing proxies and widely used for security consideration.

Multiple lower transport protocols may be specified, in that case they are tried one at a time (if the setup of one fails, the next one is tried). For the muxer, only the ‘tcp’ and ‘udp’ options are supported.

rtsp_flags

Set RTSP flags.

The following values are accepted:

filter_src

Accept packets only from negotiated peer address and port.

listen

Act as a server, listening for an incoming connection.

prefer_tcp

Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.

satip_raw

Export raw MPEG-TS stream instead of demuxing. The flag will simply write out the raw stream, with the original PAT/PMT/PIDs intact.

Default value is ‘none’.

allowed_media_types

Set media types to accept from the server.

The following flags are accepted:

video
audio
data
subtitle

By default it accepts all media types.

min_port

Set minimum local UDP port. Default value is 5000.

max_port

Set maximum local UDP port. Default value is 65000.

listen_timeout

Set maximum timeout (in seconds) to establish an initial connection. Setting listen_timeout > 0 sets rtsp_flags to ‘listen’. Default is -1 which means an infinite timeout when ‘listen’ mode is set.

reorder_queue_size

Set number of packets to buffer for handling of reordered packets.

timeout

Set socket TCP I/O timeout in microseconds.

user_agent

Override User-Agent header. If not specified, it defaults to the libavformat identifier string.

buffer_size

Set the maximum socket buffer size in bytes.

When receiving data over UDP, the demuxer tries to reorder received packets (since they may arrive out of order, or packets may get lost totally). This can be disabled by setting the maximum demuxing delay to zero (via the max_delay field of AVFormatContext).

When watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be chosen with -vst n and -ast n for video and audio respectively, and can be switched on the fly by pressing v and a.

24.34.3 Examples

The following examples all make use of the ffplay and ffmpeg tools.

  • Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
    ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
    
  • Watch a stream tunneled over HTTP:
    ffplay -rtsp_transport http rtsp://server/video.mp4
    
  • Send a stream in realtime to a RTSP server, for others to watch:
    ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
    
  • Receive a stream in realtime:
    ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp output
    

24.35 sap

Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams regularly on a separate port.

24.35.1 Muxer

The syntax for a SAP url given to the muxer is:

sap://destination[:port][?options]

The RTP packets are sent to destination on port port, or to port 5004 if no port is specified. options is a &-separated list. The following options are supported:

announce_addr=address

Specify the destination IP address for sending the announcements to. If omitted, the announcements are sent to the commonly used SAP announcement multicast address 224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.

announce_port=port

Specify the port to send the announcements on, defaults to 9875 if not specified.

ttl=ttl

Specify the time to live value for the announcements and RTP packets, defaults to 255.

same_port=0|1

If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports.

Example command lines follow.

To broadcast a stream on the local subnet, for watching in VLC:

ffmpeg -re -i input -f sap sap://224.0.0.255?same_port=1

Similarly, for watching in ffplay:

ffmpeg -re -i input -f sap sap://224.0.0.255

And for watching in ffplay, over IPv6:

ffmpeg -re -i input -f sap sap://[ff0e::1:2:3:4]

24.35.2 Demuxer

The syntax for a SAP url given to the demuxer is:

sap://[address][:port]

address is the multicast address to listen for announcements on, if omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted.

The demuxers listens for announcements on the given address and port. Once an announcement is received, it tries to receive that particular stream.

Example command lines follow.

To play back the first stream announced on the normal SAP multicast address:

ffplay sap://

To play back the first stream announced on one the default IPv6 SAP multicast address:

ffplay sap://[ff0e::2:7ffe]

24.36 sctp

Stream Control Transmission Protocol.

The accepted URL syntax is:

sctp://host:port[?options]

The protocol accepts the following options:

listen

If set to any value, listen for an incoming connection. Outgoing connection is done by default.

max_streams

Set the maximum number of streams. By default no limit is set.

24.37 srt

Haivision Secure Reliable Transport Protocol via libsrt.

The supported syntax for a SRT URL is:

srt://hostname:port[?options]

options contains a list of &-separated options of the form key=val.

or

options srt://hostname:port

options contains a list of ’-key val’ options.

This protocol accepts the following options.

connect_timeout=milliseconds

Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with the default connect timeout of 3 seconds. This option applies to the caller and rendezvous connection modes. The connect timeout is 10 times the value set for the rendezvous mode (which can be used as a workaround for this connection problem with earlier versions).

ffs=bytes

Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and you should set it to not less than recv_buffer_size and mss. The default value is relatively large, therefore unless you set a very large receiver buffer, you do not need to change this option. Default value is 25600.

inputbw=bytes/seconds

Sender nominal input rate, in bytes per seconds. Used along with oheadbw, when maxbw is set to relative (0), to calculate maximum sending rate when recovery packets are sent along with the main media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set while maxbw is set to relative (0), the actual input rate is evaluated inside the library. Default value is 0.

iptos=tos

IP Type of Service. Applies to sender only. Default value is 0xB8.

ipttl=ttl

IP Time To Live. Applies to sender only. Default value is 64.

latency=microseconds

Timestamp-based Packet Delivery Delay. Used to absorb bursts of missed packet retransmissions. This flag sets both rcvlatency and peerlatency to the same value. Note that prior to version 1.3.0 this is the only flag to set the latency, however this is effectively equivalent to setting peerlatency, when side is sender and rcvlatency when side is receiver, and the bidirectional stream sending is not supported.

listen_timeout=microseconds

Set socket listen timeout.

maxbw=bytes/seconds

Maximum sending bandwidth, in bytes per seconds. -1 infinite (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0 absolute limit value Default value is 0 (relative)

mode=caller|listener|rendezvous

Connection mode. caller opens client connection. listener starts server to listen for incoming connections. rendezvous use Rendez-Vous connection mode. Default value is caller.

mss=bytes

Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using a packet counter assuming fully filled packets. The smallest MSS between the peers is used. This is 1500 by default in the overall internet. This is the maximum size of the UDP packet and can be only decreased, unless you have some unusual dedicated network settings. Default value is 1500.

nakreport=1|0

If set to 1, Receiver will send ‘UMSG_LOSSREPORT‘ messages periodically until a lost packet is retransmitted or intentionally dropped. Default value is 1.

oheadbw=percents

Recovery bandwidth overhead above input rate, in percents. See inputbw. Default value is 25%.

passphrase=string

HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The passphrase is the shared secret between the sender and the receiver. It is used to generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function). It is used only if pbkeylen is non-zero. It is used on the receiver only if the received data is encrypted. The configured passphrase cannot be recovered (write-only).

enforced_encryption=1|0

If true, both connection parties must have the same password set (including empty, that is, with no encryption). If the password doesn’t match or only one side is unencrypted, the connection is rejected. Default is true.

kmrefreshrate=packets

The number of packets to be transmitted after which the encryption key is switched to a new key. Default is -1. -1 means auto (0x1000000 in srt library). The range for this option is integers in the 0 - INT_MAX.

kmpreannounce=packets

The interval between when a new encryption key is sent and when switchover occurs. This value also applies to the subsequent interval between when switchover occurs and when the old encryption key is decommissioned. Default is -1. -1 means auto (0x1000 in srt library). The range for this option is integers in the 0 - INT_MAX.

snddropdelay=microseconds

The sender’s extra delay before dropping packets. This delay is added to the default drop delay time interval value.

Special value -1: Do not drop packets on the sender at all.

payload_size=bytes

Sets the maximum declared size of a packet transferred during the single call to the sending function in Live mode. Use 0 if this value isn’t used (which is default in file mode). Default is -1 (automatic), which typically means MPEG-TS; if you are going to use SRT to send any different kind of payload, such as, for example, wrapping a live stream in very small frames, then you can use a bigger maximum frame size, though not greater than 1456 bytes.

pkt_size=bytes

Alias for ‘payload_size’.

peerlatency=microseconds

The latency value (as described in rcvlatency) that is set by the sender side as a minimum value for the receiver.

pbkeylen=bytes

Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. Enable sender encryption if not 0. Not required on receiver (set to 0), key size obtained from sender in HaiCrypt handshake. Default value is 0.

rcvlatency=microseconds

The time that should elapse since the moment when the packet was sent and the moment when it’s delivered to the receiver application in the receiving function. This time should be a buffer time large enough to cover the time spent for sending, unexpectedly extended RTT time, and the time needed to retransmit the lost UDP packet. The effective latency value will be the maximum of this options’ value and the value of peerlatency set by the peer side. Before version 1.3.0 this option is only available as latency.

recv_buffer_size=bytes

Set UDP receive buffer size, expressed in bytes.

send_buffer_size=bytes

Set UDP send buffer size, expressed in bytes.

timeout=microseconds

Set raise error timeouts for read, write and connect operations. Note that the SRT library has internal timeouts which can be controlled separately, the value set here is only a cap on those.

tlpktdrop=1|0

Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not been delivered in time and delivers the following packets to the application when their time-to-play has come. It also sends a fake ACK to the sender. When enabled on sender and enabled on the receiving peer, the sender drops the older packets that have no chance of being delivered in time. It was automatically enabled in the sender if the receiver supports it.

sndbuf=bytes

Set send buffer size, expressed in bytes.

rcvbuf=bytes

Set receive buffer size, expressed in bytes.

Receive buffer must not be greater than ffs.

lossmaxttl=packets

The value up to which the Reorder Tolerance may grow. When Reorder Tolerance is > 0, then packet loss report is delayed until that number of packets come in. Reorder Tolerance increases every time a "belated" packet has come, but it wasn’t due to retransmission (that is, when UDP packets tend to come out of order), with the difference between the latest sequence and this packet’s sequence, and not more than the value of this option. By default it’s 0, which means that this mechanism is turned off, and the loss report is always sent immediately upon experiencing a "gap" in sequences.

minversion

The minimum SRT version that is required from the peer. A connection to a peer that does not satisfy the minimum version requirement will be rejected.

The version format in hex is 0xXXYYZZ for x.y.z in human readable form.

streamid=string

A string limited to 512 characters that can be set on the socket prior to connecting. This stream ID will be able to be retrieved by the listener side from the socket that is returned from srt_accept and was connected by a socket with that set stream ID. SRT does not enforce any special interpretation of the contents of this string. This option doesn’t make sense in Rendezvous connection; the result might be that simply one side will override the value from the other side and it’s the matter of luck which one would win

srt_streamid=string

Alias for ‘streamid’ to avoid conflict with ffmpeg command line option.

smoother=live|file

The type of Smoother used for the transmission for that socket, which is responsible for the transmission and congestion control. The Smoother type must be exactly the same on both connecting parties, otherwise the connection is rejected.

messageapi=1|0

When set, this socket uses the Message API, otherwise it uses Buffer API. Note that in live mode (see transtype) there’s only message API available. In File mode you can chose to use one of two modes:

Stream API (default, when this option is false). In this mode you may send as many data as you wish with one sending instruction, or even use dedicated functions that read directly from a file. The internal facility will take care of any speed and congestion control. When receiving, you can also receive as many data as desired, the data not extracted will be waiting for the next call. There is no boundary between data portions in the Stream mode.

Message API. In this mode your single sending instruction passes exactly one piece of data that has boundaries (a message). Contrary to Live mode, this message may span across multiple UDP packets and the only size limitation is that it shall fit as a whole in the sending buffer. The receiver shall use as large buffer as necessary to receive the message, otherwise the message will not be given up. When the message is not complete (not all packets received or there was a packet loss) it will not be given up.

transtype=live|file

Sets the transmission type for the socket, in particular, setting this option sets multiple other parameters to their default values as required for a particular transmission type.

live: Set options as for live transmission. In this mode, you should send by one sending instruction only so many data that fit in one UDP packet, and limited to the value defined first in payload_size (1316 is default in this mode). There is no speed control in this mode, only the bandwidth control, if configured, in order to not exceed the bandwidth with the overhead transmission (retransmitted and control packets).

file: Set options as for non-live transmission. See messageapi for further explanations

linger=seconds

The number of seconds that the socket waits for unsent data when closing. Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180 seconds in file mode). The range for this option is integers in the 0 - INT_MAX.

tsbpd=1|0

When true, use Timestamp-based Packet Delivery mode. The default behavior depends on the transmission type: enabled in live mode, disabled in file mode.

For more information see: https://github.com/Haivision/srt.

24.38 srtp

Secure Real-time Transport Protocol.

The accepted options are:

srtp_in_suite
srtp_out_suite

Select input and output encoding suites.

Supported values:

AES_CM_128_HMAC_SHA1_80
SRTP_AES128_CM_HMAC_SHA1_80
AES_CM_128_HMAC_SHA1_32
SRTP_AES128_CM_HMAC_SHA1_32
srtp_in_params
srtp_out_params

Set input and output encoding parameters, which are expressed by a base64-encoded representation of a binary block. The first 16 bytes of this binary block are used as master key, the following 14 bytes are used as master salt.

24.39 subfile

Virtually extract a segment of a file or another stream. The underlying stream must be seekable.

Accepted options:

start

Start offset of the extracted segment, in bytes.

end

End offset of the extracted segment, in bytes. If set to 0, extract till end of file.

Examples:

Extract a chapter from a DVD VOB file (start and end sectors obtained externally and multiplied by 2048):

subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

Play an AVI file directly from a TAR archive:

subfile,,start,183241728,end,366490624,,:archive.tar

Play a MPEG-TS file from start offset till end:

subfile,,start,32815239,end,0,,:video.ts

24.40 tee

Writes the output to multiple protocols. The individual outputs are separated by |

tee:file://path/to/local/this.avi|file://path/to/local/that.avi

24.41 tcp

Transmission Control Protocol.

The required syntax for a TCP url is:

tcp://hostname:port[?options]

options contains a list of &-separated options of the form key=val.

The list of supported options follows.

listen=2|1|0

Listen for an incoming connection. 0 disables listen, 1 enables listen in single client mode, 2 enables listen in multi-client mode. Default value is 0.

local_addr=addr

Local IP address of a network interface used for tcp socket connect.

local_port=port

Local port used for tcp socket connect.

timeout=microseconds

Set raise error timeout, expressed in microseconds.

This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.

listen_timeout=milliseconds

Set listen timeout, expressed in milliseconds.

recv_buffer_size=bytes

Set receive buffer size, expressed bytes.

send_buffer_size=bytes

Set send buffer size, expressed bytes.

tcp_nodelay=1|0

Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0.

Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.

tcp_mss=bytes

Set maximum segment size for outgoing TCP packets, expressed in bytes.

The following example shows how to setup a listening TCP connection with ffmpeg, which is then accessed with ffplay:

ffmpeg -i input -f format tcp://hostname:port?listen
ffplay tcp://hostname:port

24.42 tls

Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

The required syntax for a TLS/SSL url is:

tls://hostname:port[?options]

The following parameters can be set via command line options (or in code via AVOptions):

ca_file, cafile=filename

A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS library contains a default this might not need to be specified for verification to work, but not all libraries and setups have defaults built in. The file must be in OpenSSL PEM format.

tls_verify=1|0

If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With other backends, the host name is validated as well.)

This is disabled by default since it requires a CA database to be provided by the caller in many cases.

cert_file, cert=filename

A file containing a certificate to use in the handshake with the peer. (When operating as server, in listen mode, this is more often required by the peer, while client certificates only are mandated in certain setups.)

key_file, key=filename

A file containing the private key for the certificate.

listen=1|0

If enabled, listen for connections on the provided port, and assume the server role in the handshake instead of the client role.

http_proxy

The HTTP proxy to tunnel through, e.g. http://example.com:1234. The proxy must support the CONNECT method.

Example command lines:

To create a TLS/SSL server that serves an input stream.

ffmpeg -i input -f format tls://hostname:port?listen&cert=server.crt&key=server.key

To play back a stream from the TLS/SSL server using ffplay:

ffplay tls://hostname:port

24.43 udp

User Datagram Protocol.

The required syntax for an UDP URL is:

udp://hostname:port[?options]

options contains a list of &-separated options of the form key=val.

In case threading is enabled on the system, a circular buffer is used to store the incoming data, which allows one to reduce loss of data due to UDP socket buffer overruns. The fifo_size and overrun_nonfatal options are related to this buffer.

The list of supported options follows.

buffer_size=size

Set the UDP maximum socket buffer size in bytes. This is used to set either the receive or send buffer size, depending on what the socket is used for. Default is 32 KB for output, 384 KB for input. See also fifo_size.

bitrate=bitrate

If set to nonzero, the output will have the specified constant bitrate if the input has enough packets to sustain it.

burst_bits=bits

When using bitrate this specifies the maximum number of bits in packet bursts.

localport=port

Override the local UDP port to bind with.

localaddr=addr

Local IP address of a network interface used for sending packets or joining multicast groups.

pkt_size=size

Set the size in bytes of UDP packets.

reuse=1|0

Explicitly allow or disallow reusing UDP sockets.

ttl=ttl

Set the time to live value (for multicast only).

connect=1|0

Initialize the UDP socket with connect(). In this case, the destination address can’t be changed with ff_udp_set_remote_url later. If the destination address isn’t known at the start, this option can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if "destination unreachable" is received. For receiving, this gives the benefit of only receiving packets from the specified peer address/port.

sources=address[,address]

Only receive packets sent from the specified addresses. In case of multicast, also subscribe to multicast traffic coming from these addresses only.

block=address[,address]

Ignore packets sent from the specified addresses. In case of multicast, also exclude the source addresses in the multicast subscription.

fifo_size=units

Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes. If not specified defaults to 7*4096.

overrun_nonfatal=1|0

Survive in case of UDP receiving circular buffer overrun. Default value is 0.

timeout=microseconds

Set raise error timeout, expressed in microseconds.

This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.

broadcast=1|0

Explicitly allow or disallow UDP broadcasting.

Note that broadcasting may not work properly on networks having a broadcast storm protection.

24.43.1 Examples

  • Use ffmpeg to stream over UDP to a remote endpoint:
    ffmpeg -i input -f format udp://hostname:port
    
  • Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
    ffmpeg -i input -f mpegts udp://hostname:port?pkt_size=188&buffer_size=65535
    
  • Use ffmpeg to receive over UDP from a remote endpoint:
    ffmpeg -i udp://[multicast-address]:port ...
    

24.44 unix

Unix local socket

The required syntax for a Unix socket URL is:

unix://filepath

The following parameters can be set via command line options (or in code via AVOptions):

timeout

Timeout in ms.

listen

Create the Unix socket in listening mode.

24.45 zmq

ZeroMQ asynchronous messaging using the libzmq library.

This library supports unicast streaming to multiple clients without relying on an external server.

The required syntax for streaming or connecting to a stream is:

zmq:tcp://ip-address:port

Example: Create a localhost stream on port 5555:

ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555

Multiple clients may connect to the stream using:

ffplay zmq:tcp://127.0.0.1:5555

Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern. The server side binds to a port and publishes data. Clients connect to the server (via IP address/port) and subscribe to the stream. The order in which the server and client start generally does not matter.

ffmpeg must be compiled with the –enable-libzmq option to support this protocol.

Options can be set on the ffmpeg/ffplay command line. The following options are supported:

pkt_size

Forces the maximum packet size for sending/receiving data. The default value is 131,072 bytes. On the server side, this sets the maximum size of sent packets via ZeroMQ. On the clients, it sets an internal buffer size for receiving packets. Note that pkt_size on the clients should be equal to or greater than pkt_size on the server. Otherwise the received message may be truncated causing decoding errors.

25 Device Options

The libavdevice library provides the same interface as libavformat. Namely, an input device is considered like a demuxer, and an output device like a muxer, and the interface and generic device options are the same provided by libavformat (see the ffmpeg-formats manual).

In addition each input or output device may support so-called private options, which are specific for that component.

Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in the device AVFormatContext options or using the libavutil/opt.h API for programmatic use.

26 Input Devices

Input devices are configured elements in FFmpeg which enable accessing the data coming from a multimedia device attached to your system.

When you configure your FFmpeg build, all the supported input devices are enabled by default. You can list all available ones using the configure option "–list-indevs".

You can disable all the input devices using the configure option "–disable-indevs", and selectively enable an input device using the option "–enable-indev=INDEV", or you can disable a particular input device using the option "–disable-indev=INDEV".

The option "-devices" of the ff* tools will display the list of supported input devices.

A description of the currently available input devices follows.

26.1 alsa

ALSA (Advanced Linux Sound Architecture) input device.

To enable this input device during configuration you need libasound installed on your system.

This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA card identifier.

An ALSA identifier has the syntax:

hw:CARD[,DEV[,SUBDEV]]

where the DEV and SUBDEV components are optional.

The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device number and subdevice number (-1 means any).

To see the list of cards currently recognized by your system check the files /proc/asound/cards and /proc/asound/devices.

For example to capture with ffmpeg from an ALSA device with card id 0, you may run the command:

ffmpeg -f alsa -i hw:0 alsaout.wav

For more information see: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html

26.1.1 Options

sample_rate

Set the sample rate in Hz. Default is 48000.

channels

Set the number of channels. Default is 2.

26.2 android_camera

Android camera input device.

This input devices uses the Android Camera2 NDK API which is available on devices with API level 24+. The availability of android_camera is autodetected during configuration.

This device allows capturing from all cameras on an Android device, which are integrated into the Camera2 NDK API.

The available cameras are enumerated internally and can be selected with the camera_index parameter. The input file string is discarded.

Generally the back facing camera has index 0 while the front facing camera has index 1.

26.2.1 Options

video_size

Set the video size given as a string such as 640x480 or hd720. Falls back to the first available configuration reported by Android if requested video size is not available or by default.

framerate

Set the video framerate. Falls back to the first available configuration reported by Android if requested framerate is not available or by default (-1).

camera_index

Set the index of the camera to use. Default is 0.

input_queue_size

Set the maximum number of frames to buffer. Default is 5.

26.3 avfoundation

AVFoundation input device.

AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.

The input filename has to be given in the following syntax:

-i "[[VIDEO]:[AUDIO]]"

The first entry selects the video input while the latter selects the audio input. The stream has to be specified by the device name or the device index as shown by the device list. Alternatively, the video and/or audio input device can be chosen by index using the -video_device_index <INDEX> and/or -audio_device_index <INDEX> , overriding any device name or index given in the input filename.

All available devices can be enumerated by using -list_devices true, listing all device names and corresponding indices.

There are two device name aliases:

default

Select the AVFoundation default device of the corresponding type.

none

Do not record the corresponding media type. This is equivalent to specifying an empty device name or index.

26.3.1 Options

AVFoundation supports the following options:

-list_devices <TRUE|FALSE>

If set to true, a list of all available input devices is given showing all device names and indices.

-video_device_index <INDEX>

Specify the video device by its index. Overrides anything given in the input filename.

-audio_device_index <INDEX>

Specify the audio device by its index. Overrides anything given in the input filename.

-pixel_format <FORMAT>

Request the video device to use a specific pixel format. If the specified format is not supported, a list of available formats is given and the first one in this list is used instead. Available pixel formats are: monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0, bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10, yuv420p, nv12, yuyv422, gray

-framerate

Set the grabbing frame rate. Default is ntsc, corresponding to a frame rate of 30000/1001.

-video_size

Set the video frame size.

-capture_cursor

Capture the mouse pointer. Default is 0.

-capture_mouse_clicks

Capture the screen mouse clicks. Default is 0.

-capture_raw_data

Capture the raw device data. Default is 0. Using this option may result in receiving the underlying data delivered to the AVFoundation framework. E.g. for muxed devices that sends raw DV data to the framework (like tape-based camcorders), setting this option to false results in extracted video frames captured in the designated pixel format only. Setting this option to true results in receiving the raw DV stream untouched.

26.3.2 Examples

  • Print the list of AVFoundation supported devices and exit:
    $ ffmpeg -f avfoundation -list_devices true -i ""
    
  • Record video from video device 0 and audio from audio device 0 into out.avi:
    $ ffmpeg -f avfoundation -i "0:0" out.avi
    
  • Record video from video device 2 and audio from audio device 1 into out.avi:
    $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
    
  • Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi:
    $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
    
  • Record raw DV data from a suitable input device and write the output into out.dv:
    $ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
    

26.4 bktr

BSD video input device. Deprecated and will be removed - please contact the developers if you are interested in maintaining it.

26.4.1 Options

framerate

Set the frame rate.

video_size

Set the video frame size. Default is vga.

standard

Available values are:

pal
ntsc
secam
paln
palm
ntscj

26.5 decklink

The decklink input device provides capture capabilities for Blackmagic DeckLink devices.

To enable this input device, you need the Blackmagic DeckLink SDK and you need to configure with the appropriate --extra-cflags and --extra-ldflags. On Windows, you need to run the IDL files through widl.

DeckLink is very picky about the formats it supports. Pixel format of the input can be set with raw_format. Framerate and video size must be determined for your device with -list_formats 1. Audio sample rate is always 48 kHz and the number of channels can be 2, 8 or 16. Note that all audio channels are bundled in one single audio track.

26.5.1 Options

list_devices

If set to true, print a list of devices and exit. Defaults to false. This option is deprecated, please use the -sources option of ffmpeg to list the available input devices.

list_formats

If set to true, print a list of supported formats and exit. Defaults to false.

format_code <FourCC>

This sets the input video format to the format given by the FourCC. To see the supported values of your device(s) use list_formats. Note that there is a FourCC 'pal ' that can also be used as pal (3 letters). Default behavior is autodetection of the input video format, if the hardware supports it.

raw_format

Set the pixel format of the captured video. Available values are:

auto

This is the default which means 8-bit YUV 422 or 8-bit ARGB if format autodetection is used, 8-bit YUV 422 otherwise.

uyvy422

8-bit YUV 422.

yuv422p10

10-bit YUV 422.

argb

8-bit RGB.

bgra

8-bit RGB.

rgb10

10-bit RGB.

teletext_lines

If set to nonzero, an additional teletext stream will be captured from the vertical ancillary data. Both SD PAL (576i) and HD (1080i or 1080p) sources are supported. In case of HD sources, OP47 packets are decoded.

This option is a bitmask of the SD PAL VBI lines captured, specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB in the mask. Selected lines which do not contain teletext information will be ignored. You can use the special all constant to select all possible lines, or standard to skip lines 6, 318 and 319, which are not compatible with all receivers.

For SD sources, ffmpeg needs to be compiled with --enable-libzvbi. For HD sources, on older (pre-4K) DeckLink card models you have to capture in 10 bit mode.

channels

Defines number of audio channels to capture. Must be ‘2’, ‘8’ or ‘16’. Defaults to ‘2’.

duplex_mode

Sets the decklink device duplex/profile mode. Must be ‘unset’, ‘half’, ‘full’, ‘one_sub_device_full’, ‘one_sub_device_half’, ‘two_sub_device_full’, ‘four_sub_device_half’ Defaults to ‘unset’.

Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property. For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2 sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.

Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0): ‘one_sub_device_full’, ‘one_sub_device_half’, ‘two_sub_device_full’, ‘four_sub_device_half

Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: ‘half’, ‘full

timecode_format

Timecode type to include in the frame and video stream metadata. Must be ‘none’, ‘rp188vitc’, ‘rp188vitc2’, ‘rp188ltc’, ‘rp188hfr’, ‘rp188any’, ‘vitc’, ‘vitc2’, or ‘serial’. Defaults to ‘none’ (not included).

In order to properly support 50/60 fps timecodes, the ordering of the queried timecode types for ‘rp188any’ is HFR, VITC1, VITC2 and LTC for >30 fps content. Note that this is slightly different to the ordering used by the DeckLink API, which is HFR, VITC1, LTC, VITC2.

video_input

Sets the video input source. Must be ‘unset’, ‘sdi’, ‘hdmi’, ‘optical_sdi’, ‘component’, ‘composite’ or ‘s_video’. Defaults to ‘unset’.

audio_input

Sets the audio input source. Must be ‘unset’, ‘embedded’, ‘aes_ebu’, ‘analog’, ‘analog_xlr’, ‘analog_rca’ or ‘microphone’. Defaults to ‘unset’.

video_pts

Sets the video packet timestamp source. Must be ‘video’, ‘audio’, ‘reference’, ‘wallclock’ or ‘abs_wallclock’. Defaults to ‘video’.

audio_pts

Sets the audio packet timestamp source. Must be ‘video’, ‘audio’, ‘reference’, ‘wallclock’ or ‘abs_wallclock’. Defaults to ‘audio’.

draw_bars

If set to ‘true’, color bars are drawn in the event of a signal loss. Defaults to ‘true’. This option is deprecated, please use the signal_loss_action option.

signal_loss_action

Sets the action to take in the event of a signal loss. Accepts one of the following values:

1, none

Do nothing on signal loss. This usually results in black frames.

2, bars

Draw color bars on signal loss. Only supported for 8-bit input signals.

3, repeat

Repeat the last video frame on signal loss.

Defaults to ‘bars’.

queue_size

Sets maximum input buffer size in bytes. If the buffering reaches this value, incoming frames will be dropped. Defaults to ‘1073741824’.

audio_depth

Sets the audio sample bit depth. Must be ‘16’ or ‘32’. Defaults to ‘16’.

decklink_copyts

If set to true, timestamps are forwarded as they are without removing the initial offset. Defaults to false.

timestamp_align

Capture start time alignment in seconds. If set to nonzero, input frames are dropped till the system timestamp aligns with configured value. Alignment difference of up to one frame duration is tolerated. This is useful for maintaining input synchronization across N different hardware devices deployed for ’N-way’ redundancy. The system time of different hardware devices should be synchronized with protocols such as NTP or PTP, before using this option. Note that this method is not foolproof. In some border cases input synchronization may not happen due to thread scheduling jitters in the OS. Either sync could go wrong by 1 frame or in a rarer case timestamp_align seconds. Defaults to ‘0’.

wait_for_tc (bool)

Drop frames till a frame with timecode is received. Sometimes serial timecode isn’t received with the first input frame. If that happens, the stored stream timecode will be inaccurate. If this option is set to true, input frames are dropped till a frame with timecode is received. Option timecode_format must be specified. Defaults to false.

enable_klv(bool)

If set to true, extracts KLV data from VANC and outputs KLV packets. KLV VANC packets are joined based on MID and PSC fields and aggregated into one KLV packet. Defaults to false.

26.5.2 Examples

  • List input devices:
    ffmpeg -sources decklink
    
  • List supported formats:
    ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
    
  • Capture video clip at 1080i50:
    ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
    
  • Capture video clip at 1080i50 10 bit:
    ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
    
  • Capture video clip at 1080i50 with 16 audio channels:
    ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
    

26.6 dshow

Windows DirectShow input device.

DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. Currently only audio and video devices are supported.

Multiple devices may be opened as separate inputs, but they may also be opened on the same input, which should improve synchronism between them.

The input name should be in the format:

TYPE=NAME[:TYPE=NAME]

where TYPE can be either audio or video, and NAME is the device’s name or alternative name..

26.6.1 Options

If no options are specified, the device’s defaults are used. If the device does not support the requested options, it will fail to open.

video_size

Set the video size in the captured video.

framerate

Set the frame rate in the captured video.

sample_rate

Set the sample rate (in Hz) of the captured audio.

sample_size

Set the sample size (in bits) of the captured audio.

channels

Set the number of channels in the captured audio.

list_devices

If set to true, print a list of devices and exit.

list_options

If set to true, print a list of selected device’s options and exit.

video_device_number

Set video device number for devices with the same name (starts at 0, defaults to 0).

audio_device_number

Set audio device number for devices with the same name (starts at 0, defaults to 0).

pixel_format

Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo.

audio_buffer_size

Set audio device buffer size in milliseconds (which can directly impact latency, depending on the device). Defaults to using the audio device’s default buffer size (typically some multiple of 500ms). Setting this value too low can degrade performance. See also http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx

video_pin_name

Select video capture pin to use by name or alternative name.

audio_pin_name

Select audio capture pin to use by name or alternative name.

crossbar_video_input_pin_number

Select video input pin number for crossbar device. This will be routed to the crossbar device’s Video Decoder output pin. Note that changing this value can affect future invocations (sets a new default) until system reboot occurs.

crossbar_audio_input_pin_number

Select audio input pin number for crossbar device. This will be routed to the crossbar device’s Audio Decoder output pin. Note that changing this value can affect future invocations (sets a new default) until system reboot occurs.

show_video_device_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to change video filter properties and configurations manually. Note that for crossbar devices, adjusting values in this dialog may be needed at times to toggle between PAL (25 fps) and NTSC (29.97) input frame rates, sizes, interlacing, etc. Changing these values can enable different scan rates/frame rates and avoiding green bars at the bottom, flickering scan lines, etc. Note that with some devices, changing these properties can also affect future invocations (sets new defaults) until system reboot occurs.

show_audio_device_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to change audio filter properties and configurations manually.

show_video_crossbar_connection_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify crossbar pin routings, when it opens a video device.

show_audio_crossbar_connection_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify crossbar pin routings, when it opens an audio device.

show_analog_tv_tuner_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV channels and frequencies.

show_analog_tv_tuner_audio_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV audio (like mono vs. stereo, Language A,B or C).

audio_device_load

Load an audio capture filter device from file instead of searching it by name. It may load additional parameters too, if the filter supports the serialization of its properties to. To use this an audio capture source has to be specified, but it can be anything even fake one.

audio_device_save

Save the currently used audio capture filter device and its parameters (if the filter supports it) to a file. If a file with the same name exists it will be overwritten.

video_device_load

Load a video capture filter device from file instead of searching it by name. It may load additional parameters too, if the filter supports the serialization of its properties to. To use this a video capture source has to be specified, but it can be anything even fake one.

video_device_save

Save the currently used video capture filter device and its parameters (if the filter supports it) to a file. If a file with the same name exists it will be overwritten.

use_video_device_timestamps

If set to false, the timestamp for video frames will be derived from the wallclock instead of the timestamp provided by the capture device. This allows working around devices that provide unreliable timestamps.

26.6.2 Examples

  • Print the list of DirectShow supported devices and exit:
    $ ffmpeg -list_devices true -f dshow -i dummy
    
  • Open video device Camera:
    $ ffmpeg -f dshow -i video="Camera"
    
  • Open second video device with name Camera:
    $ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
    
  • Open video device Camera and audio device Microphone:
    $ ffmpeg -f dshow -i video="Camera":audio="Microphone"
    
  • Print the list of supported options in selected device and exit:
    $ ffmpeg -list_options true -f dshow -i video="Camera"
    
  • Specify pin names to capture by name or alternative name, specify alternative device name:
    $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"
    
  • Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties at startup:
    $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
         -crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
    

26.7 fbdev

Linux framebuffer input device.

The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually /dev/fb0.

For more detailed information read the file Documentation/fb/framebuffer.txt included in the Linux source tree.

See also http://linux-fbdev.sourceforge.net/, and fbset(1).

To record from the framebuffer device /dev/fb0 with ffmpeg:

ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi

You can take a single screenshot image with the command:

ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg

26.7.1 Options

framerate

Set the frame rate. Default is 25.

26.8 gdigrab

Win32 GDI-based screen capture device.

This device allows you to capture a region of the display on Windows.

Amongst options for the imput filenames are such elements as:

desktop

or

title=window_title

or

hwnd=window_hwnd

The first option will capture the entire desktop, or a fixed region of the desktop. The second and third options will instead capture the contents of a single window, regardless of its position on the screen.

For example, to grab the entire desktop using ffmpeg:

ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

Grab a 640x480 region at position 10,20:

ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg

Grab the contents of the window named "Calculator"

ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

26.8.1 Options

draw_mouse

Specify whether to draw the mouse pointer. Use the value 0 to not draw the pointer. Default value is 1.

framerate

Set the grabbing frame rate. Default value is ntsc, corresponding to a frame rate of 30000/1001.

show_region

Show grabbed region on screen.

If show_region is specified with 1, then the grabbing region will be indicated on screen. With this option, it is easy to know what is being grabbed if only a portion of the screen is grabbed.

Note that show_region is incompatible with grabbing the contents of a single window.

For example:

ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg
video_size

Set the video frame size. The default is to capture the full screen if desktop is selected, or the full window size if title=window_title is selected.

offset_x

When capturing a region with video_size, set the distance from the left edge of the screen or desktop.

Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned to the left of your primary monitor, you will need to use a negative offset_x value to move the region to that monitor.

offset_y

When capturing a region with video_size, set the distance from the top edge of the screen or desktop.

Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned above your primary monitor, you will need to use a negative offset_y value to move the region to that monitor.

26.9 iec61883

FireWire DV/HDV input device using libiec61883.

To enable this input device, you need libiec61883, libraw1394 and libavc1394 installed on your system. Use the configure option --enable-libiec61883 to compile with the device enabled.

The iec61883 capture device supports capturing from a video device connected via IEEE1394 (FireWire), using libiec61883 and the new Linux FireWire stack (juju). This is the default DV/HDV input method in Linux Kernel 2.6.37 and later, since the old FireWire stack was removed.

Specify the FireWire port to be used as input file, or "auto" to choose the first port connected.

26.9.1 Options

dvtype

Override autodetection of DV/HDV. This should only be used if auto detection does not work, or if usage of a different device type should be prohibited. Treating a DV device as HDV (or vice versa) will not work and result in undefined behavior. The values auto, dv and hdv are supported.

dvbuffer

Set maximum size of buffer for incoming data, in frames. For DV, this is an exact value. For HDV, it is not frame exact, since HDV does not have a fixed frame size.

dvguid

Select the capture device by specifying its GUID. Capturing will only be performed from the specified device and fails if no device with the given GUID is found. This is useful to select the input if multiple devices are connected at the same time. Look at /sys/bus/firewire/devices to find out the GUIDs.

26.9.2 Examples

  • Grab and show the input of a FireWire DV/HDV device.
    ffplay -f iec61883 -i auto
    
  • Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000 packets if the source is HDV.
    ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
    

26.10 jack

JACK input device.

To enable this input device during configuration you need libjack installed on your system.

A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the FFmpeg input device.

Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients.

To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect programs, or do it through a graphical interface, for example with qjackctl.

To list the JACK clients and their properties you can invoke the command jack_lsp.

Follows an example which shows how to capture a JACK readable client with ffmpeg.

# Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav

# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000

# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm

# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1

For more information read: http://jackaudio.org/

26.10.1 Options

channels

Set the number of channels. Default is 2.

26.11 kmsgrab

KMS video input device.

Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a DRM object that can be passed to other hardware functions.

Requires either DRM master or CAP_SYS_ADMIN to run.

If you don’t understand what all of that means, you probably don’t want this. Look at x11grab instead.

26.11.1 Options

device

DRM device to capture on. Defaults to /dev/dri/card0.

format

Pixel format of the framebuffer. This can be autodetected if you are running Linux 5.7 or later, but needs to be provided for earlier versions. Defaults to bgr0, which is the most common format used by the Linux console and Xorg X server.

format_modifier

Format modifier to signal on output frames. This is necessary to import correctly into some APIs. It can be autodetected if you are running Linux 5.7 or later, but will need to be provided explicitly when needed in earlier versions. See the libdrm documentation for possible values.

crtc_id

KMS CRTC ID to define the capture source. The first active plane on the given CRTC will be used.

plane_id

KMS plane ID to define the capture source. Defaults to the first active plane found if neither crtc_id nor plane_id are specified.

framerate

Framerate to capture at. This is not synchronised to any page flipping or framebuffer changes - it just defines the interval at which the framebuffer is sampled. Sampling faster than the framebuffer update rate will generate independent frames with the same content. Defaults to 30.

26.11.2 Examples

  • Capture from the first active plane, download the result to normal frames and encode. This will only work if the framebuffer is both linear and mappable - if not, the result may be scrambled or fail to download.
    ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
    
  • Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
    ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
    
  • To capture only part of a plane the output can be cropped - this can be used to capture a single window, as long as it has a known absolute position and size. For example, to capture and encode the middle quarter of a 1920x1080 plane:
    ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
    

26.12 lavfi

Libavfilter input virtual device.

This input device reads data from the open output pads of a libavfilter filtergraph.

For each filtergraph open output, the input device will create a corresponding stream which is mapped to the generated output. The filtergraph is specified through the option graph.

26.12.1 Options

graph

Specify the filtergraph to use as input. Each video open output must be labelled by a unique string of the form "outN", where N is a number starting from 0 corresponding to the mapped input stream generated by the device. The first unlabelled output is automatically assigned to the "out0" label, but all the others need to be specified explicitly.

The suffix "+subcc" can be appended to the output label to create an extra stream with the closed captions packets attached to that output (experimental; only for EIA-608 / CEA-708 for now). The subcc streams are created after all the normal streams, in the order of the corresponding stream. For example, if there is "out19+subcc", "out7+subcc" and up to "out42", the stream #43 is subcc for stream #7 and stream #44 is subcc for stream #19.

If not specified defaults to the filename specified for the input device.

graph_file

Set the filename of the filtergraph to be read and sent to the other filters. Syntax of the filtergraph is the same as the one specified by the option graph.

dumpgraph

Dump graph to stderr.

26.12.2 Examples

  • Create a color video stream and play it back with ffplay:
    ffplay -f lavfi -graph "color=c=pink [out0]" dummy
    
  • As the previous example, but use filename for specifying the graph description, and omit the "out0" label:
    ffplay -f lavfi color=c=pink
    
  • Create three different video test filtered sources and play them:
    ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
    
  • Read an audio stream from a file using the amovie source and play it back with ffplay:
    ffplay -f lavfi "amovie=test.wav"
    
  • Read an audio stream and a video stream and play it back with ffplay:
    ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
    
  • Dump decoded frames to images and Closed Captions to an RCWT backup:
    ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rcwt subcc.bin
    

26.13 libcdio

Audio-CD input device based on libcdio.

To enable this input device during configuration you need libcdio installed on your system. It requires the configure option --enable-libcdio.

This device allows playing and grabbing from an Audio-CD.

For example to copy with ffmpeg the entire Audio-CD in /dev/sr0, you may run the command:

ffmpeg -f libcdio -i /dev/sr0 cd.wav

26.13.1 Options

speed

Set drive reading speed. Default value is 0.

The speed is specified CD-ROM speed units. The speed is set through the libcdio cdio_cddap_speed_set function. On many CD-ROM drives, specifying a value too large will result in using the fastest speed.

paranoia_mode

Set paranoia recovery mode flags. It accepts one of the following values:

disable
verify
overlap
neverskip
full

Default value is ‘disable’.

For more information about the available recovery modes, consult the paranoia project documentation.

26.14 libdc1394

IIDC1394 input device, based on libdc1394 and libraw1394.

Requires the configure option --enable-libdc1394.

26.14.1 Options

framerate

Set the frame rate. Default is ntsc, corresponding to a frame rate of 30000/1001.

pixel_format

Select the pixel format. Default is uyvy422.

video_size

Set the video size given as a string such as 640x480 or hd720. Default is qvga.

26.15 openal

The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1 implementation.

To enable this input device during configuration, you need OpenAL headers and libraries installed on your system, and need to configure FFmpeg with --enable-openal.

OpenAL headers and libraries should be provided as part of your OpenAL implementation, or as an additional download (an SDK). Depending on your installation you may need to specify additional flags via the --extra-cflags and --extra-ldflags for allowing the build system to locate the OpenAL headers and libraries.

An incomplete list of OpenAL implementations follows:

Creative

The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See http://openal.org/.

OpenAL Soft

Portable, open source (LGPL) software implementation. Includes backends for the most common sound APIs on the Windows, Linux, Solaris, and BSD operating systems. See http://kcat.strangesoft.net/openal.html.

Apple

OpenAL is part of Core Audio, the official Mac OS X Audio interface. See http://developer.apple.com/technologies/mac/audio-and-video.html

This device allows one to capture from an audio input device handled through OpenAL.

You need to specify the name of the device to capture in the provided filename. If the empty string is provided, the device will automatically select the default device. You can get the list of the supported devices by using the option list_devices.

26.15.1 Options

channels

Set the number of channels in the captured audio. Only the values 1 (monaural) and 2 (stereo) are currently supported. Defaults to 2.

sample_size

Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are currently supported. Defaults to 16.

sample_rate

Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k.

list_devices

If set to true, print a list of devices and exit. Defaults to false.

26.15.2 Examples

Print the list of OpenAL supported devices and exit:

$ ffmpeg -list_devices true -f openal -i dummy out.ogg

Capture from the OpenAL device DR-BT101 via PulseAudio:

$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg

Capture from the default device (note the empty string ” as filename):

$ ffmpeg -f openal -i '' out.ogg

Capture from two devices simultaneously, writing to two different files, within the same ffmpeg command:

$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg

Note: not all OpenAL implementations support multiple simultaneous capture - try the latest OpenAL Soft if the above does not work.

26.16 oss

Open Sound System input device.

The filename to provide to the input device is the device node representing the OSS input device, and is usually set to /dev/dsp.

For example to grab from /dev/dsp using ffmpeg use the command:

ffmpeg -f oss -i /dev/dsp /tmp/oss.wav

For more information about OSS see: http://manuals.opensound.com/usersguide/dsp.html

26.16.1 Options

sample_rate

Set the sample rate in Hz. Default is 48000.

channels

Set the number of channels. Default is 2.

26.17 pulse

PulseAudio input device.

To enable this output device you need to configure FFmpeg with --enable-libpulse.

The filename to provide to the input device is a source device or the string "default"

To list the PulseAudio source devices and their properties you can invoke the command pactl list sources.

More information about PulseAudio can be found on http://www.pulseaudio.org.

26.17.1 Options

server

Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not provided.

name

Specify the application name PulseAudio will use when showing active clients, by default it is the LIBAVFORMAT_IDENT string.

stream_name

Specify the stream name PulseAudio will use when showing active streams, by default it is "record".

sample_rate

Specify the samplerate in Hz, by default 48kHz is used.

channels

Specify the channels in use, by default 2 (stereo) is set.

frame_size

This option does nothing and is deprecated.

fragment_size

Specify the size in bytes of the minimal buffering fragment in PulseAudio, it will affect the audio latency. By default it is set to 50 ms amount of data.

wallclock

Set the initial PTS using the current time. Default is 1.

26.17.2 Examples

Record a stream from default device:

ffmpeg -f pulse -i default /tmp/pulse.wav

26.18 sndio

sndio input device.

To enable this input device during configuration you need libsndio installed on your system.

The filename to provide to the input device is the device node representing the sndio input device, and is usually set to /dev/audio0.

For example to grab from /dev/audio0 using ffmpeg use the command:

ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

26.18.1 Options

sample_rate

Set the sample rate in Hz. Default is 48000.

channels

Set the number of channels. Default is 2.

26.19 video4linux2, v4l2

Video4Linux2 input video device.

"v4l2" can be used as alias for "video4linux2".

If FFmpeg is built with v4l-utils support (by using the --enable-libv4l2 conf