diff --git a/index.html b/index.html index bac1f07..3a6e287 100644 --- a/index.html +++ b/index.html @@ -71,15 +71,15 @@

Basic structure of an FFmpeg command

At its basis, an FFmpeg command is relatively simple. After you have installed FFmpeg (see instructions here), the program is invoked simply by typing ffmpeg at the command prompt.

-

Subsequently, each instruction that you supply to FFmpeg is actually a pair: a flag, which designates the type of action you want to carry out; and then the specifics of that action. Flags are always prepended with a hyphen.

-

For example, in the instruction -i input_file.ext, the -i flag tells FFmpeg that you are supplying an input file, and input_file.ext states which file it is.

+

Subsequently, each instruction that you supply to FFmpeg is actually a pair: a flag, which designates the type of action you want to carry out; and then the specifics of that action. Flags are always prepended with a hyphen.

+

For example, in the instruction -i input_file.ext, the -i flag tells FFmpeg that you are supplying an input file, and input_file.ext states which file it is.

Likewise, in the instruction -c:v prores, the flag -c:v tells FFmpeg that you want to encode the video stream, and prores specifies which codec is to be used. (-c:v is shorthand for -codec:v/-codec:video).

A very basic FFmpeg command looks like this:

ffmpeg
starts the command
-
-i input_file.ext
path and name of the input file
-
-flag some_action
tell FFmpeg to do something, by supplying a valid flag and action
-
output_file.ext
path and name of the output file.
+
-i input_file.ext
path and name of the input file
+
-flag some_action
tell FFmpeg to do something, by supplying a valid flag and action
+
output_file.ext
path and name of the output file.
Because this is the last part of the command, the filename you type here does not have a flag designating it as the output file.
@@ -111,10 +111,10 @@

Filtergraphs

Many FFmpeg commands use filters that manipulate the video or audio stream in some way: for example, hflip to horizontally flip a video, or amerge to merge two or more audio tracks into a single stream.

The use of a filter is signalled by the flag -vf (video filter) or -af (audio filter), followed by the name and options of the filter itself. For example, take the convert colourspace command:

-

ffmpeg -i input_file -c:v libx264 -vf colormatrix=src:dst output_file -

Here, colormatrix is the filter used, with src and dst representing the source and destination colourspaces. This part following the -vf is a filtergraph.

+

ffmpeg -i input_file -c:v libx264 -vf colormatrix=src:dst output_file +

Here, colormatrix is the filter used, with src and dst representing the source and destination colourspaces. This part following the -vf is a filtergraph.

It is also possible to apply multiple filters to an input, which are sequenced together in the filtergraph. A chained set of filters is called a filter chain, and a filtergraph may include multiple filter chains. Filters in a filterchain are separated from each other by commas (,), and filterchains are separated from each other by semicolons (;). For example, take the inverse telecine command:

-

ffmpeg -i input_file -c:v libx264 -vf "fieldmatch,yadif,decimate" output_file

+

ffmpeg -i input_file -c:v libx264 -vf "fieldmatch,yadif,decimate" output_file

Here we have a filtergraph including one filter chain, which is made up of three video filters.

It is often prudent to enclose your filtergraph in quotation marks; this means that you can use spaces within the filtergraph. Using the inverse telecine example again, the following filter commands are all valid and equivalent:

-

To map all streams in the input file to the output file, use -map 0. However, note that not all container formats can include all stream types: for example, .mp4 cannot contain timecode.

+

To map all streams in the input file to the output file, use -map 0. However, note that not all container formats can include all stream types: for example, .mp4 cannot contain timecode.

When no mapping is specified in an ffmpeg command, the default for video files is to take just one video and one audio stream for the output: other stream types, such as timecode or subtitles, will not be copied to the output file by default. If multiple video or audio streams are present, the best quality one is automatically selected by FFmpeg.

For more information, check out the FFmpeg wiki Map page, and the official FFmpeg documentation on -map.

@@ -172,16 +172,16 @@

Rewrap a file

-

ffmpeg -i input_file.ext -c copy -map 0 output_file.ext

+

ffmpeg -i input_file.ext -c copy -map 0 output_file.ext

This script will rewrap a video file. It will create a new video video file where the inner content (the video, audio, and subtitle data) of the original file is unchanged, but these streams are rehoused within a different container format.

-

Note: rewrapping is also known as remuxing, short for re-multiplexing.

+

Note: rewrapping is also known as remuxing, short for re-multiplexing.

ffmpeg
starts the command
-
-i input_file.ext
path and name of the input file
+
-i input_file.ext
path and name of the input file
-c copy
copy the streams directly, without re-encoding.
-map 0
map all streams of the input to the output.
By default, FFmpeg will only map one stream of each type (video, audio, subtitles) to the output file. However, files may have multiple streams of a given type - for example, a video may have several audio tracks for different languages. Therefore, if you want to preserve all the streams in the original, it's necessary to use this option.
-
output_file.ext
path and name of the output file.
+
output_file.ext
path and name of the output file.
The new container you are rewrapping to is defined by the filename extension used here, e.g. .mkv, .mp4, .mov.

Important caveat

@@ -196,17 +196,17 @@

MKV to MP4

-

ffmpeg -i input_file.mkv -c:v copy -c:a aac output_file.mp4

+

ffmpeg -i input_file.mkv -c:v copy -c:a aac output_file.mp4

This will convert your Matroska (MKV) files to MP4 files.

ffmpeg
starts the command
-
-i input_file
path and name of the input file
+
-i input_file
path and name of the input file
The extension for the Matroska container is .mkv.
-c:v copy
copies the video stream without re-encoding it
-c:a aac
re-encodes the audio stream using the AAC audio codec
Note that sadly MP4 cannot contain sound encoded by a PCM (Pulse-Code Modulation) audio codec.
For silent videos you can replace -c:a aac by -an, which means that there will be no audio track in the output file.
-
output_file
path and name of the output file
+
output_file
path and name of the output file
The extension for the MP4 container is .mp4.
@@ -218,14 +218,14 @@

Rewrap DV video to .dv file

-

ffmpeg -i input_file -f rawvideo -c:v copy output_file.dv

+

ffmpeg -i input_file -f rawvideo -c:v copy output_file.dv

This script will take a video that is encoded in the DV Codec but wrapped in a different container (such as MOV) and rewrap it into a raw DV file (with the .dv extension). Since DV files potentially contain a great deal of provenance metadata within the DV stream, it is necessary to rewrap files in this method to avoid unintentional stripping of this metadata.

ffmpeg
starts the command
-
-i input_file
path and name of the input file
+
-i input_file
path and name of the input file
-f rawvideo
this tells FFmpeg to pass the video stream as raw video data without remuxing. This step is what ensures the survival of embedded metadata versus a standard rewrap.
-c:v copy
copy the DV stream directly, without re-encoding.
-
output_file.dv
tells FFmpeg to use the DV wrapper for the output.
+
output_file.dv
tells FFmpeg to use the DV wrapper for the output.
@@ -240,13 +240,13 @@

Transcode into a deinterlaced Apple ProRes LT

-

ffmpeg -i input_file -c:v prores -profile:v 1 -vf yadif -c:a pcm_s16le output_file.mov

+

ffmpeg -i input_file -c:v prores -profile:v 1 -vf yadif -c:a pcm_s16le output_file.mov

This command transcodes an input file into a deinterlaced Apple ProRes 422 LT file with 16-bit linear PCM encoded audio. The file is deinterlaced using the yadif filter (Yet Another De-Interlacing Filter).

ffmpeg
starts the command
-
-i input_file
path, name and extension of the input file
+
-i input_file
path, name and extension of the input file
-c:v prores
tells FFmpeg to transcode the video stream into Apple ProRes 422
-
-profile:v 1
Declares profile of ProRes you want to use. The profiles are explained below: +
-profile:v 1
Declares profile of ProRes you want to use. The profiles are explained below:
  • 0 = ProRes 422 (Proxy)
  • 1 = ProRes 422 (LT)
  • @@ -255,7 +255,7 @@
-vf yadif
Runs a deinterlacing video filter (yet another deinterlacing filter) on the new file. -vf is an alias for -filter:v.
-c:a pcm_s16le
tells FFmpeg to encode the audio stream in 16-bit linear PCM
-
output_file
path, name and extension of the output file
+
output_file
path, name and extension of the output file
The extension for the QuickTime container is .mov.

FFmpeg comes with more than one ProRes encoder:

@@ -272,23 +272,23 @@

Transcode to H.264

-

ffmpeg -i input_file -c:v libx264 -pix_fmt yuv420p -c:a aac output_file

+

ffmpeg -i input_file -c:v libx264 -pix_fmt yuv420p -c:a aac output_file

This command takes an input file and transcodes it to H.264 with an .mp4 wrapper, keeping the audio the same codec as the original. The libx264 codec defaults to a “medium” preset for compression quality and a CRF of 23. CRF stands for constant rate factor and determines the quality and file size of the resulting H.264 video. A low CRF means high quality and large file size; a high CRF means the opposite.

ffmpeg
starts the command
-
-i input_file
path, name and extension of the input file
+
-i input_file
path, name and extension of the input file
-c:v libx264
tells FFmpeg to encode the video stream as H.264
-pix_fmt yuv420p
libx264 will use a chroma subsampling scheme that is the closest match to that of the input. This can result in Y′CBCR 4:2:0, 4:2:2, or 4:4:4 chroma subsampling. QuickTime and most other non-FFmpeg based players can’t decode H.264 files that are not 4:2:0. In order to allow the video to play in all players, you can specify 4:2:0 chroma subsampling.
-c:a aac
encode audio as AAC.
AAC is the codec most often used for audio streams within an .mp4 container.
-
output_file
path, name and extension of the output file
+
output_file
path, name and extension of the output file

In order to use the same basic command to make a higher quality file, you can add some of these presets:

-

ffmpeg -i input_file -c:v libx264 -pix_fmt yuv420p -preset veryslow -crf 18 -c:a aac output_file

+

ffmpeg -i input_file -c:v libx264 -pix_fmt yuv420p -preset veryslow -crf 18 -c:a aac output_file

-
-preset veryslow
This option tells FFmpeg to use the slowest preset possible for the best compression quality.
+
-preset veryslow
This option tells FFmpeg to use the slowest preset possible for the best compression quality.
Available presets, from slowest to fastest, are: veryslow, slower, slow, medium, fast, faster, veryfast, superfast, ultrafast.
-
-crf 18
Specifying a lower CRF will make a larger file with better visual quality. For H.264 files being encoded with a 4:2:0 chroma subsampling scheme (i.e., using -pix_fmt yuv420p), the scale ranges between 0-51, with 0 being lossless and 51 the worst possible quality.
+
-crf 18
Specifying a lower CRF will make a larger file with better visual quality. For H.264 files being encoded with a 4:2:0 chroma subsampling scheme (i.e., using -pix_fmt yuv420p), the scale ranges between 0-51, with 0 being lossless and 51 the worst possible quality.
If no crf is specified, libx264 will use a default value of 23. 18 is often considered a “visually lossless” compression.

For more information, see the FFmpeg and H.264 Encoding Guide on the FFmpeg wiki.

@@ -301,23 +301,23 @@

H.264 from DCP

-

ffmpeg -i input_video_file.mxf -i input_audio_file.mxf -c:v libx264 -pix_fmt yuv420p -c:a aac output_file.mp4

+

ffmpeg -i input_video_file.mxf -i input_audio_file.mxf -c:v libx264 -pix_fmt yuv420p -c:a aac output_file.mp4

This will transcode MXF wrapped video and audio files to an H.264 encoded MP4 file. Please note this only works for unencrypted, single reel DCPs.

ffmpeg
starts the command
-
-i input_video_file
path and name of the video input file. This extension must be .mxf
-
-i input_audio_file
path and name of the audio input file. This extension must be .mxf
+
-i input_video_file
path and name of the video input file. This extension must be .mxf
+
-i input_audio_file
path and name of the audio input file. This extension must be .mxf
-c:v libx264
transcodes video to H.264
-pix_fmt yuv420p
sets pixel format to yuv420p for greater compatibility with media players
-c:a aac
re-encodes using the AAC audio codec
Note that sadly MP4 cannot contain sound encoded by a PCM (Pulse-Code Modulation) audio codec
-
output_file.mp4
path, name and .mp4 extension of the output file
+
output_file.mp4
path, name and .mp4 extension of the output file

Variation: Copy PCM audio streams by using Matroska instead of the MP4 container

-

ffmpeg -i input_video_file.mxf -i input_audio_file.mxf -c:v libx264 -pix_fmt yuv420p -c:a copy output_file.mkv

+

ffmpeg -i input_video_file.mxf -i input_audio_file.mxf -c:v libx264 -pix_fmt yuv420p -c:a copy output_file.mkv

-c:a copy
re-encodes using the same audio codec
-
output_file.mkv
path, name and .mkv extension of the output file
+
output_file.mkv
path, name and .mkv extension of the output file
@@ -328,11 +328,11 @@

Create FFV1 Version 3 video in a Matroska container with framemd5 of input

-

ffmpeg -i input_file -map 0 -dn -c:v ffv1 -level 3 -g 1 -slicecrc 1 -slices 16 -c:a copy output_file.mkv -f framemd5 -an framemd5_output_file

+

ffmpeg -i input_file -map 0 -dn -c:v ffv1 -level 3 -g 1 -slicecrc 1 -slices 16 -c:a copy output_file.mkv -f framemd5 -an framemd5_output_file

This will losslessly transcode your video with the FFV1 Version 3 codec in a Matroska container. In order to verify losslessness, a framemd5 of the source video is also generated. For more information on FFV1 encoding, try the FFmpeg wiki.

ffmpeg
starts the command.
-
-i input_file
path, name and extension of the input file.
+
-i input_file
path, name and extension of the input file.
-map 0
Map all streams that are present in the input file. This is important as FFmpeg will map only one stream of each type (video, audio, subtitles) by default to the output video.
-dn
ignore data streams (data no). The Matroska container does not allow data tracks.
-c:v ffv1
specifies the FFV1 video codec.
@@ -341,10 +341,10 @@
-slicecrc 1
Adds CRC information for each slice. This makes it possible for a decoder to detect errors in the bitstream, rather than blindly decoding a broken slice. (Read more here).
-slices 16
Each frame is split into 16 slices. 16 is a good trade-off between filesize and encoding time.
-c:a copy
copies all mapped audio streams.
-
output_file.mkv
path and name of the output file. Use the .mkv extension to save your file in a Matroska container. Optionally, choose a different extension if you want a different container, such as .mov or .avi.
+
output_file.mkv
path and name of the output file. Use the .mkv extension to save your file in a Matroska container. Optionally, choose a different extension if you want a different container, such as .mov or .avi.
-f framemd5
Decodes video with the framemd5 muxer in order to generate MD5 checksums for every frame of your input file. This allows you to verify losslessness when compared against the framemd5s of the output file.
-an
ignores the audio stream when creating framemd5 (audio no)
-
framemd5_output_file
path, name and extension of the framemd5 file.
+
framemd5_output_file
path, name and extension of the framemd5 file.
@@ -355,27 +355,27 @@

Convert DVD to H.264

-

ffmpeg -i concat:input_file_1\|input_file_2\|input_file_3 -c:v libx264 -c:a aac output_file.mp4

+

ffmpeg -i concat:input_file_1\|input_file_2\|input_file_3 -c:v libx264 -c:a aac output_file.mp4

This command allows you to create an H.264 file from a DVD source that is not copy-protected.

Before encoding, you’ll need to establish which of the .VOB files on the DVD or .iso contain the content that you wish to encode. Inside the VIDEO_TS directory, you will see a series of files with names like VTS_01_0.VOB, VTS_01_1.VOB, etc. Some of the .VOB files will contain menus, special features, etc, so locate the ones that contain target content by playing them back in VLC.

ffmpeg
starts the command
-
-i concat:input files
lists the input VOB files and directs FFmpeg to concatenate them. Each input file should be separated by a backslash and a pipe, like so:
+
-i concat:input files
lists the input VOB files and directs FFmpeg to concatenate them. Each input file should be separated by a backslash and a pipe, like so:
-i concat:VTS_01_1.VOB\|VTS_01_2.VOB\|VTS_01_3.VOB
- The backslash is simply an escape character for the pipe (|).
+ The backslash is simply an escape character for the pipe (|).
-c:v libx264
sets the video codec as H.264
-c:a aac
encode audio as AAC.
AAC is the codec most often used for audio streams within an .mp4 container.
-
output_file.mp4
path and name of the output file
+
output_file.mp4
path and name of the output file
-

It’s also possible to adjust the quality of your output by setting the -crf and -preset values:

-

ffmpeg -i concat:input_file_1\|input_file_2\|input_file_3 -c:v libx264 -crf 18 -preset veryslow -c:a aac output_file.mp4

+

It’s also possible to adjust the quality of your output by setting the -crf and -preset values:

+

ffmpeg -i concat:input_file_1\|input_file_2\|input_file_3 -c:v libx264 -crf 18 -preset veryslow -c:a aac output_file.mp4

-crf 18
sets the constant rate factor to a visually lossless value. Libx264 defaults to a crf of 23, considered medium quality; a smaller CRF value produces a larger and higher quality video.
-
-preset veryslow
A slower preset will result in better compression and therefore a higher-quality file. The default is medium; slower presets are slow, slower, and veryslow.
+
-preset veryslow
A slower preset will result in better compression and therefore a higher-quality file. The default is medium; slower presets are slow, slower, and veryslow.
-

Bear in mind that by default, libx264 will only encode a single video stream and a single audio stream, picking the ‘best’ of the options available. To preserve all video and audio streams, add -map parameters:

-

ffmpeg -i concat:input_file_1\|input_file_2 -map 0:v -map 0:a -c:v libx264 -c:a aac output_file.mp4

+

Bear in mind that by default, libx264 will only encode a single video stream and a single audio stream, picking the ‘best’ of the options available. To preserve all video and audio streams, add -map parameters:

+

ffmpeg -i concat:input_file_1\|input_file_2 -map 0:v -map 0:a -c:v libx264 -c:a aac output_file.mp4

-map 0:v
encodes all video streams
-map 0:a
encodes all audio streams
@@ -389,24 +389,24 @@

Transcode to H.265/HEVC

-

ffmpeg -i input_file -c:v libx265 -pix_fmt yuv420p -c:a copy output_file

+

ffmpeg -i input_file -c:v libx265 -pix_fmt yuv420p -c:a copy output_file

This command takes an input file and transcodes it to H.265/HEVC in an .mp4 wrapper, keeping the audio codec the same as in the original file.

-

Note: FFmpeg must be compiled with libx265, the library of the H.265 codec, for this script to work. (Add the flag --with-x265 if using the brew install ffmpeg method).

+

Note: FFmpeg must be compiled with libx265, the library of the H.265 codec, for this script to work. (Add the flag --with-x265 if using the brew install ffmpeg method).

ffmpeg
starts the command
-
-i input file
path, name and extension of the input file
+
-i input file
path, name and extension of the input file
-c:v libx265
tells FFmpeg to encode the video as H.265
-pix_fmt yuv420p
libx265 will use a chroma subsampling scheme that is the closest match to that of the input. This can result in Y′CBCR 4:2:0, 4:2:2, or 4:4:4 chroma subsampling. For widest accessibility, it’s a good idea to specify 4:2:0 chroma subsampling.
-c:a copy
tells FFmpeg not to change the audio codec
-
output file
path, name and extension of the output file
+
output file
path, name and extension of the output file

The libx265 encoding library defaults to a ‘medium’ preset for compression quality and a CRF of 28. CRF stands for ‘constant rate factor’ and determines the quality and file size of the resulting H.265 video. The CRF scale ranges from 0 (best quality [lossless]; largest file size) to 51 (worst quality; smallest file size).

A CRF of 28 for H.265 can be considered a medium setting, corresponding to a CRF of 23 in encoding H.264, but should result in about half the file size.

To create a higher quality file, you can add these presets:

-

ffmpeg -i input_file -c:v libx265 -pix_fmt yuv420p -preset veryslow -crf 18 -c:a copy output_file

+

ffmpeg -i input_file -c:v libx265 -pix_fmt yuv420p -preset veryslow -crf 18 -c:a copy output_file

-
-preset veryslow
This option tells FFmpeg to use the slowest preset possible for the best compression quality.
-
-crf 18
Specifying a lower CRF will make a larger file with better visual quality. 18 is often considered a ‘visually lossless’ compression.
+
-preset veryslow
This option tells FFmpeg to use the slowest preset possible for the best compression quality.
+
-crf 18
Specifying a lower CRF will make a larger file with better visual quality. 18 is often considered a ‘visually lossless’ compression.
@@ -417,15 +417,15 @@

Transcode to H.265/HEVC

-

ffmpeg -i input_file -acodec libvorbis -b:v 690k output_file

+

ffmpeg -i input_file -acodec libvorbis -b:v 690k output_file

This command takes an input file and transcodes it to Ogg/Theora in an .ogv wrapper with 690k video bitrate.

-

Note: FFmpeg must be installed with support for Ogg Theora. If you are using Homebrew, you can check with brew info ffmpeg and then update it with brew upgrade ffmpeg --with-theora --with-libvorbis if necessary.

+

Note: FFmpeg must be installed with support for Ogg Theora. If you are using Homebrew, you can check with brew info ffmpeg and then update it with brew upgrade ffmpeg --with-theora --with-libvorbis if necessary.

ffmpeg
starts the command
-
-i input file
path, name and extension of the input file
+
-i input file
path, name and extension of the input file
-acodec libvorbis
tells FFmpeg to encode the audio using libvorbis
-b:v 690k
specifies the 690k video bitrate
-
output file
path, name and extension of the output file (make sure to include the .ogv filename suffix)
+
output file
path, name and extension of the output file (make sure to include the .ogv filename suffix)

This recipe is based on Paul Rouget's recipes.

@@ -441,17 +441,17 @@

WAV to MP3

-

ffmpeg -i input_file.wav -write_id3v1 1 -id3v2_version 3 -dither_method rectangular -out_sample_rate 48k -qscale:a 1 output_file.mp3

+

ffmpeg -i input_file.wav -write_id3v1 1 -id3v2_version 3 -dither_method rectangular -out_sample_rate 48k -qscale:a 1 output_file.mp3

This will convert your WAV files to MP3s.

ffmpeg
starts the command
-
-i input_file
path and name of the input file
+
-i input_file
path and name of the input file
-write_id3v1 1
This will write metadata to an ID3v1 tag at the head of the file, assuming you’ve embedded metadata into the WAV file.
-id3v2_version 3
This will write metadata to an ID3v2.3 tag at the tail of the file, assuming you’ve embedded metadata into the WAV file.
-dither_method rectangular
Dither makes sure you don’t unnecessarily truncate the dynamic range of your audio.
-out_sample_rate 48k
Sets the audio sampling frequency to 48 kHz. This can be omitted to use the same sampling frequency as the input.
-qscale:a 1
This sets the encoder to use a constant quality with a variable bitrate of between 190-250kbit/s. If you would prefer to use a constant bitrate, this could be replaced with -b:a 320k to set to the maximum bitrate allowed by the MP3 format. For more detailed discussion on variable vs constant bitrates see here.
-
output_file
path and name of the output file
+
output_file
path and name of the output file

A couple notes

    @@ -467,22 +467,22 @@

    Generate two access MP3s from input. One with appended audio (such as a copyright notice) and one unmodified.

    -

    ffmpeg -i input_file -i input_file_to_append -filter_complex "[0:a:0]asplit=2[a][b];[b]afifo[bb];[1:a:0][bb]concat=n=2:v=0:a=1[concatout]" -map "[a]" -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2 output_file.mp3 -map "[concatout]" -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2 output_file_appended.mp3

    +

    ffmpeg -i input_file -i input_file_to_append -filter_complex "[0:a:0]asplit=2[a][b];[b]afifo[bb];[1:a:0][bb]concat=n=2:v=0:a=1[concatout]" -map "[a]" -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2 output_file.mp3 -map "[concatout]" -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2 output_file_appended.mp3

    This script allows you to generate two derivative audio files from a master while appending audio from a separate file (for example a copyright or institutional notice) to one of them.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file (the master file)
    -
    -i input_file_to_append
    path, name and extension of the input file (the file to be appended to access file)
    +
    -i input_file
    path, name and extension of the input file (the master file)
    +
    -i input_file_to_append
    path, name and extension of the input file (the file to be appended to access file)
    -filter_complex
    enables the complex filtering to manage splitting the input to two audio streams
    [0:a:0]asplit=2[a][b];
    asplit allows audio streams to be split up for separate manipulation. This command splits the audio from the first input (the master file) into two streams "a" and "b"
    [b]afifo[bb];
    this buffers the stream "b" to help prevent dropped samples and renames stream to "bb"
    [1:a:0][bb]concat=n=2:v=0:a=1[concatout]
    concat is used to join files. n=2 tells the filter there are two inputs. v=0:a=1 Tells the filter there are 0 video outputs and 1 audio output. This command appends the audio from the second input to the beginning of stream "bb" and names the output "concatout"
    -map "[a]"
    this maps the unmodified audio stream to the first output
    -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2
    sets up MP3 options (using constant quality)
    -
    output_file
    path, name and extension of the output file (unmodified)
    +
    output_file
    path, name and extension of the output file (unmodified)
    -map "[concatout]"
    this maps the modified stream to the second output
    -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2
    sets up MP3 options (using constant quality)
    -
    output_file_appended
    path, name and extension of the output file (with appended notice)
    +
    output_file_appended
    path, name and extension of the output file (with appended notice)
    @@ -493,16 +493,16 @@

    WAV to AAC/MP4

    -

    ffmpeg -i input_file.wav -c:a aac -b:a 128k -dither_method rectangular -ar 44100 output_file.mp4

    +

    ffmpeg -i input_file.wav -c:a aac -b:a 128k -dither_method rectangular -ar 44100 output_file.mp4

    This will convert your WAV file to AAC/MP4.

    ffmpeg
    starts the command
    -
    -i input_file
    path and name of the input file
    +
    -i input_file
    path and name of the input file
    -c:a aac
    sets the audio codec to AAC
    -b:a 128k
    sets the bitrate of the audio to 128k
    -dither_method rectangular
    Dither makes sure you don’t unnecessarily truncate the dynamic range of your audio.
    -ar 44100
    sets the audio sampling frequency to 44100 Hz, or 44.1 kHz, or “CD quality”
    -
    output_file
    path and name of the output file
    +
    output_file
    path and name of the output file

    A note about dither methods. FFmpeg comes with a variety of dither algorithms, outlined in the official docs, though some may lead to unintended, not-subtle digital clipping on some systems.

    @@ -519,14 +519,14 @@

    Transform 4:3 aspect ratio into 16:9 with pillarbox

    Transform a video file with 4:3 aspect ratio into a video file with 16:9 aspect ratio by correct pillarboxing.

    -

    ffmpeg -i input_file -filter:v "pad=ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" -c:a copy output_file

    +

    ffmpeg -i input_file -filter:v "pad=ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" -c:a copy output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -filter:v "pad=ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
    video padding
    This resolution independent formula is actually padding any aspect ratio into 16:9 by pillarboxing, because the video filter uses relative values for input width (iw), input height (ih), output width (ow) and output height (oh).
    -c:a copy
    re-encodes using the same audio codec
    For silent videos you can replace -c:a copy by -an.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -538,15 +538,15 @@

    Transform 16:9 aspect ratio video into 4:3 with letterbox

    Transform a video file with 16:9 aspect ratio into a video file with 4:3 aspect ratio by correct letterboxing.

    -

    ffmpeg -i input_file -filter:v "pad=iw:iw*3/4:(ow-iw)/2:(oh-ih)/2" -c:a copy output_file

    +

    ffmpeg -i input_file -filter:v "pad=iw:iw*3/4:(ow-iw)/2:(oh-ih)/2" -c:a copy output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -filter:v "pad=iw:iw*3/4:(ow-iw)/2:(oh-ih)/2"
    video padding
    This resolution independent formula is actually padding any aspect ratio into 4:3 by letterboxing, because the video filter uses relative values for input width (iw), input height (ih), output width (ow) and output height (oh).
    -c:a copy
    re-encodes using the same audio codec
    For silent videos you can replace -c:a copy by -an.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -557,14 +557,14 @@

    Flip the video image horizontally and/or vertically

    -

    ffmpeg -i input_file -filter:v "hflip,vflip" -c:a copy output_file

    +

    ffmpeg -i input_file -filter:v "hflip,vflip" -c:a copy output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -filter:v "hflip,vflip"
    flips the image horizontally and vertically
    By using only one of the parameters hflip or vflip for filtering the image is flipped on that axis only. The quote marks are not mandatory.
    -c:a copy
    re-encodes using the same audio codec
    For silent videos you can replace -c:a copy by -an.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -576,10 +576,10 @@

    Transform SD into HD with pillarbox

    Transform a SD video file with 4:3 aspect ratio into an HD video file with 16:9 aspect ratio by correct pillarboxing.

    -

    ffmpeg -i input_file -filter:v "colormatrix=bt601:bt709, scale=1440:1080:flags=lanczos, pad=1920:1080:240:0" -c:a copy output_file

    +

    ffmpeg -i input_file -filter:v "colormatrix=bt601:bt709, scale=1440:1080:flags=lanczos, pad=1920:1080:240:0" -c:a copy output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -filter:v "colormatrix=bt601:bt709, scale=1440:1080:flags=lanczos, pad=1920:1080:240:0"
    set colour matrix, video scaling and padding
    Three filters are applied:
    1. The luma coefficients are modified from SD video (according to Rec. 601) to HD video (according to Rec. 709) by a colour matrix. Note that today Rec. 709 is often used also for SD and therefore you may cancel this parameter.
    2. @@ -588,7 +588,7 @@
    -c:a copy
    re-encodes using the same audio codec
    For silent videos you can replace -c:a copy with -an.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -599,13 +599,13 @@

    Change Display Aspect Ratio without reencoding video

    -

    ffmpeg -i input_file -c:v copy -aspect 4:3 output_file

    +

    ffmpeg -i input_file -c:v copy -aspect 4:3 output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -c:v copy
    Copy all mapped video streams.
    -aspect 4:3
    Change Display Aspect Ratio to 4:3. Experiment with other aspect ratios such as 16:9. If used together with -c:v copy, it will affect the aspect ratio stored at container level, but not the aspect ratio stored in encoded frames, if it exists.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -617,45 +617,45 @@

    Transcode video to a different colourspace

    This command uses a filter to convert the video to a different colour space.

    -

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=src:dst output_file

    +

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=src:dst output_file

    ffmpeg
    starts the command
    -
    -i input file
    path, name and extension of the input file
    +
    -i input file
    path, name and extension of the input file
    -c:v libx264
    tells FFmpeg to encode the video stream as H.264
    -
    -vf colormatrix=src:dst
    the video filter colormatrix will be applied, with the given source and destination colourspaces.
    +
    -vf colormatrix=src:dst
    the video filter colormatrix will be applied, with the given source and destination colourspaces.
    Accepted values include bt601 (Rec.601), smpte170m (Rec.601, 525-line/NTSC version), bt470bg (Rec.601, 625-line/PAL version), bt709 (Rec.709), and bt2020 (Rec.2020).
    For example, to convert from Rec.601 to Rec.709, you would use -vf colormatrix=bt601:bt709.
    -
    output file
    path, name and extension of the output file
    +
    output file
    path, name and extension of the output file
    -

    Note: Converting between colourspaces with FFmpeg can be done via either the colormatrix or colorspace filters, with colorspace allowing finer control (individual setting of colourspace, transfer characteristics, primaries, range, pixel format, etc). See this entry on the FFmpeg wiki, and the FFmpeg documentation for colormatrix and colorspace.

    +

    Note: Converting between colourspaces with FFmpeg can be done via either the colormatrix or colorspace filters, with colorspace allowing finer control (individual setting of colourspace, transfer characteristics, primaries, range, pixel format, etc). See this entry on the FFmpeg wiki, and the FFmpeg documentation for colormatrix and colorspace.


    Convert colourspace and embed colourspace metadata

    -

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=src:dst -color_primaries val -color_trc val -colorspace val output_file

    +

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=src:dst -color_primaries val -color_trc val -colorspace val output_file

    ffmpeg
    starts the command
    -
    -i input file
    path, name and extension of the input file
    +
    -i input file
    path, name and extension of the input file
    -c:v libx264
    encode video as H.264
    -
    -vf colormatrix=src:dst
    the video filter colormatrix will be applied, with the given source and destination colourspaces.
    -
    -color_primaries val
    tags video with the given colour primaries.
    +
    -vf colormatrix=src:dst
    the video filter colormatrix will be applied, with the given source and destination colourspaces.
    +
    -color_primaries val
    tags video with the given colour primaries.
    Accepted values include smpte170m (Rec.601, 525-line/NTSC version), bt470bg (Rec.601, 625-line/PAL version), bt709 (Rec.709), and bt2020 (Rec.2020). -
    -color_trc val
    tags video with the given transfer characteristics (gamma).
    +
    -color_trc val
    tags video with the given transfer characteristics (gamma).
    Accepted values include smpte170m (Rec.601, 525-line/NTSC version), gamma28 (Rec.601, 625-line/PAL version)1, bt709 (Rec.709), bt2020_10 (Rec.2020 10-bit), and bt2020_12 (Rec.2020 12-bit).
    -
    -colorspace val
    tags video as being in the given colourspace.
    +
    -colorspace val
    tags video as being in the given colourspace.
    Accepted values include smpte170m (Rec.601, 525-line/NTSC version), bt470bg (Rec.601, 625-line/PAL version), bt709 (Rec.709), bt2020_cl (Rec.2020 constant luminance), and bt2020_ncl (Rec.2020 non-constant luminance).
    -
    output file
    path, name and extension of the output file
    +
    output file
    path, name and extension of the output file

    Examples

    To Rec.601 (525-line/NTSC):

    -

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=bt709:smpte170m -color_primaries smpte170m -color_trc smpte170m -colorspace smpte170m output_file

    +

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=bt709:smpte170m -color_primaries smpte170m -color_trc smpte170m -colorspace smpte170m output_file

    To Rec.601 (625-line/PAL):

    -

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=bt709:bt470bg -color_primaries bt470bg -color_trc gamma28 -colorspace bt470bg output_file

    +

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=bt709:bt470bg -color_primaries bt470bg -color_trc gamma28 -colorspace bt470bg output_file

    To Rec.709:

    -

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=bt601:bt709 -color_primaries bt709 -color_trc bt709 -colorspace bt709 output_file

    +

    ffmpeg -i input_file -c:v libx264 -vf colormatrix=bt601:bt709 -color_primaries bt709 -color_trc bt709 -colorspace bt709 output_file

    MediaInfo output examples:

    MediaInfo screenshots of colourspace metadata

    Using this command it is possible to add Rec.709 tags to a file that is actually Rec.601 (etc), so apply with caution!

    These commands are relevant for H.264 and H.265 videos, encoded with libx264 and libx265 respectively.

    -

    Note: If you wish to embed colourspace metadata without changing to another colourspace, omit -vf colormatrix=src:dst. However, since it is libx264/libx265 that writes the metadata, it’s not possible to add these tags without reencoding the video stream.

    +

    Note: If you wish to embed colourspace metadata without changing to another colourspace, omit -vf colormatrix=src:dst. However, since it is libx264/libx265 that writes the metadata, it’s not possible to add these tags without reencoding the video stream.

    For all possible values for -color_primaries, -color_trc, and -colorspace, see the FFmpeg documentation on codec options.


    1. Out of step with the regular pattern, -color_trc doesn’t accept bt470bg; it is instead here referred to directly as gamma.
    @@ -670,11 +670,11 @@

    Modify image and sound speed

    E.g. for converting 24fps to 25fps with audio pitch compensation for PAL access copies. (Thanks @kieranjol!)

    -

    ffmpeg -i input_file -filter_complex "[0:v]setpts=input_fps/output_fps*PTS[v]; [0:a]atempo=output_fps/input_fps[a]" -map "[v]" -map "[a]" output_file

    +

    ffmpeg -i input_file -filter_complex "[0:v]setpts=input_fps/output_fps*PTS[v]; [0:a]atempo=output_fps/input_fps[a]" -map "[v]" -map "[a]" output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    -
    -filter_complex "[0:v]setpts=input_fps/output_fps*PTS[v]; [0:a]atempo=output_fps/input_fps[a]"
    A complex filter is needed here, in order to handle video stream and the audio stream separately. The setpts video filter modifies the PTS (presentation time stamp) of the video stream, and the atempo audio filter modifies the speed of the audio stream while keeping the same sound pitch. Note that the parameter order for the image and for the sound are inverted: +
    -i input_file
    path, name and extension of the input file
    +
    -filter_complex "[0:v]setpts=input_fps/output_fps*PTS[v]; [0:a]atempo=output_fps/input_fps[a]"
    A complex filter is needed here, in order to handle video stream and the audio stream separately. The setpts video filter modifies the PTS (presentation time stamp) of the video stream, and the atempo audio filter modifies the speed of the audio stream while keeping the same sound pitch. Note that the parameter order for the image and for the sound are inverted:
    • In the video filter setpts the numerator input_fps sets the input speed and the denominator output_fps sets the output speed; both values are given in frames per second.
    • In the sound filter atempo the numerator output_fps sets the output speed and the denominator input_fps sets the input speed; both values are given in frames per second.
    • @@ -682,7 +682,7 @@ The different filters in a complex filter can be divided either by comma or semicolon. The quotation marks allow to insert a space between the filters for readability.
    -map "[v]"
    maps the video stream and:
    -map "[a]"
    maps the audio stream together into:
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -695,30 +695,30 @@

    Set stream properties

    Find undetermined or unknown stream properties

    These examples use QuickTime inputs and outputs. The strategy will vary or may not be possible in other file formats. In the case of these examples it is the intention to make a lossless copy while clarifying an unknown characteristic of the stream.

    -

    ffprobe input_file -show_streams

    +

    ffprobe input_file -show_streams

    ffprobe
    starts the command
    -
    input_file
    path, name and extension of the input file
    +
    input_file
    path, name and extension of the input file
    -show_streams
    Shows metadata of stream properties

    Values that are set to 'unknown' and 'undetermined' may be unspecified within the stream. An unknown aspect ratio would be expressed as '0:1'. Streams with many unknown properties may have interoperability issues or not play as intended. In many cases, an unknown or undetermined value may be accurate because the information about the source is unclear, but often the value is intended to be known. In many cases the stream will played with an assumed value if undetermined (for instance a display_aspect_ratio of '0:1' may be played as 'WIDTH:HEIGHT'), but this may or may not be what is intended. Use carefully.

    Set aspect ratio

    -

    If the display_aspect_ratio is set to '0:1' it may be clarified with the -aspect option and stream copy.

    -

    ffmpeg -i input_file -c copy -map 0 -aspect DAR_NUM:DAR_DEN output_file

    +

    If the display_aspect_ratio is set to '0:1' it may be clarified with the -aspect option and stream copy.

    +

    ffmpeg -i input_file -c copy -map 0 -aspect DAR_NUM:DAR_DEN output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -c copy
    Usings stream copy for all streams
    -map 0
    tells FFmpeg to map all streams of the input to the output.
    -
    -aspect DAR_NUM:DAR_DEN
    Replace DAR_NUM with the display aspect ratio numerator and DAR_DEN with the display aspect ratio denominator, such as -aspect 4:3 or -aspect 16:9.
    -
    output_file
    path, name and extension of the output file
    +
    -aspect DAR_NUM:DAR_DEN
    Replace DAR_NUM with the display aspect ratio numerator and DAR_DEN with the display aspect ratio denominator, such as -aspect 4:3 or -aspect 16:9.
    +
    output_file
    path, name and extension of the output file

    Adding other stream properties.

    -

    Other properties may be clarified in a similar way. Replace -aspect and its value with other properties such as shown in the options below. Note that setting color values in QuickTime requires that -movflags write_colr is set.

    +

    Other properties may be clarified in a similar way. Replace -aspect and its value with other properties such as shown in the options below. Note that setting color values in QuickTime requires that -movflags write_colr is set.

    -
    -color_primaries VALUE -movflags write_colr
    Set a new color_primaries value.
    -
    -color_trc VALUE -movflags write_colr
    Set a new color_transfer value.
    -
    -field_order VALUE
    Set interlacement values.
    +
    -color_primaries VALUE -movflags write_colr
    Set a new color_primaries value.
    +
    -color_trc VALUE -movflags write_colr
    Set a new color_transfer value.
    +
    -field_order VALUE
    Set interlacement values.

    The possible values for -color_primaries, -color_trc, and -field_order are given in the Codec Options section of the FFmpeg docs - scroll down to near the bottom of the section.

    @@ -730,26 +730,26 @@

    Crop video

    -

    ffmpeg -i input_file -vf "crop=width:height" output_file

    +

    ffmpeg -i input_file -vf "crop=width:height" output_file

    This command crops the input video to the dimensions defined

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    -
    -vf "width:height"
    Crops the video to the given width and height (in pixels).
    - By default, the crop area is centred: that is, the position of the top left of the cropped area is set to x = (input_width - output_width) / 2, y = input_height - output_height) / 2. +
    -i input_file
    path, name and extension of the input file
    +
    -vf "width:height"
    Crops the video to the given width and height (in pixels).
    + By default, the crop area is centred: that is, the position of the top left of the cropped area is set to x = (input_width - output_width) / 2, y = input_height - output_height) / 2.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file

    It's also possible to specify the crop position by adding the x and y coordinates representing the top left of your cropped area to your crop filter, as such:

    -

    ffmpeg -i input_file -vf "crop=width:height[:x_position:y_position]" output_file

    +

    ffmpeg -i input_file -vf "crop=width:height[:x_position:y_position]" output_file

    Examples

    The original frame, a screenshot of the SMPTE colourbars:

    VLC screenshot of Maggie Cheung -

    Result of the command ffmpeg -i smpte_coloursbars.mov -vf "crop=500:500" output_file:

    +

    Result of the command ffmpeg -i smpte_coloursbars.mov -vf "crop=500:500" output_file:

    VLC screenshot of Maggie Cheung, cropped from original -

    Result of the command ffmpeg -i smpte_coloursbars.mov -vf "crop=500:500:0:0" output_file, appending :0:0 to crop from the top left corner:

    +

    Result of the command ffmpeg -i smpte_coloursbars.mov -vf "crop=500:500:0:0" output_file, appending :0:0 to crop from the top left corner:

    VLC screenshot of Maggie Cheung, cropped from original -

    Result of the command ffmpeg -i smpte_coloursbars.mov -vf "crop=500:300:500:30" output_file:

    +

    Result of the command ffmpeg -i smpte_coloursbars.mov -vf "crop=500:300:500:30" output_file:

    VLC screenshot of Maggie Cheung, cropped from original
    @@ -764,14 +764,14 @@

    Extract audio from an AV file

    -

    ffmpeg -i input_file -c:a copy -vn output_file

    +

    ffmpeg -i input_file -c:a copy -vn output_file

    This command extracts the audio stream without loss from an audiovisual file.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -c:a copy
    re-encodes using the same audio codec
    -vn
    no video stream
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -782,11 +782,11 @@

    Combine audio tracks into one in a video file

    -

    ffmpeg -i input_file -filter_complex "[0:a:0][0:a:1]amerge[out]" -map 0:v -map "[out]" -c:v copy -shortest output_file

    +

    ffmpeg -i input_file -filter_complex "[0:a:0][0:a:1]amerge[out]" -map 0:v -map "[out]" -c:v copy -shortest output_file

    This command combines two audio tracks present in a video file into one stream. It can be useful in situations where a downstream process, like YouTube’s automatic captioning, expect one audio track. To ensure that you’re mapping the right audio tracks run ffprobe before writing the script to identify which tracks are desired. More than two audio streams can be combined by extending the pattern present in the -filter_complex option.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -filter_complex
    tells ffmpeg that we will be using a complex filter
    "
    quotation mark to start filtergraph
    [0:a:0][0:a:1]amerge[out]
    combines the two audio tracks into one
    @@ -795,7 +795,7 @@
    -map "[out]"
    map the combined audio defined by the filter
    -c:v copy
    copy the video
    -shortest
    limit to the shortest stream
    -
    output_file
    path, name and extension of the video output file
    +
    output_file
    path, name and extension of the video output file
    @@ -806,15 +806,15 @@

    Flip audio phase shift

    -

    ffmpeg -i input_file -af pan="stereo|c0=c0|c1=-1*c1" output_file

    +

    ffmpeg -i input_file -af pan="stereo|c0=c0|c1=-1*c1" output_file

    This command inverses the audio phase of the second channel by rotating it 180°.

    ffmpeg
    starts the command
    -
    -i input file
    path, name and extension of the input file
    +
    -i input file
    path, name and extension of the input file
    -af
    specifies that the next section should be interpreted as an audio filter
    pan=
    tell the quoted text below to use the pan filter
    "stereo|c0=c0|c1=-1*c1"
    maps the output's first channel (c0) to the input's first channel and the output's second channel (c1) to the inverse of the input's second channel
    -
    output file
    path, name and extension of the output file
    +
    output file
    path, name and extension of the output file
    @@ -825,16 +825,16 @@

    Calculate Loudness Levels

    -

    ffmpeg -i input_file -af loudnorm=print_format=json -f null -

    +

    ffmpeg -i input_file -af loudnorm=print_format=json -f null -

    This filter calculates and outputs loudness information in json about an input file (labeled input) as well as what the levels would be if loudnorm were applied in its one pass mode (labeled output). The values generated can be used as inputs for a 'second pass' of the loudnorm filter allowing more accurate loudness normalization than if it is used in a single pass.

    These instructions use the loudnorm defaults, which align well with PBS recommendations for target loudness. More information can be found at the loudnorm documentation.

    Information about PBS loudness standards can be found in the PBS Technical Operating Specifications document. Information about EBU loudness standards can be found in the EBU R 128 recommendation document.

    ffmpeg
    starts the command
    -
    input_file
    path, name and extension of the input file
    +
    input_file
    path, name and extension of the input file
    -af loudnorm
    activates the loudnorm filter
    print_format=json
    sets the output format for loudness information to json. This format makes it easy to use in a second pass. For a more human readable output, this can be set to print_format=summary
    -
    -f null -
    sets the file output to null (since we are only interested in the metadata generated)
    +
    -f null -
    sets the file output to null (since we are only interested in the metadata generated)
    @@ -845,13 +845,13 @@

    RIAA Equalization

    -

    ffmpeg -i input_file -af aemphasis=type=riaa output_file

    +

    ffmpeg -i input_file -af aemphasis=type=riaa output_file

    This will apply RIAA equalization to an input file allowing correct listening of audio transferred 'flat' (without EQ) from records that used this EQ curve. For more information about RIAA equalization see the Wikipedia page on the subject.

    ffmpeg
    starts the command
    -
    input_file
    path, name and extension of the input file
    +
    input_file
    path, name and extension of the input file
    -af aemphasis=type=riaa
    activates the aemphasis filter and sets it to use RIAA equalization
    -
    output_file
    path and name of output file
    +
    output_file
    path and name of output file
    @@ -862,16 +862,16 @@

    One Pass Loudness Normalization

    -

    ffmpeg -i input_file -af loudnorm=dual_mono=true -ar 48k output_file

    +

    ffmpeg -i input_file -af loudnorm=dual_mono=true -ar 48k output_file

    This will normalize the loudness of an input using one pass, which is quicker but less accurate than using two passes. This command uses the loudnorm filter defaults for target loudness. These defaults align well with PBS recommendations, but loudnorm does allow targeting of specific loudness levels. More information can be found at the loudnorm documentation.

    Information about PBS loudness standards can be found in the PBS Technical Operating Specifications document. Information about EBU loudness standards can be found in the EBU R 128 recommendation document.

    ffmpeg
    starts the command
    -
    input_file
    path, name and extension of the input file
    +
    input_file
    path, name and extension of the input file
    -af loudnorm
    activates the loudnorm filter with default settings
    dual_mono=true
    (optional) Use this for mono files meant to be played back on stereo systems for correct loudness. Not necessary for multi-track inputs.
    -ar 48k
    Sets the output sample rate to 48 kHz. (The loudnorm filter upsamples to 192 kHz so it is best to manually set a desired output sample rate).
    -
    output_file
    path, name and extension for output file
    +
    output_file
    path, name and extension for output file
    @@ -882,22 +882,22 @@

    Two Pass Loudness Normalization

    -

    ffmpeg -i input_file -af loudnorm=dual_mono=true:measured_I=input_i:measured_TP=input_tp:measured_LRA=input_lra:measured_thresh=input_thresh:offset=target_offset:linear=true -ar 48k output_file

    +

    ffmpeg -i input_file -af loudnorm=dual_mono=true:measured_I=input_i:measured_TP=input_tp:measured_LRA=input_lra:measured_thresh=input_thresh:offset=target_offset:linear=true -ar 48k output_file

    This command allows using the levels calculated using a first pass of the loudnorm filter to more accurately normalize loudness. This command uses the loudnorm filter defaults for target loudness. These defaults align well with PBS recommendations, but loudnorm does allow targeting of specific loudness levels. More information can be found at the loudnorm documentation.

    Information about PBS loudness standards can be found in the PBS Technical Operating Specifications document. Information about EBU loudness standards can be found in the EBU R 128 recommendation document.

    ffmpeg
    starts the command
    -
    input_file
    path, name and extension of the input file
    +
    input_file
    path, name and extension of the input file
    -af loudnorm
    activates the loudnorm filter with default settings
    dual_mono=true
    (optional) use this for mono files meant to be played back on stereo systems for correct loudness. Not necessary for multi-track inputs.
    -
    measured_I=input_i
    use the 'input_i' value (integrated loudness) from the first pass in place of input_i
    -
    measured_TP=input_tp
    use the 'input_tp' value (true peak) from the first pass in place of input_tp
    -
    measured_LRA=input_lra
    use the 'input_lra' value (loudness range) from the first pass in place of input_lra
    -
    measured_LRA=input_thresh
    use the 'input_thresh' value (threshold) from the first pass in place of input_thresh
    -
    offset=target_offset
    use the 'target_offset' value (offset) from the first pass in place of target_offset
    +
    measured_I=input_i
    use the 'input_i' value (integrated loudness) from the first pass in place of input_i
    +
    measured_TP=input_tp
    use the 'input_tp' value (true peak) from the first pass in place of input_tp
    +
    measured_LRA=input_lra
    use the 'input_lra' value (loudness range) from the first pass in place of input_lra
    +
    measured_LRA=input_thresh
    use the 'input_thresh' value (threshold) from the first pass in place of input_thresh
    +
    offset=target_offset
    use the 'target_offset' value (offset) from the first pass in place of target_offset
    linear=true
    tells loudnorm to use linear normalization
    -ar 48k
    Sets the output sample rate to 48 kHz. (The loudnorm filter upsamples to 192 kHz so it is best to manually set a desired output sample rate).
    -
    output_file
    path, name and extension for output file
    +
    output_file
    path, name and extension for output file
    @@ -908,14 +908,14 @@

    Fix AV Sync: Resample audio

    -

    ffmpeg -i input_file -c:v copy -c:a pcm_s16le -af "aresample=async=1000" output_file

    +

    ffmpeg -i input_file -c:v copy -c:a pcm_s16le -af "aresample=async=1000" output_file

    ffmpeg
    starts the command
    -
    input_file
    path, name and extension of the input file
    +
    input_file
    path, name and extension of the input file
    -c:v copy
    Copy all mapped video streams.
    -c:a pcm_s16le
    tells FFmpeg to encode the audio stream in 16-bit linear PCM (little endian)
    -af "aresample=async=1000"
    Uses the aresample filter to stretch/squeeze samples to given timestamps, with a maximum of 1000 samples per second compensation.
    -
    output_file
    path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
    +
    output_file
    path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
    @@ -930,22 +930,22 @@

    Join files together

    -

    ffmpeg -f concat -i mylist.txt -c copy output_file

    +

    ffmpeg -f concat -i mylist.txt -c copy output_file

    This command takes two or more files of the same file type and joins them together to make a single file. All that the program needs is a text file with a list specifying the files that should be joined. However, it only works properly if the files to be combined have the exact same codec and technical specifications. Be careful, FFmpeg may appear to have successfully joined two video files with different codecs, but may only bring over the audio from the second file or have other weird behaviors. Don’t use this command for joining files with different codecs and technical specs and always preview your resulting video file!

    ffmpeg
    starts the command
    -f concat
    forces ffmpeg to concatenate the files and to keep the same file format
    -
    -i mylist.txt
    path, name and extension of the input file. Per the FFmpeg documentation, it is preferable to specify relative rather than absolute file paths, as allowing absolute file paths may pose a security risk.
    +
    -i mylist.txt
    path, name and extension of the input file. Per the FFmpeg documentation, it is preferable to specify relative rather than absolute file paths, as allowing absolute file paths may pose a security risk.
    This text file contains the list of files to be concatenated and should be formatted as follows: -
    file './first_file.ext'
    -  file './second_file.ext'
    +        
    file './first_file.ext'
    +  file './second_file.ext'
       . . .
    -  file './last_file.ext'
    - In the above, file is simply the word "file". Straight apostrophes ('like this') rather than curved quotation marks (‘like this’) must be used to enclose the file paths.
    - Note: If specifying absolute file paths in the .txt file, add -safe 0 before the input file.
    - e.g.: ffmpeg -f concat -safe 0 -i mylist.txt -c copy output_file
    + file './last_file.ext' + In the above, file is simply the word "file". Straight apostrophes ('like this') rather than curved quotation marks (‘like this’) must be used to enclose the file paths.
    + Note: If specifying absolute file paths in the .txt file, add -safe 0 before the input file.
    + e.g.: ffmpeg -f concat -safe 0 -i mylist.txt -c copy output_file
    -c copy
    use stream copy mode to re-mux instead of re-encode
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file

    For more information, see the FFmpeg wiki page on concatenating files.

    @@ -957,18 +957,18 @@

    Join files together

    -

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0][0:a:0][1:v:0][1:a:0]concat=n=2:v=1:a=1[video_out][audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    +

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0][0:a:0][1:v:0][1:a:0]concat=n=2:v=1:a=1[video_out][audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    This command takes two or more files of the different file types and joins them together to make a single file.

    The input files may differ in many respects - container, codec, chroma subsampling scheme, framerate, etc. However, the above command only works properly if the files to be combined have the same dimensions (e.g., 720x576). Also note that if the input files have different framerates, then the output file will be of variable framerate.

    Some aspects of the input files will be normalised: for example, if an input file contains a video track and an audio track that do not have exactly the same duration, the shorter one will be padded. In the case of a shorter video track, the last frame will be repeated in order to cover the missing video; in the case of a shorter audio track, the audio stream will be padded with silence.

    ffmpeg
    starts the command
    -
    -i input_1.ext
    path, name and extension of the first input file
    -
    -i input_2.ext
    path, name and extension of the second input file
    +
    -i input_1.ext
    path, name and extension of the first input file
    +
    -i input_2.ext
    path, name and extension of the second input file
    -filter_complex
    states that a complex filtergraph will be used
    "
    quotation mark to start filtergraph
    [0:v:0][0:a:0]
    selects the first video stream and first audio stream from the first input.
    - Each reference to a specific stream is enclosed in square brackets. In the first stream reference, 0:v:0, the first zero refers to the first input file, v means video stream, and the second zero indicates that it is the first video stream in the file that should be selected. Likewise, 0:a:0 means the first audio stream in the first input file.
    + Each reference to a specific stream is enclosed in square brackets. In the first stream reference, 0:v:0, the first zero refers to the first input file, v means video stream, and the second zero indicates that it is the first video stream in the file that should be selected. Likewise, 0:a:0 means the first audio stream in the first input file.
    As demonstrated above, ffmpeg uses zero-indexing: 0 means the first input/stream/etc, 1 means the second input/stream/etc, and 4 would mean the fifth input/stream/etc.
    [1:v:0][1:a:0]
    As described above, this means select the first video and audio streams from the second input file.
    concat=
    starts the concat filter
    @@ -984,7 +984,7 @@
    "
    quotation mark to end filtergraph
    -map "[video_out]"
    map the concatenated video stream into the output file by referencing the variable defined above
    -map "[audio_out]"
    map the concatenated audio stream into the output file by referencing the variable defined above
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file

    If no characteristics of the output files are specified, ffmpeg will use the default encodings associated with the given output file type. To specify the characteristics of the output stream(s), add flags after each -map "[out]" part of the command.

    For example, to ensure that the video stream of the output file is visually lossless H.264 with a 4:2:0 chroma subsampling scheme, the command above could be amended to include the following:
    @@ -996,15 +996,15 @@

    -vf scale=1920:1080:flags=lanczos

    (The Lanczos scaling algorithm is recommended, as it is slower but better than the default bilinear algorithm).

    The rescaling should be applied just before the point where the streams to be used in the output file are listed. Select the stream you want to rescale, apply the filter, and assign that to a variable name (rescaled_video in the below example). Then you use this variable name in the list of streams to be concatenated.

    -

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0] scale=1920:1080:flags=lanczos [rescaled_video], [rescaled_video] [0:a:0] [1:v:0] [1:a:0] concat=n=2:v=1:a=1 [video_out] [audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    +

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0] scale=1920:1080:flags=lanczos [rescaled_video], [rescaled_video] [0:a:0] [1:v:0] [1:a:0] concat=n=2:v=1:a=1 [video_out] [audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    However, this will only have the desired visual output if the inputs have the same aspect ratio. If you wish to concatenate an SD and an HD file, you will also wish to pillarbox the SD file while upscaling. (See the Convert 4:3 to pillarboxed HD command). The full command would look like this:

    -

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0] scale=1440:1080:flags=lanczos, pad=1920:1080:(ow-iw)/2:(oh-ih)/2 [to_hd_video], [to_hd_video] [0:a:0] [1:v:0] [1:a:0] concat=n=2:v=1:a=1 [video_out] [audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    +

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0] scale=1440:1080:flags=lanczos, pad=1920:1080:(ow-iw)/2:(oh-ih)/2 [to_hd_video], [to_hd_video] [0:a:0] [1:v:0] [1:a:0] concat=n=2:v=1:a=1 [video_out] [audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    Here, the first input is an SD file which needs to be upscaled to match the second input, which is 1920x1080. The scale filter enlarges the SD input to the height of the HD frame, keeping the 4:3 aspect ratio; then, the video is pillarboxed within a 1920x1080 frame.

    Variation: concatenating files of different framerates

    If the input files have different framerates, then the output file may be of variable framerate. To explicitly obtain an output file of constant framerate, you may wish convert an input (or multiple inputs) to a different framerate prior to concatenation.

    You can speed up or slow down a file using the fps and atempo filters (see also the Modify speed command).

    Here's an example of the full command, in which input_1 is 30fps, input_2 is 25fps, and 25fps is the desired output speed.

    -

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0] fps=fps=25 [video_to_25fps]; [0:a:0] atempo=(25/30) [audio_to_25fps]; [video_to_25fps] [audio_to_25fps] [1:v:0] [1:a:0] concat=n=2:v=1:a=1 [video_out] [audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    +

    ffmpeg -i input_1.avi -i input_2.mp4 -filter_complex "[0:v:0] fps=fps=25 [video_to_25fps]; [0:a:0] atempo=(25/30) [audio_to_25fps]; [video_to_25fps] [audio_to_25fps] [1:v:0] [1:a:0] concat=n=2:v=1:a=1 [video_out] [audio_out]" -map "[video_out]" -map "[audio_out]" output_file

    Note that the fps filter will drop or repeat frames as necessary in order to achieve the desired frame rate - see the FFmpeg fps docs for more details.

    For more information, see the FFmpeg wiki page on concatenating files of different types.

    @@ -1016,16 +1016,16 @@

    Split file into segments

    -

    ffmpeg -i input_file -c copy -map 0 -f segment -segment_time 60 -reset_timestamps 1 output_file-%03d.mkv

    +

    ffmpeg -i input_file -c copy -map 0 -f segment -segment_time 60 -reset_timestamps 1 output_file-%03d.mkv

    ffmpeg
    Starts the command.
    -
    -i input_file
    Takes in a normal file.
    +
    -i input_file
    Takes in a normal file.
    -c copy
    Use stream copy mode to re-mux instead of re-encode.
    -map 0
    tells FFmpeg to map all streams of the input to the output.
    -f segment
    Use segment muxer for generating the output.
    -segment_time 60
    Set duration of each segment (in seconds). This example creates segments with max. duration of 60s each.
    -reset_timestamps 1
    Reset timestamps of each segment to 0. Meant to ease the playback of the generated segments.
    -
    output_file-%03d.mkv
    +
    output_file-%03d.mkv

    Path, name and extension of the output file.
    In order to have an incrementing number in each segment filename, FFmpeg supports printf-style syntax for a counter.

    @@ -1047,20 +1047,20 @@

    Trim a video without re-encoding

    -

    ffmpeg -i input_file -ss 00:02:00 -to 00:55:00 -c copy -map 0 output_file

    +

    ffmpeg -i input_file -ss 00:02:00 -to 00:55:00 -c copy -map 0 output_file

    This command allows you to create an excerpt from a video file without re-encoding the image data.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -ss 00:02:00
    sets in point at 00:02:00
    -to 00:55:00
    sets out point at 00:55:00
    -c copy
    use stream copy mode (no re-encoding)
    -map 0
    tells FFmpeg to map all streams of the input to the output.
    - Note: watch out when using -ss with -c copy if the source is encoded with an interframe codec (e.g., H.264). Since FFmpeg must split on i-frames, it will seek to the nearest i-frame to begin the stream copy.
    -
    output_file
    path, name and extension of the output file
    + Note: watch out when using -ss with -c copy if the source is encoded with an interframe codec (e.g., H.264). Since FFmpeg must split on i-frames, it will seek to the nearest i-frame to begin the stream copy.
    +
    output_file
    path, name and extension of the output file

    Variation: trim video by setting duration, by using -t instead of -to

    -

    ffmpeg -i input_file -ss 00:05:00 -t 10 -c copy output_file

    +

    ffmpeg -i input_file -ss 00:05:00 -t 10 -c copy output_file

    -ss 00:05:00 -t 10
    Beginning five minutes into the original video, this command will create a 10-second-long excerpt.
    @@ -1074,15 +1074,15 @@

    Excerpt from beginning

    -

    ffmpeg -i input_file -t 5 -c copy -map 0 output_file

    +

    ffmpeg -i input_file -t 5 -c copy -map 0 output_file

    This command captures a certain portion of a video file, starting from the beginning and continuing for the amount of time (in seconds) specified in the script. This can be used to create a preview file, or to remove unwanted content from the end of the file. To be more specific, use timecode, such as 00:00:05.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    -
    -t 5
    tells FFmpeg to stop copying from the input file after a certain time, and specifies the number of seconds after which to stop copying. In this case, 5 seconds is specified.
    +
    -i input_file
    path, name and extension of the input file
    +
    -t 5
    tells FFmpeg to stop copying from the input file after a certain time, and specifies the number of seconds after which to stop copying. In this case, 5 seconds is specified.
    -c copy
    use stream copy mode to re-mux instead of re-encode
    -map 0
    tells FFmpeg to map all streams of the input to the output.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -1093,15 +1093,15 @@

    Excerpt to end

    -

    ffmpeg -i input_file -ss 5 -c copy -map 0 output_file

    +

    ffmpeg -i input_file -ss 5 -c copy -map 0 output_file

    This command copies a video file starting from a specified time, removing the first few seconds from the output. This can be used to create an excerpt, or remove unwanted content from the beginning of a video file.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    -
    -ss 5
    tells FFmpeg what timecode in the file to look for to start copying, and specifies the number of seconds into the video that FFmpeg should start copying. To be more specific, you can use timecode such as 00:00:05.
    +
    -i input_file
    path, name and extension of the input file
    +
    -ss 5
    tells FFmpeg what timecode in the file to look for to start copying, and specifies the number of seconds into the video that FFmpeg should start copying. To be more specific, you can use timecode such as 00:00:05.
    -c copy
    use stream copy mode to re-mux instead of re-encode
    -map 0
    tells FFmpeg to map all streams of the input to the output.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -1112,15 +1112,15 @@

    Excerpt from end

    -

    ffmpeg -sseof -5 -i input_file -c copy -map 0 output_file

    +

    ffmpeg -sseof -5 -i input_file -c copy -map 0 output_file

    This command copies a video file starting from a specified time before the end of the file, removing everything before from the output. This can be used to create an excerpt, or extract content from the end of a video file (e.g. for extracting the closing credits).

    ffmpeg
    starts the command
    -
    -sseof -5
    This parameter must stay before the input file. It tells FFmpeg what timecode in the file to look for to start copying, and specifies the number of seconds from the end of the video that FFmpeg should start copying. The end of the file has index 0 and the minus sign is needed to reference earlier portions. To be more specific, you can use timecode such as -00:00:05. Note that in most file formats it is not possible to seek exactly, so FFmpeg will seek to the closest point before.
    -
    -i input_file
    path, name and extension of the input file
    +
    -sseof -5
    This parameter must stay before the input file. It tells FFmpeg what timecode in the file to look for to start copying, and specifies the number of seconds from the end of the video that FFmpeg should start copying. The end of the file has index 0 and the minus sign is needed to reference earlier portions. To be more specific, you can use timecode such as -00:00:05. Note that in most file formats it is not possible to seek exactly, so FFmpeg will seek to the closest point before.
    +
    -i input_file
    path, name and extension of the input file
    -c copy
    use stream copy mode to re-mux instead of re-encode
    -map 0
    tells FFmpeg to map all streams of the input to the output.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -1135,22 +1135,22 @@

    Upscaled, Pillar-boxed HD H.264 Access Files from SD NTSC source

    -

    ffmpeg -i input_file -c:v libx264 -filter:v "yadif, scale=1440:1080:flags=lanczos, pad=1920:1080:(ow-iw)/2:(oh-ih)/2, format=yuv420p" output_file

    +

    ffmpeg -i input_file -c:v libx264 -filter:v "yadif, scale=1440:1080:flags=lanczos, pad=1920:1080:(ow-iw)/2:(oh-ih)/2, format=yuv420p" output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -c:v libx264
    encodes video stream with libx264 (h264)
    -filter:v
    a video filter will be used
    "
    quotation mark to start filtergraph
    yadif
    deinterlacing filter (‘yet another deinterlacing filter’)
    - By default, yadif will output one frame for each frame. Outputting one frame for each field (thereby doubling the frame rate) with yadif=1 may produce visually better results.
    + By default, yadif will output one frame for each frame. Outputting one frame for each field (thereby doubling the frame rate) with yadif=1 may produce visually better results.
    scale=1440:1080:flags=lanczos
    resizes the image to 1440x1080, using the Lanczos scaling algorithm, which is slower but better than the default bilinear algorithm.
    pad=1920:1080:(ow-iw)/2:(oh-ih)/2
    pads the area around the 4:3 input video to create a 16:9 output video
    format=yuv420p
    specifies a pixel format of Y′CBCR 4:2:0
    "
    quotation mark to end filtergraph
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    -

    Note: the very same scaling filter also downscales a bigger image size into HD.

    +

    Note: the very same scaling filter also downscales a bigger image size into HD.

    @@ -1160,25 +1160,25 @@

    Deinterlace a video

    -

    ffmpeg -i input_file -c:v libx264 -vf "yadif,format=yuv420p" output_file

    +

    ffmpeg -i input_file -c:v libx264 -vf "yadif,format=yuv420p" output_file

    This command takes an interlaced input file and outputs a deinterlaced H.264 MP4.

    ffmpeg
    starts the command
    -
    -i input file
    path, name and extension of the input file
    +
    -i input file
    path, name and extension of the input file
    -c:v libx264
    tells FFmpeg to encode the video stream as H.264
    -vf
    video filtering will be used (-vf is an alias of -filter:v)
    "
    start of filtergraph (see below)
    yadif
    deinterlacing filter (‘yet another deinterlacing filter’)
    - By default, yadif will output one frame for each frame. Outputting one frame for each field (thereby doubling the frame rate) with yadif=1 may produce visually better results.
    + By default, yadif will output one frame for each frame. Outputting one frame for each field (thereby doubling the frame rate) with yadif=1 may produce visually better results.
    ,
    separates filters
    format=yuv420p
    chroma subsampling set to 4:2:0
    By default, libx264 will use a chroma subsampling scheme that is the closest match to that of the input. This can result in Y′CBCR 4:2:0, 4:2:2, or 4:4:4 chroma subsampling. QuickTime and most other non-FFmpeg based players can’t decode H.264 files that are not 4:2:0, therefore it’s advisable to specify 4:2:0 chroma subsampling.
    "
    end of filtergraph
    -
    output file
    path, name and extension of the output file
    +
    output file
    path, name and extension of the output file

    "yadif,format=yuv420p" is an FFmpeg filtergraph. Here the filtergraph is made up of one filter chain, which is itself made up of the two filters (separated by the comma).
    The enclosing quote marks are necessary when you use spaces within the filtergraph, e.g. -vf "yadif, format=yuv420p", and are included above as an example of good practice.

    -

    Note: FFmpeg includes several deinterlacers apart from yadif: bwdif, w3fdif, kerndeint, and nnedi.

    +

    Note: FFmpeg includes several deinterlacers apart from yadif: bwdif, w3fdif, kerndeint, and nnedi.

    For more H.264 encoding options, see the latter section of the encode H.264 command.

    Example

    @@ -1195,17 +1195,17 @@

    Inverse telecine a video file

    -

    ffmpeg -i input_file -c:v libx264 -vf "fieldmatch,yadif,decimate" output_file

    +

    ffmpeg -i input_file -c:v libx264 -vf "fieldmatch,yadif,decimate" output_file

    The inverse telecine procedure reverses the 3:2 pull down process, restoring 29.97fps interlaced video to the 24fps frame rate of the original film source.

    ffmpeg
    starts the command
    -
    -i input file
    path, name and extension of the input file
    +
    -i input file
    path, name and extension of the input file
    -c:v libx264
    encode video as H.264
    -vf "fieldmatch,yadif,decimate"
    applies these three video filters to the input video.
    Fieldmatch is a field matching filter for inverse telecine - it reconstructs the progressive frames from a telecined stream.
    Yadif (‘yet another deinterlacing filter’) deinterlaces the video. (Note that FFmpeg also includes several other deinterlacers).
    Decimate deletes duplicated frames.
    -
    output file
    path, name and extension of the output file
    +
    output file
    path, name and extension of the output file

    "fieldmatch,yadif,decimate" is an FFmpeg filtergraph. Here the filtergraph is made up of one filter chain, which is itself made up of the three filters (separated by commas).
    The enclosing quote marks are necessary when you use spaces within the filtergraph, e.g. -vf "fieldmatch, yadif, decimate", and are included above as an example of good practice.

    @@ -1226,13 +1226,13 @@

    Change field order of an interlaced video

    -

    ffmpeg -i input_file -c:v video_codec -filter:v setfield=tff output_file

    +

    ffmpeg -i input_file -c:v video_codec -filter:v setfield=tff output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -filter:v setfield=tff
    Sets the field order to top field first. Use setfield=bff for bottom field first.
    -
    -c:v video_codec
    As a video filter is used, it is not possible to use -c copy. The video must be re-encoded with whatever video codec is chosen, e.g. ffv1, v210 or prores.
    -
    output_file
    path, name and extension of the output file
    +
    -c:v video_codec
    As a video filter is used, it is not possible to use -c copy. The video must be re-encoded with whatever video codec is chosen, e.g. ffv1, v210 or prores.
    +
    output_file
    path, name and extension of the output file
    @@ -1243,10 +1243,10 @@

    Check video file interlacement patterns

    -

    ffmpeg -i input file -filter:v idet -f null -

    +

    ffmpeg -i input file -filter:v idet -f null -

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -filter:v idet
    This calls the idet (detect video interlacing type) filter.
    -f null
    Video is decoded with the null muxer. This allows video decoding without creating an output file.
    -
    FFmpeg syntax requires a specified output, and - is just a place holder. No file is actually created.
    @@ -1265,21 +1265,21 @@

    Create centered, transparent text watermark

    E.g For creating access copies with your institutions name

    -

    ffmpeg -i input_file -vf drawtext="fontfile=font_path:fontsize=font_size:text=watermark_text:fontcolor=font_colour:alpha=0.4:x=(w-text_w)/2:y=(h-text_h)/2" output_file

    +

    ffmpeg -i input_file -vf drawtext="fontfile=font_path:fontsize=font_size:text=watermark_text:fontcolor=font_colour:alpha=0.4:x=(w-text_w)/2:y=(h-text_h)/2" output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -vf drawtext=
    This calls the drawtext filter with the following options:
    -
    fontfile=font_path
    Set path to font. For example in macOS: fontfile=/Library/Fonts/AppleGothic.ttf
    -
    fontsize=font_size
    Set font size. 35 is a good starting point for SD. Ideally this value is proportional to video size, for example use ffprobe to acquire video height and divide by 14.
    -
    text=watermark_text
    Set the content of your watermark text. For example: text='FFMPROVISR EXAMPLE TEXT'
    -
    fontcolor=font_colour
    Set colour of font. Can be a text string such as fontcolor=white or a hexadecimal value such as fontcolor=0xFFFFFF
    +
    fontfile=font_path
    Set path to font. For example in macOS: fontfile=/Library/Fonts/AppleGothic.ttf
    +
    fontsize=font_size
    Set font size. 35 is a good starting point for SD. Ideally this value is proportional to video size, for example use ffprobe to acquire video height and divide by 14.
    +
    text=watermark_text
    Set the content of your watermark text. For example: text='FFMPROVISR EXAMPLE TEXT'
    +
    fontcolor=font_colour
    Set colour of font. Can be a text string such as fontcolor=white or a hexadecimal value such as fontcolor=0xFFFFFF
    alpha=0.4
    Set transparency value.
    -
    x=(w-text_w)/2:y=(h-text_h)/2
    Sets x and y coordinates for the watermark. These relative values will centre your watermark regardless of video dimensions.
    +
    x=(w-text_w)/2:y=(h-text_h)/2
    Sets x and y coordinates for the watermark. These relative values will centre your watermark regardless of video dimensions.
    Note: -vf is a shortcut for -filter:v.
    -
    output_file
    path, name and extension of the output file.
    +
    output_file
    path, name and extension of the output file.
    @@ -1290,13 +1290,13 @@

    Overlay image watermark on video

    -

    ffmpeg -i input_video file -i input_image_file -filter_complex overlay=main_w-overlay_w-5:5 output_file

    +

    ffmpeg -i input_video file -i input_image_file -filter_complex overlay=main_w-overlay_w-5:5 output_file

    ffmpeg
    starts the command
    -
    -i input_video_file
    path, name and extension of the input video file
    -
    -i input_image_file
    path, name and extension of the image file
    +
    -i input_video_file
    path, name and extension of the input video file
    +
    -i input_image_file
    path, name and extension of the image file
    -filter_complex overlay=main_w-overlay_w-5:5
    This calls the overlay filter and sets x and y coordinates for the position of the watermark on the video. Instead of hardcoding specific x and y coordinates, main_w-overlay_w-5:5 uses relative coordinates to place the watermark in the upper right hand corner, based on the width of your input files. Please see the FFmpeg documentation for more examples.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -1307,22 +1307,22 @@

    Create a burnt in timecode on your image

    -

    ffmpeg -i input_file -vf drawtext="fontfile=font_path:fontsize=font_size:timecode=starting_timecode:fontcolor=font_colour:box=1:boxcolor=box_colour:rate=timecode_rate:x=(w-text_w)/2:y=h/1.2" output_file

    +

    ffmpeg -i input_file -vf drawtext="fontfile=font_path:fontsize=font_size:timecode=starting_timecode:fontcolor=font_colour:box=1:boxcolor=box_colour:rate=timecode_rate:x=(w-text_w)/2:y=h/1.2" output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -vf drawtext=
    This calls the drawtext filter with the following options:
    "
    quotation mark to start drawtext filter command
    -
    fontfile=font_path
    Set path to font. For example in macOS: fontfile=/Library/Fonts/AppleGothic.ttf
    -
    fontsize=font_size
    Set font size. 35 is a good starting point for SD. Ideally this value is proportional to video size, for example use ffprobe to acquire video height and divide by 14.
    -
    timecode=starting_timecode
    Set the timecode to be displayed for the first frame. Timecode is to be represented as hh:mm:ss[:;.]ff. Colon escaping is determined by O.S, for example in Ubuntu timecode='09\\:50\\:01\\:23'. Ideally, this value would be generated from the file itself using ffprobe.
    -
    fontcolor=font_colour
    Set colour of font. Can be a text string such as fontcolor=white or a hexadecimal value such as fontcolor=0xFFFFFF
    +
    fontfile=font_path
    Set path to font. For example in macOS: fontfile=/Library/Fonts/AppleGothic.ttf
    +
    fontsize=font_size
    Set font size. 35 is a good starting point for SD. Ideally this value is proportional to video size, for example use ffprobe to acquire video height and divide by 14.
    +
    timecode=starting_timecode
    Set the timecode to be displayed for the first frame. Timecode is to be represented as hh:mm:ss[:;.]ff. Colon escaping is determined by O.S, for example in Ubuntu timecode='09\\:50\\:01\\:23'. Ideally, this value would be generated from the file itself using ffprobe.
    +
    fontcolor=font_colour
    Set colour of font. Can be a text string such as fontcolor=white or a hexadecimal value such as fontcolor=0xFFFFFF
    box=1
    Enable box around timecode
    -
    boxcolor=box_colour
    Set colour of box. Can be a text string such as fontcolor=black or a hexadecimal value such as fontcolor=0x000000
    -
    rate=timecode_rate
    Framerate of video. For example 25/1
    -
    x=(w-text_w)/2:y=h/1.2
    Sets x and y coordinates for the timecode. These relative values will horizontally centre your timecode in the bottom third regardless of video dimensions.
    +
    boxcolor=box_colour
    Set colour of box. Can be a text string such as fontcolor=black or a hexadecimal value such as fontcolor=0x000000
    +
    rate=timecode_rate
    Framerate of video. For example 25/1
    +
    x=(w-text_w)/2:y=h/1.2
    Sets x and y coordinates for the timecode. These relative values will horizontally centre your timecode in the bottom third regardless of video dimensions.
    "
    quotation mark to end drawtext filter command
    -
    output_file
    path, name and extension of the output file.
    +
    output_file
    path, name and extension of the output file.

    Note: -vf is a shortcut for -filter:v.

    @@ -1334,14 +1334,14 @@

    Embed a subtitle file into a movie file

    -

    ffmpeg -i input_file -i subtitles_file -c copy -c:s mov_text output_file

    +

    ffmpeg -i input_file -i subtitles_file -c copy -c:s mov_text output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    -
    -i subtitles_file
    path to subtitles file, e.g. subtitles.srt
    +
    -i input_file
    path, name and extension of the input file
    +
    -i subtitles_file
    path to subtitles file, e.g. subtitles.srt
    -c copy
    enable stream copy (no re-encode)
    -c:s mov_text
    Encode subtitles using the mov_text codec. Note: The mov_text codec works for MP4 and MOV containers. For the MKV container, acceptable formats are ASS, SRT, and SSA.
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file

    Note: -c:s is a shortcut for -scodec

    @@ -1357,14 +1357,14 @@

    One thumbnail

    -

    ffmpeg -i input_file -ss 00:00:20 -vframes 1 thumb.png

    +

    ffmpeg -i input_file -ss 00:00:20 -vframes 1 thumb.png

    This command will grab a thumbnail 20 seconds into the video.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    -
    -ss 00:00:20
    seeks video file to 20 seconds into the video
    -
    -vframes 1
    sets the number of frames (in this example, one frame)
    -
    output file
    path, name and extension of the output file
    +
    -i input_file
    path, name and extension of the input file
    +
    -ss 00:00:20
    seeks video file to 20 seconds into the video
    +
    -vframes 1
    sets the number of frames (in this example, one frame)
    +
    output file
    path, name and extension of the output file
    @@ -1375,14 +1375,14 @@

    Many thumbnails

    -

    ffmpeg -i input_file -vf fps=1/60 out%d.png

    +

    ffmpeg -i input_file -vf fps=1/60 out%d.png

    This will grab a thumbnail every minute and output sequential png files.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    -
    -ss 00:00:20
    seeks video file to 20 seconds into the video
    +
    -i input_file
    path, name and extension of the input file
    +
    -ss 00:00:20
    seeks video file to 20 seconds into the video
    -vf fps=1/60
    Creates a filtergraph to use for the streams. The rest of the command identifies filtering by frames per second, and sets the frames per second at 1/60 (which is one per minute). Omitting this will output all frames from the video.
    -
    output file
    path, name and extension of the output file. In the example out%d.png where %d is a regular expression that adds a number (d is for digit) and increments with each frame (out1.png, out2.png, out3.png…). You may also chose a regular expression like out%04d.png which gives 4 digits with leading 0 (out0001.png, out0002.png, out0003.png, …).
    +
    output file
    path, name and extension of the output file. In the example out%d.png where %d is a regular expression that adds a number (d is for digit) and increments with each frame (out1.png, out2.png, out3.png…). You may also chose a regular expression like out%04d.png which gives 4 digits with leading 0 (out0001.png, out0002.png, out0003.png, …).
    @@ -1393,17 +1393,17 @@

    Images to GIF

    -

    ffmpeg -f image2 -framerate 9 -pattern_type glob -i "input_image_*.jpg" -vf scale=250x250 output_file.gif

    +

    ffmpeg -f image2 -framerate 9 -pattern_type glob -i "input_image_*.jpg" -vf scale=250x250 output_file.gif

    This will convert a series of image files into a GIF.

    ffmpeg
    starts the command
    -f image2
    forces input or output file format. image2 specifies the image file demuxer.
    -framerate 9
    sets framerate to 9 frames per second
    -pattern_type glob
    tells FFmpeg that the following mapping should "interpret like a glob" (a "global command" function that relies on the * as a wildcard and finds everything that matches)
    -
    -i "input_image_*.jpg"
    maps all files in the directory that start with input_image_, for example input_image_001.jpg, input_image_002.jpg, input_image_003.jpg... etc.
    +
    -i "input_image_*.jpg"
    maps all files in the directory that start with input_image_, for example input_image_001.jpg, input_image_002.jpg, input_image_003.jpg... etc.
    (The quotation marks are necessary for the above “glob” pattern!)
    -vf scale=250x250
    filter the video to scale it to 250x250; -vf is an alias for -filter:v
    -
    output_file.gif
    path and name of the output file
    +
    output_file.gif
    path and name of the output file
    @@ -1415,20 +1415,20 @@

    Create GIF

    Create high quality GIF

    -

    ffmpeg -ss HH:MM:SS -i input_file -filter_complex "fps=10,scale=500:-1:flags=lanczos,palettegen" -t 3 palette.png

    -

    ffmpeg -ss HH:MM:SS -i input_file -i palette.png -filter_complex "[0:v]fps=10, scale=500:-1:flags=lanczos[v], [v][1:v]paletteuse" -t 3 -loop 6 output_file

    +

    ffmpeg -ss HH:MM:SS -i input_file -filter_complex "fps=10,scale=500:-1:flags=lanczos,palettegen" -t 3 palette.png

    +

    ffmpeg -ss HH:MM:SS -i input_file -i palette.png -filter_complex "[0:v]fps=10, scale=500:-1:flags=lanczos[v], [v][1:v]paletteuse" -t 3 -loop 6 output_file

    The first command will use the palettegen filter to create a custom palette, then the second command will create the GIF with the paletteuse filter. The result is a high quality GIF.

    ffmpeg
    starts the command
    -
    -ss HH:MM:SS
    starting point of the GIF. If a plain numerical value is used it will be interpreted as seconds
    -
    -i input_file
    path, name and extension of the input file
    -
    -filter_complex "fps=framerate, scale=width:height, palettegen"
    a complex filtergraph.
    +
    -ss HH:MM:SS
    starting point of the GIF. If a plain numerical value is used it will be interpreted as seconds
    +
    -i input_file
    path, name and extension of the input file
    +
    -filter_complex "fps=framerate, scale=width:height, palettegen"
    a complex filtergraph.
    Firstly, the fps filter sets the frame rate.
    - Then the scale filter resizes the image. You can specify both the width and the height, or specify a value for one and use a scale value of -1 for the other to preserve the aspect ratio. (For example, 500:-1 would create a GIF 500 pixels wide and with a height proportional to the original video). In the first script above, :flags=lanczos specifies that the Lanczos rescaling algorithm will be used to resize the image.
    + Then the scale filter resizes the image. You can specify both the width and the height, or specify a value for one and use a scale value of -1 for the other to preserve the aspect ratio. (For example, 500:-1 would create a GIF 500 pixels wide and with a height proportional to the original video). In the first script above, :flags=lanczos specifies that the Lanczos rescaling algorithm will be used to resize the image.
    Lastly, the palettegen filter generates the palette.
    -
    -t 3
    duration in seconds (here 3; can be specified also with a full timestamp, i.e. here 00:00:03)
    -
    -loop 6
    sets the number of times to loop the GIF. A value of -1 will disable looping. Omitting -loop will use the default, which will loop infinitely.
    -
    output_file
    path, name and extension of the output file
    +
    -t 3
    duration in seconds (here 3; can be specified also with a full timestamp, i.e. here 00:00:03)
    +
    -loop 6
    sets the number of times to loop the GIF. A value of -1 will disable looping. Omitting -loop will use the default, which will loop infinitely.
    +
    output_file
    path, name and extension of the output file

    The second command has a slightly different filtergraph, which breaks down as follows:

    @@ -1436,7 +1436,7 @@ [v][1:v]paletteuse": applies the paletteuse filter, setting the second input file (the palette) as the reference file.

    Simpler GIF creation

    -

    ffmpeg -ss HH:MM:SS -i input_file -vf "fps=10,scale=500:-1" -t 3 -loop 6 output_file

    +

    ffmpeg -ss HH:MM:SS -i input_file -vf "fps=10,scale=500:-1" -t 3 -loop 6 output_file

    This is a quick and easy method. Dithering is more apparent than the above method using the palette filters, but the file size will be smaller. Perfect for that “legacy” GIF look.

    @@ -1451,15 +1451,15 @@

    Transcode an image sequence into uncompressed 10-bit video

    -

    ffmpeg -f image2 -framerate 24 -i input_file_%06d.ext -c:v v210 output_file

    +

    ffmpeg -f image2 -framerate 24 -i input_file_%06d.ext -c:v v210 output_file

    ffmpeg
    starts the command
    -f image2
    forces the image file de-muxer for single image files
    -framerate 24
    Sets the input framerate to 24 fps. The image2 demuxer defaults to 25 fps.
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    This must match the naming convention actually used! The regex %06d matches six digits long numbers, possibly with leading zeroes. This allows to read in ascending order, one image after the other, the full sequence inside one folder. For image sequences starting with 086400 (i.e. captured with a timecode starting at 01:00:00:00 and at 24 fps), add the flag -start_number 086400 before -i input_file_%06d.ext. The extension for TIFF files is .tif or maybe .tiff; the extension for DPX files is .dpx (or eventually .cin for old files).
    -c:v v210
    encodes an uncompressed 10-bit video stream
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file
    @@ -1470,18 +1470,18 @@

    Create a video from an image and audio file.

    -

    ffmpeg -r 1 -loop 1 -i image_file -i audio_file -acodec copy -shortest -vf scale=1280:720 output_file

    +

    ffmpeg -r 1 -loop 1 -i image_file -i audio_file -acodec copy -shortest -vf scale=1280:720 output_file

    This command will take an image file (e.g. image.jpg) and an audio file (e.g. audio.mp3) and combine them into a video file that contains the audio track with the image used as the video. It can be useful in a situation where you might want to upload an audio file to a platform like YouTube. You may want to adjust the scaling with -vf to suit your needs.

    ffmpeg
    starts the command
    -
    -r 1
    set the framerate
    -
    -loop 1
    loop the first input stream
    -
    -i image_file
    path, name and extension of the image file
    -
    -i audio_file
    path, name and extension of the audio file
    +
    -r 1
    set the framerate
    +
    -loop 1
    loop the first input stream
    +
    -i image_file
    path, name and extension of the image file
    +
    -i audio_file
    path, name and extension of the audio file
    -acodec copy
    copy the audio. -acodec is an alias for -c:a
    -shortest
    finish encoding when the shortest input stream ends
    -vf scale=1280:720
    filter the video to scale it to 1280x720 for YouTube. -vf is an alias for -filter:v
    -
    output_file
    path, name and extension of the video output file
    +
    output_file
    path, name and extension of the video output file
    @@ -1496,12 +1496,12 @@

    Creates a visualization of the bits in an audio stream

    -

    ffplay -f lavfi "amovie=input_file, asplit=2[out1][a], [a]abitscope=colors=purple|yellow[out0]"

    +

    ffplay -f lavfi "amovie=input_file, asplit=2[out1][a], [a]abitscope=colors=purple|yellow[out0]"

    This filter allows visual analysis of the information held in various bit depths of an audio stream. This can aid with identifying when a file that is nominally of a higher bit depth actually has been 'padded' with null information. The provided GIF shows a 16 bit WAV file (left) and then the results of converting that same WAV to 32 bit (right). Note that in the 32 bit version, there is still only information in the first 16 bits.

    ffplay -f lavfi
    starts the command and tells ffplay that you will be using the lavfi virtual device to create the input
    "
    quotation mark to start the lavfi filtergraph
    -
    amovie=input_file
    path, name and extension of the input file
    +
    amovie=input_file
    path, name and extension of the input file
    asplit=2[out1][a]
    splits the audio stream in two. One of these [a] will be passed to the filter, and the other [out1] will be the audible stream.
    [a]abitscope=colors=purple|yellow[out0]
    sends stream [a] into the abitscope filter, sets the colors for the channels to purple and yellow, and outputs the results to [out0]. This is what will be the visualization.
    "
    quotation mark to end the lavfi filtergraph
    @@ -1524,7 +1524,7 @@
    ffplay
    starts the command
    -f lavfi
    tells ffplay to use the Libavfilter input virtual device
    "
    quotation mark to start the lavfi filtergraph
    -
    movie='input.mp3'
    declares audio source file on which to apply filter
    +
    movie='input.mp3'
    declares audio source file on which to apply filter
    ,
    comma signifies the end of audio source section and the beginning of the filter section
    astats=metadata=1
    tells the astats filter to ouput metadata that can be passed to another filter (in this case adrawgraph)
    :
    divides between options of the same filter
    @@ -1548,12 +1548,12 @@

    Shows all pixels outside of broadcast range

    -

    ffplay -f lavfi "movie='input.mp4', signalstats=out=brng:color=cyan[out]"

    +

    ffplay -f lavfi "movie='input.mp4', signalstats=out=brng:color=cyan[out]"

    ffplay
    starts the command
    -f lavfi
    tells ffplay to use the Libavfilter input virtual device
    "
    quotation mark to start the lavfi filtergraph
    -
    movie='input.mp4'
    declares video file source to apply filter
    +
    movie='input.mp4'
    declares video file source to apply filter
    ,
    comma signifies closing of video source assertion and ready for filter assertion
    signalstats=out=brng
    tells ffplay to use the signalstats command, output the data, use the brng filter
    :
    indicates there’s another parameter coming
    @@ -1573,10 +1573,10 @@

    Plays vectorscope of video

    -

    ffplay input_file -vf "split=2[m][v], [v]vectorscope=b=0.7:m=color3:g=green[v], [m][v]overlay=x=W-w:y=H-h"

    +

    ffplay input_file -vf "split=2[m][v], [v]vectorscope=b=0.7:m=color3:g=green[v], [m][v]overlay=x=W-w:y=H-h"

    ffplay
    starts the command
    -
    input_file
    path, name and extension of the input file
    +
    input_file
    path, name and extension of the input file
    -vf
    creates a filtergraph to use for the streams
    "
    quotation mark to start filtergraph
    split=2[m][v]
    Splits the input into two identical outputs and names them [m] and [v]
    @@ -1598,7 +1598,7 @@

    ffmpeg -i input01 -i input02 -filter_complex "[0:v:0]tblend=all_mode=difference128[a];[1:v:0]tblend=all_mode=difference128[b];[a][b]hstack[out]" -map [out] -f nut -c:v rawvideo - | ffplay -

    ffmpeg
    starts the command
    -
    -i input01 -i input02
    Designates the files to use for inputs one and two respectively
    +
    -i input01 -i input02
    Designates the files to use for inputs one and two respectively
    -filter_complex
    Lets FFmpeg know we will be using a complex filter (this must be used for multiple inputs)
    "
    quotation mark to start filtergraph
    [0:v:0]tblend=all_mode=difference128[a]
    Applies the tblend filter (with the settings all_mode and difference128) to the first video stream from the first input and assigns the result to the output [a]
    @@ -1629,11 +1629,11 @@

    Pull specs from video file

    -

    ffprobe -i input_file -show_format -show_streams -show_data -print_format xml

    +

    ffprobe -i input_file -show_format -show_streams -show_data -print_format xml

    This command extracts technical metadata from a video file and displays it in xml.

    ffprobe
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -show_format
    outputs file container informations
    -show_streams
    outputs audio and video codec informations
    -show_data
    adds a short “hexdump” to show_streams command output
    @@ -1649,14 +1649,14 @@

    Strips metadata from video file

    -

    ffmpeg -i input_file -map_metadata -1 -c:v copy -c:a copy output_file

    +

    ffmpeg -i input_file -map_metadata -1 -c:v copy -c:a copy output_file

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -map_metadata -1
    sets metadata copying to -1, which copies nothing
    -c:v copy
    copies video track
    -c:a copy
    copies audio track
    -
    output_file
    Makes copy of original file and names output file
    +
    output_file
    Makes copy of original file and names output file

    Note: -c:v and -c:a are shortcuts for -vcodec and -acodec.

    @@ -1679,17 +1679,17 @@
    for file in *.mxf
    starts the loop, and states what the input files will be. Here, the FFmpeg command within the loop will be applied to all files with an extension of .mxf.
    The word ‘file’ is an arbitrary variable which will represent each .mxf file in turn as it is looped over.
    do ffmpeg -i "$file"
    carry out the following FFmpeg command for each input file.
    - Per Bash syntax, within the command the variable is referred to by “$file”. The dollar sign is used to reference the variable ‘file’, and the enclosing quotation marks prevents reinterpretation of any special characters that may occur within the filename, ensuring that the original filename is retained.
    + Per Bash syntax, within the command the variable is referred to by “$file”. The dollar sign is used to reference the variable ‘file’, and the enclosing quotation marks prevents reinterpretation of any special characters that may occur within the filename, ensuring that the original filename is retained.
    -map 0
    retain all streams
    -c copy
    enable stream copy (no re-encode)
    "${file%.mxf}.mov";
    retaining the original file name, set the output file wrapper as .mov
    done
    complete; all items have been processed.
    -

    Note: the shell script (.sh file) and all .mxf files to be processed must be contained within the same directory, and the script must be run from that directory.
    +

    Note: the shell script (.sh file) and all .mxf files to be processed must be contained within the same directory, and the script must be run from that directory.
    Execute the .sh file with the command sh Rewrap-MXF.sh.

    Modify the script as needed to perform different transcodes, or to use with ffprobe. :)

    The basic pattern will look similar to this:
    - for item in *.ext; do ffmpeg -i $item (FFmpeg options here) "${item%.ext}_suffix.ext"

    + for item in *.ext; do ffmpeg -i $item (FFmpeg options here) "${item%.ext}_suffix.ext"

    e.g., if an input file is bestmovie002.avi, its output will be bestmovie002_suffix.avi.

    Variation: recursively process all MXF files in subdirectories using find instead of for:

    find input_directory -iname "*.mxf" -exec ffmpeg -i {} -map 0 -c copy {}.mov \;

    @@ -1703,7 +1703,7 @@

    Create PowerShell script to batch process with FFmpeg

    As of Windows 10, it is possible to run Bash via Bash on Ubuntu on Windows, allowing you to use bash scripting. To enable Bash on Windows, see these instructions.

    -

    On Windows, the primary native command line programme is PowerShell. PowerShell scripts are plain text files saved with a .ps1 extension. This entry explains how they work with the example of a PowerShell script named “rewrap-mp4.ps1”, which rewraps .mp4 files in a given directory to .mkv files.

    +

    On Windows, the primary native command line programme is PowerShell. PowerShell scripts are plain text files saved with a .ps1 extension. This entry explains how they work with the example of a PowerShell script named “rewrap-mp4.ps1”, which rewraps .mp4 files in a given directory to .mkv files.

    “rewrap-mp4.ps1” contains the following text:

    $inputfiles = ls *.mp4
       foreach ($file in $inputfiles) {
    @@ -1718,13 +1718,13 @@
             
    {
    Opens the code block.
    $output = [io.path]::ChangeExtension($file, '.mkv')
    Sets up the output file: it will be located in the current folder and keep the same filename, but will have an .mkv extension instead of .mp4.
    ffmpeg -i $file
    Carry out the following FFmpeg command for each input file.
    - Note: To call FFmpeg here as just ‘ffmpeg’ (rather than entering the full path to ffmpeg.exe), you must make sure that it’s correctly configured. See this article, especially the section ‘Add to Path’.
    + Note: To call FFmpeg here as just ‘ffmpeg’ (rather than entering the full path to ffmpeg.exe), you must make sure that it’s correctly configured. See this article, especially the section ‘Add to Path’.
    -map 0
    retain all streams
    -c copy
    enable stream copy (no re-encode)
    $output
    The output file is set to the value of the $output variable declared above: i.e., the current file name with an .mkv extension.
    }
    Closes the code block.
    -

    Note: the PowerShell script (.ps1 file) and all .mp4 files to be rewrapped must be contained within the same directory, and the script must be run from that directory.

    +

    Note: the PowerShell script (.ps1 file) and all .mp4 files to be rewrapped must be contained within the same directory, and the script must be run from that directory.

    Execute the .ps1 file by typing .\rewrap-mp4.ps1 in PowerShell.

    Modify the script as needed to perform different transcodes, or to use with ffprobe. :)

    @@ -1736,11 +1736,11 @@

    Check decoder errors

    -

    ffmpeg -i input_file -f null -

    +

    ffmpeg -i input_file -f null -

    This decodes your video and prints any errors or found issues to the screen.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -f null
    Video is decoded with the null muxer. This allows video decoding without creating an output file.
    -
    FFmpeg syntax requires a specified output, and - is just a place holder. No file is actually created.
    @@ -1753,13 +1753,13 @@

    Check FFV1 Version 3 fixity

    -

    ffmpeg -report -i input_file -f null -

    +

    ffmpeg -report -i input_file -f null -

    This decodes your video and displays any CRC checksum mismatches. These errors will display in your terminal like this: [ffv1 @ 0x1b04660] CRC mismatch 350FBD8A!at 0.272000 seconds

    Frame CRCs are enabled by default in FFV1 Version 3.

    ffmpeg
    starts the command
    -
    -report
    Dump full command line and console output to a file named ffmpeg-YYYYMMDD-HHMMSS.log in the current directory. It also implies -loglevel verbose.
    -
    -i input_file
    path, name and extension of the input file
    +
    -report
    Dump full command line and console output to a file named ffmpeg-YYYYMMDD-HHMMSS.log in the current directory. It also implies -loglevel verbose.
    +
    -i input_file
    path, name and extension of the input file
    -f null
    Video is decoded with the null muxer. This allows video decoding without creating an output file.
    -
    FFmpeg syntax requires a specified output, and - is just a place holder. No file is actually created.
    @@ -1772,14 +1772,14 @@

    Create MD5 checksums (video frames)

    -

    ffmpeg -i input_file -f framemd5 -an output_file

    +

    ffmpeg -i input_file -f framemd5 -an output_file

    This will create an MD5 checksum per video frame.

    ffmpeg
    starts the command
    -
    -i input_file
    path, name and extension of the input file
    +
    -i input_file
    path, name and extension of the input file
    -f framemd5
    library used to calculate the MD5 checksums
    -an
    ignores the audio stream (audio no)
    -
    output_file
    path, name and extension of the output file
    +
    output_file
    path, name and extension of the output file

    You may verify an MD5 checksum file created this way by using a Bash script.

    @@ -1791,7 +1791,7 @@

    Create MD5 checksums (audio samples)

    -

    ffmpeg -i input_file -af "asetnsamples=n=48000" -f framemd5 -vn output_file

    +

    ffmpeg -i input_file -af "asetnsamples=n=48000" -f framemd5 -vn output_file

    This will create an MD5 checksum for each group of 48000 audio samples.
    The number of samples per group can be set arbitrarily, but it's good practice to match the samplerate of the media file (so you will get one checksum per second).

    Examples for other samplerates:

    @@ -1799,14 +1799,14 @@
  • 44.1 kHz: "asetnsamples=n=44100"
  • 96 kHz: "asetnsamples=n=96000"
-

Note: This filter trandscodes audio to 16 bit PCM by default. The generated framemd5s will represent this value. Validating these framemd5s will require using the same default settings. Alternatively, when your file has another quantisation rates (e.g. 24 bit), then you might add the audio codec -c:a pcm_s24le to the command, for compatibility reasons with other tools, like BWF MetaEdit.

+

Note: This filter trandscodes audio to 16 bit PCM by default. The generated framemd5s will represent this value. Validating these framemd5s will require using the same default settings. Alternatively, when your file has another quantisation rates (e.g. 24 bit), then you might add the audio codec -c:a pcm_s24le to the command, for compatibility reasons with other tools, like BWF MetaEdit.

ffmpeg
starts the command
-
-i input_file
path, name and extension of the input file
-
-af "asetnsamples=n=48000"
the audio filter sets the sampling rate
+
-i input_file
path, name and extension of the input file
+
-af "asetnsamples=n=48000"
the audio filter sets the sampling rate
-f framemd5
library used to calculate the MD5 checksums
-vn
ignores the video stream (video no)
-
output_file
path, name and extension of the output file
+
output_file
path, name and extension of the output file

You may verify an MD5 checksum file created this way by using a Bash script.

@@ -1818,19 +1818,19 @@

Create stream MD5s

-

ffmpeg -i input_file -map 0:v:0 -c:v copy -f md5 output_file_1 -map 0:a:0 -c:a copy -f md5 output_file_2

+

ffmpeg -i input_file -map 0:v:0 -c:v copy -f md5 output_file_1 -map 0:a:0 -c:a copy -f md5 output_file_2

This will create MD5 checksums for the first video and the first audio stream in a file. If only one of these is necessary (for example if used on a WAV file) either part of the command can be excluded to create the desired MD5 only. Use of this kind of checksum enables integrity of the A/V information to be verified independently of any changes to surrounding metadata.

ffmpeg
starts the command
-
-i input_file
path, name and extension of the input file
+
-i input_file
path, name and extension of the input file
-map 0:v:0
selects the first video stream from the input
-c:v copy
ensures that FFmpeg will not transcode the video to a different codec before generating the MD5
-
output_file_1
is the output file for the video stream MD5. Example file extensions are .md5 and .txt
+
output_file_1
is the output file for the video stream MD5. Example file extensions are .md5 and .txt
-map 0:a:0
selects the first audio stream from the input
-c:a copy
ensures that FFmpeg will not transcode the audio to a different codec before generating the MD5 (by default FFmpeg will use 16 bit PCM for audio MD5s).
-
output_file_2
is the output file for the audio stream MD5.
+
output_file_2
is the output file for the audio stream MD5.
-

Note:The MD5s generated by running this command on WAV files are compatible with those embedded by the BWF MetaEdit tool and can be compared.

+

Note:The MD5s generated by running this command on WAV files are compatible with those embedded by the BWF MetaEdit tool and can be compared.

@@ -1840,13 +1840,13 @@

Creates a QCTools report

-

ffprobe -f lavfi -i "movie=input_file:s=v+a[in0][in1], [in0]signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom, split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim[out0];[in1]ebur128=metadata=1, astats=metadata=1:reset=1:length=0.4[out1]" -show_frames -show_versions -of xml=x=1:q=1 -noprivate | gzip > input_file.qctools.xml.gz

+

ffprobe -f lavfi -i "movie=input_file:s=v+a[in0][in1], [in0]signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom, split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim[out0];[in1]ebur128=metadata=1, astats=metadata=1:reset=1:length=0.4[out1]" -show_frames -show_versions -of xml=x=1:q=1 -noprivate | gzip > input_file.qctools.xml.gz

This will create an XML report for use in QCTools for a video file with one video track and one audio track. See also the QCTools documentation.

ffprobe
starts the command
-f lavfi
tells ffprobe to use the Libavfilter input virtual device
-i
input file and parameters
-
"movie=input_file:s=v+a[in0][in1], [in0]signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom, split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim[out0];[in1]ebur128=metadata=1, astats=metadata=1:reset=1:length=0.4[out1]"
+
"movie=input_file:s=v+a[in0][in1], [in0]signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom, split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim[out0];[in1]ebur128=metadata=1, astats=metadata=1:reset=1:length=0.4[out1]"
This very large lump of commands declares the input file and passes in a request for all potential data signal information for a file with one video and one audio track
-show_frames
asks for information about each frame and subtitle contained in the input multimedia stream
-show_versions
asks for information related to program and library versions
@@ -1854,7 +1854,7 @@
-noprivate
hides any private data that might exist in the file
| gzip
The | is to "pipe" (or push) the data into a compressed file format
>
redirects the standard output (the data made by ffprobe about the video)
-
input_file.qctools.xml.gz
names the zipped data output file, which can be named anything but needs the extension qctools.xml.gz for compatibility issues
+
input_file.qctools.xml.gz
names the zipped data output file, which can be named anything but needs the extension qctools.xml.gz for compatibility issues
@@ -1865,13 +1865,13 @@

Creates a QCTools report

-

ffprobe -f lavfi -i "movie=input_file,signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom,split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim" -show_frames -show_versions -of xml=x=1:q=1 -noprivate | gzip > input_file.qctools.xml.gz

+

ffprobe -f lavfi -i "movie=input_file,signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom,split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim" -show_frames -show_versions -of xml=x=1:q=1 -noprivate | gzip > input_file.qctools.xml.gz

This will create an XML report for use in QCTools for a video file with one video track and NO audio track. See also the QCTools documentation.

ffprobe
starts the command
-f lavfi
tells ffprobe to use the Libavfilter input virtual device
-i
input file and parameters
-
"movie=input_file,signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom,split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim"
+
"movie=input_file,signalstats=stat=tout+vrep+brng, cropdetect=reset=1:round=1, idet=half_life=1, split[a][b];[a]field=top[a1];[b]field=bottom,split[b1][b2];[a1][b1]psnr[c1];[c1][b2]ssim"
This very large lump of commands declares the input file and passes in a request for all potential data signal information for a file with one video and one audio track
-show_frames
asks for information about each frame and subtitle contained in the input multimedia stream
-show_versions
asks for information related to program and library versions
@@ -1879,7 +1879,7 @@
-noprivate
hides any private data that might exist in the file
| gzip
The | is to "pipe" (or push) the data into a compressed file format
>
redirects the standard output (the data made by ffprobe about the video)
-
input_file.qctools.xml.gz
names the zipped data output file, which can be named anything but needs the extension qctools.xml.gz for compatibility issues
+
input_file.qctools.xml.gz
names the zipped data output file, which can be named anything but needs the extension qctools.xml.gz for compatibility issues
@@ -1890,16 +1890,16 @@

Read/Extract EIA-608 (Line 21) closed captioning

-

ffprobe -f lavfi -i movie=input_file,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.line,lavfi.readeia608.0.cc,lavfi.readeia608.1.line,lavfi.readeia608.1.cc -of csv > input_file.csv

+

ffprobe -f lavfi -i movie=input_file,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.line,lavfi.readeia608.0.cc,lavfi.readeia608.1.line,lavfi.readeia608.1.cc -of csv > input_file.csv

This command uses FFmpeg's readeia608 filter to extract the hexadecimal values hidden within EIA-608 (Line 21) Closed Captioning, outputting a csv file. For more information about EIA-608, check out Adobe's Introduction to Closed Captions.

If hex isn't your thing, closed captioning character and code sets can be found in the documentation for SCTools.

ffprobe
starts the command
-f lavfi
tells ffprobe to use the libavfilter input virtual device
-
-i input_file
input file and parameters
+
-i input_file
input file and parameters
readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.line,lavfi.readeia608.0.cc,lavfi.readeia608.1.line,lavfi.readeia608.1.cc -of csv
specifies the first two lines of video in which EIA-608 data (hexadecimal byte pairs) are identifiable by ffprobe, outputting comma separated values (CSV)
>
redirects the standard output (the data created by ffprobe about the video)
-
output_file.csv
names the CSV output file
+
output_file.csv
names the CSV output file

Example

@@ -1919,14 +1919,14 @@

Makes a mandelbrot test pattern video

-

ffmpeg -f lavfi -i mandelbrot=size=1280x720:rate=25 -c:v libx264 -t 10 output_file

+

ffmpeg -f lavfi -i mandelbrot=size=1280x720:rate=25 -c:v libx264 -t 10 output_file

ffmpeg
starts the command
-f lavfi
tells FFmpeg to use the Libavfilter input virtual device
-i mandelbrot=size=1280x720:rate=25
asks for the mandelbrot test filter as input. Adjusting the size and rate options allows you to choose a specific frame size and framerate.
-c:v libx264
transcodes video from rawvideo to H.264. Set -pix_fmt to yuv420p for greater H.264 compatibility with media players.
-t 10
specifies recording time of 10 seconds
-
output_file
path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
+
output_file
path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
@@ -1937,14 +1937,14 @@

Makes a SMPTE bars test pattern video

-

ffmpeg -f lavfi -i smptebars=size=720x576:rate=25 -c:v prores -t 10 output_file

+

ffmpeg -f lavfi -i smptebars=size=720x576:rate=25 -c:v prores -t 10 output_file

ffmpeg
starts the command
-f lavfi
tells FFmpeg to use the Libavfilter input virtual device
-i smptebars=size=720x576:rate=25
asks for the smptebars test filter as input. Adjusting the size and rate options allows you to choose a specific frame size and framerate.
-c:v prores
transcodes video from rawvideo to Apple ProRes 4:2:2.
-t 10
specifies recording time of 10 seconds
-
output_file
path, name and extension of the output file. Try different file extensions such as mov or avi.
+
output_file
path, name and extension of the output file. Try different file extensions such as mov or avi.
@@ -1955,7 +1955,7 @@

Make a test pattern video

-

ffmpeg -f lavfi -i testsrc=size=720x576:rate=25 -c:v v210 -t 10 output_file

+

ffmpeg -f lavfi -i testsrc=size=720x576:rate=25 -c:v v210 -t 10 output_file

ffmpeg
starts the command
-f lavfi
tells FFmpeg to use the libavfilter input virtual device
@@ -1963,7 +1963,7 @@ The different test patterns that can be generated are listed here.
-c:v v210
transcodes video from rawvideo to 10-bit Uncompressed Y′CBCR 4:2:2. Alter this setting to set your desired codec.
-t 10
specifies recording time of 10 seconds
-
output_file
path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
+
output_file
path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
@@ -2007,13 +2007,13 @@

Sine wave

Generate a test audio file playing a sine wave.

-

ffmpeg -f lavfi -i "sine=frequency=1000:sample_rate=48000:duration=5" -c:a pcm_s16le output_file.wav

+

ffmpeg -f lavfi -i "sine=frequency=1000:sample_rate=48000:duration=5" -c:a pcm_s16le output_file.wav

ffmpeg
starts the command
-f lavfi
tells FFmpeg to use the Libavfilter input virtual device
-i "sine=frequency=1000:sample_rate=48000:duration=5"
Sets the signal to 1000 Hz, sampling at 48 kHz, and for 5 seconds
-c:a pcm_s16le
encodes the audio codec in pcm_s16le (the default encoding for wav files). pcm represents pulse-code modulation format (raw bytes), 16 means 16 bits per sample, and le means "little endian"
-
output_file.wav
path, name and extension of the output file
+
output_file.wav
path, name and extension of the output file
@@ -2025,7 +2025,7 @@

SMPTE bars + Sine wave audio

Generate a SMPTE bars test video + a 1kHz sine wave as audio testsignal.

-

ffmpeg -f lavfi -i smptebars=size=720x576:rate=25 -f lavfi -i "sine=frequency=1000:sample_rate=48000" -c:a pcm_s16le -t 10 -c:v ffv1 output_file

+

ffmpeg -f lavfi -i smptebars=size=720x576:rate=25 -f lavfi -i "sine=frequency=1000:sample_rate=48000" -c:a pcm_s16le -t 10 -c:v ffv1 output_file

ffmpeg
starts the command
-f lavfi
tells FFmpeg to use the libavfilter input virtual device
@@ -2035,7 +2035,7 @@
-c:a pcm_s16le
encodes the audio codec in pcm_s16le (the default encoding for wav files). pcm represents pulse-code modulation format (raw bytes), 16 means 16 bits per sample, and le means "little endian"
-t 10
specifies recording time of 10 seconds
-c:v ffv1
Encodes to FFV1. Alter this setting to set your desired codec.
-
output_file
path, name and extension of the output file
+
output_file
path, name and extension of the output file
@@ -2047,13 +2047,13 @@

Makes a broken test file

Modifies an existing, functioning file and intentionally breaks it for testing purposes.

-

ffmpeg -i input_file -bsf noise=1 -c copy output_file

+

ffmpeg -i input_file -bsf noise=1 -c copy output_file

ffmpeg
starts the command
-
-i input_file
takes in a normal file
+
-i input_file
takes in a normal file
-bsf noise=1
sets bitstream filters for all to 'noise'. Filters can be set on specific filters using syntax such as -bsf:v for video, -bsf:a for audio, etc. The noise filter intentionally damages the contents of packets without damaging the container. This sets the noise level to 1 but it could be left blank or any number above 0.
-c copy
use stream copy mode to re-mux instead of re-encode
-
output_file
path, name and extension of the output file
+
output_file
path, name and extension of the output file
@@ -2079,7 +2079,7 @@
GIF of above command

To save a portion of the stream instead of playing it back infinitely, use the following command:

-

ffmpeg -f lavfi -i life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800 -t 5 output_file

+

ffmpeg -f lavfi -i life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800 -t 5 output_file

@@ -2097,7 +2097,7 @@

ffplay input_file -vf "ocr,drawtext=fontfile=/Library/Fonts/Andale Mono.ttf:text=%{metadata\\\:lavfi.ocr.text}:fontcolor=white"

ffplay
starts the command
-
input_file
path, name and extension of the input file
+
input_file
path, name and extension of the input file
-vf
creates a filtergraph to use for the streams
"
quotation mark to start filtergraph
ocr,
tells ffplay to use ocr as source and the comma signifies that the script is ready for filter assertion
@@ -2118,13 +2118,13 @@

Exports OCR data to screen

Note: FFmpeg must be compiled with the tesseract library for this script to work (--with-tesseract if using the brew install ffmpeg method)

-

ffprobe -show_entries frame_tags=lavfi.ocr.text -f lavfi -i "movie=input_file,ocr"

+

ffprobe -show_entries frame_tags=lavfi.ocr.text -f lavfi -i "movie=input_file,ocr"

ffprobe
starts the command
-show_entries
sets a list of entries to show
-
frame_tags=lavfi.ocr.text
shows the lavfi.ocr.text tag in the frame section of the video
+
frame_tags=lavfi.ocr.text
shows the lavfi.ocr.text tag in the frame section of the video
-f lavfi
tells ffprobe to use the Libavfilter input virtual device
-
-i "movie=input_file,ocr"
declares 'movie' as input_file and passes in the 'ocr' command
+
-i "movie=input_file,ocr"
declares 'movie' as input_file and passes in the 'ocr' command
@@ -2139,10 +2139,10 @@

Compare two video files for content similarity using perceptual hashing

-

ffmpeg -i input_one -i input_two -filter_complex signature=detectmode=full:nb_inputs=2 -f null -

+

ffmpeg -i input_one -i input_two -filter_complex signature=detectmode=full:nb_inputs=2 -f null -

ffmpeg
starts the command
-
-i input_one -i input_two
assigns the input files
+
-i input_one -i input_two
assigns the input files
-filter_complex
enables using more than one input file to the filter
signature=detectmode=full
Applies the signature filter to the inputs in 'full' mode. The other option is 'fast'.
nb_inputs=2
tells the filter to expect two input files
@@ -2157,9 +2157,9 @@

Generate a perceptual hash for an input video file

-

ffmpeg -i input -vf signature=format=xml:filename="output.xml" -an -f null -

+

ffmpeg -i input -vf signature=format=xml:filename="output.xml" -an -f null -

-
ffmpeg -i input
starts the command using your input file
+
ffmpeg -i input
starts the command using your input file
-vf signature=format=xml
applies the signature filter to the input file and sets the output format for the fingerprint to xml
filename="output.xml"
sets the output for the signature filter
-an
tells FFmpeg to ignore the audio stream of the input file
@@ -2179,16 +2179,16 @@

Play an image sequence

Play an image sequence directly as moving images, without having to create a video first.

-

ffplay -framerate 5 input_file_%06d.ext

+

ffplay -framerate 5 input_file_%06d.ext

ffplay
starts the command
-framerate 5
plays image sequence at rate of 5 images per second
- Note: this low framerate will produce a slideshow effect.
-
-i input_file
path, name and extension of the input file
+ Note: this low framerate will produce a slideshow effect.
+
-i input_file
path, name and extension of the input file
This must match the naming convention used! The regex %06d matches six-digit-long numbers, possibly with leading zeroes. This allows the full sequence to be read in ascending order, one image after the other.
The extension for TIFF files is .tif or maybe .tiff; the extension for DPX files is .dpx (or even .cin for old files). Screenshots are often in .png format.
-

Notes:

+

Notes:

If -framerate is omitted, the playback speed depends on the images’ file sizes and on the computer’s processing power. It may be rather slow for large image files.

You can navigate durationally by clicking within the playback window. Clicking towards the left-hand side of the playback window takes you towards the beginning of the playback sequence; clicking towards the right takes you towards the end of the sequence.

@@ -2200,15 +2200,15 @@

Split audio and video tracks

-

ffmpeg -i input_file -map 0:v:0 video_output_file -map 0:a:0 audio_output_file

+

ffmpeg -i input_file -map 0:v:0 video_output_file -map 0:a:0 audio_output_file

This command splits the original input file into a video and audio stream. The -map command identifies which streams are mapped to which file. To ensure that you’re mapping the right streams to the right file, run ffprobe before writing the script to identify which streams are desired.

ffmpeg
starts the command
-
-i input_file
path, name and extension of the input file
+
-i input_file
path, name and extension of the input file
-map 0:v:0
grabs the first video stream and maps it into:
-
video_output_file
path, name and extension of the video output file
+
video_output_file
path, name and extension of the video output file
-map 0:a:0
grabs the first audio stream and maps it into:
-
audio_output_file
path, name and extension of the audio output file
+
audio_output_file
path, name and extension of the audio output file
@@ -2219,21 +2219,21 @@

Merge audio and video tracks

-

ffmpeg -i video_file -i audio_file -map 0:v -map 1:a -c copy output_file

+

ffmpeg -i video_file -i audio_file -map 0:v -map 1:a -c copy output_file

This command takes a video file and an audio file as inputs, and creates an output file that combines the video stream in the first file with the audio stream in the second file.

ffmpeg
starts the command
-
-i video_file
path, name and extension of the first input file (the video file)
-
-i audio_file
path, name and extension of the second input file (the audio file)
-
-map 0:v
selects the video streams from the first input file
-
-map 1:a
selects the audio streams from the second input file
+
-i video_file
path, name and extension of the first input file (the video file)
+
-i audio_file
path, name and extension of the second input file (the audio file)
+
-map 0:v
selects the video streams from the first input file
+
-map 1:a
selects the audio streams from the second input file
-c copy
copies streams without re-encoding
-
output_file
path, name and extension of the output file
+
output_file
path, name and extension of the output file
-

Note: in the example above, the video input file is given prior to the audio input file. However, input files can be added any order, as long as they are indexed correctly when stream mapping with -map. See the entry on stream mapping.

+

Note: in the example above, the video input file is given prior to the audio input file. However, input files can be added any order, as long as they are indexed correctly when stream mapping with -map. See the entry on stream mapping.

Variation:

Include the audio tracks from both input files with the following command:

-

ffmpeg -i video_file -i audio_file -map 0:v -map 0:a -map 1:a -c copy output_file

+

ffmpeg -i video_file -i audio_file -map 0:v -map 0:a -map 1:a -c copy output_file

@@ -2244,14 +2244,14 @@

Create ISO files for DVD access

Create an ISO file that can be used to burn a DVD. Please note, you will have to install dvdauthor. To install dvd author using Homebrew run: brew install dvdauthor

-

ffmpeg -i input_file -aspect 4:3 -target ntsc-dvd output_file.mpg

+

ffmpeg -i input_file -aspect 4:3 -target ntsc-dvd output_file.mpg

This command will take any file and create an MPEG file that dvdauthor can use to create an ISO.

ffmpeg
starts the command
-
-i input_file
path, name and extension of the input file
+
-i input_file
path, name and extension of the input file
-aspect 4:3
declares the aspect ratio of the resulting video file. You can also use 16:9.
-target ntsc-dvd
specifies the region for your DVD. This could be also pal-dvd.
-
output_file.mpg
path and name of the output file. The extension must be .mpg
+
output_file.mpg
path and name of the output file. The extension must be .mpg
@@ -2262,12 +2262,12 @@

Exports CSV for scene detection using YDIF

-

ffprobe -f lavfi -i movie=input_file,signalstats -show_entries frame=pkt_pts_time:frame_tags=lavfi.signalstats.YDIF -of csv

+

ffprobe -f lavfi -i movie=input_file,signalstats -show_entries frame=pkt_pts_time:frame_tags=lavfi.signalstats.YDIF -of csv

This ffprobe command prints a CSV correlating timestamps and their YDIF values, useful for determining cuts.

ffprobe
starts the command
-f lavfi
uses the Libavfilter input virtual device as chosen format
-
-i movie=input file
path, name and extension of the input video file
+
-i movie=input file
path, name and extension of the input video file
,
comma signifies closing of video source assertion and ready for filter assertion
signalstats
tells ffprobe to use the signalstats command
-show_entries
sets list of entries to show per column, determined on the next line
@@ -2283,11 +2283,11 @@

Cover head switching noise

-

ffmpeg -i input_file -filter:v drawbox=w=iw:h=7:y=ih-h:t=max output_file

+

ffmpeg -i input_file -filter:v drawbox=w=iw:h=7:y=ih-h:t=max output_file

This command will draw a black box over a small area of the bottom of the frame, which can be used to cover up head switching noise.

ffmpeg
starts the command
-
-i input_file
path, name and extension of the input file
+
-i input_file
path, name and extension of the input file
-filter:v drawbox=
This calls the drawtext filter with the following options:
@@ -2297,7 +2297,7 @@
t=max
T represents the thickness of the drawn box. Default is 3.
-
output_file
path and name of the output file
+
output_file
path and name of the output file
@@ -2308,7 +2308,7 @@

Record and live-stream simultaneously

-

ffmpeg -re -i ${INPUTFILE} -map 0 -flags +global_header -vf scale="1280:-1,format=yuv420p" -pix_fmt yuv420p -level 3.1 -vsync passthrough -crf 26 -g 50 -bufsize 3500k -maxrate 1800k -c:v libx264 -c:a aac -b:a 128000 -r:a 44100 -ac 2 -t ${STREAMDURATION} -f tee "[movflags=+faststart]${TARGETFILE}|[f=flv]${STREAMTARGET}"

+

ffmpeg -re -i ${INPUTFILE} -map 0 -flags +global_header -vf scale="1280:-1,format=yuv420p" -pix_fmt yuv420p -level 3.1 -vsync passthrough -crf 26 -g 50 -bufsize 3500k -maxrate 1800k -c:v libx264 -c:a aac -b:a 128000 -r:a 44100 -ac 2 -t ${STREAMDURATION} -f tee "[movflags=+faststart]${TARGETFILE}|[f=flv]${STREAMTARGET}"

I use this script to stream to a RTMP target and record the stream locally as .mp4 with only one ffmpeg-instance.

As input, I use bmdcapture which is piped to ffmpeg. But it can also be used with a static videofile as input.

The input will be scaled to 1280px width, maintaining height. Also the stream will stop after a given time (see -t option.)

@@ -2349,7 +2349,7 @@

View information about a specific decoder, encoder, demuxer, muxer, or filter

-

ffmpeg -h type=name

+

ffmpeg -h type=name

ffmpeg
starts the command
-h
Call the help option
@@ -2397,12 +2397,12 @@

Compare two images

-

compare -metric ae image1.ext image2.ext null:

+

compare -metric ae image1.ext image2.ext null:

Compares two images to each other.

compare
starts the command
-metric ae
applies the absolute error count metric, returning the number of different pixels. Other parameters are available for image comparison.
-
image1.ext image2.ext
takes two images as input
+
image1.ext image2.ext
takes two images as input
null:
throws away the comparison image that would be generated
@@ -2423,7 +2423,7 @@
-quality 75
sets quality to 75 (out of 100), adding light compression to smaller files
-path thumbs
specifies where to save the thumbnails -- this goes to a folder within the active folder called "thumbs".
Note: You will have to make this folder if it doesn't already exist.
-
*.jpg
The astericks acts as a "wildcard" to be applied to every file in the directory.
+
*.jpg
The astericks acts as a "wildcard" to be applied to every file in the directory.
@@ -2434,13 +2434,13 @@

Create grid of images

-

montage @list.txt -tile 6x12 -geometry +0+0 output_grid.jpg

+

montage @list.txt -tile 6x12 -geometry +0+0 output_grid.jpg

montage
starts the command
@list.txt
path and name of a text file containing a list of filenames, one per each line
-tile 6x12
specifies the dimensions of the proposed grid (6 images wide, 12 images long)
-geometry +0+0
specifies to include no spacing around any of the tiles; they will be flush against each other
-
output_grid.jpg
path and name of the output file
+
output_grid.jpg
path and name of the output file
@@ -2451,12 +2451,12 @@

Get file signature data

-

convert -verbose input_file.ext | grep -i signature

+

convert -verbose input_file.ext | grep -i signature

Gets signature data from an image file, which is a hash that can be used to uniquely identify the image.

convert
starts the command
-verbose
sets verbose flag for collecting the most data
-
input_file.ext
path and name of image file
+
input_file.ext
path and name of image file
|
pipe the data into something else
grep
starts the grep command
-i signature
ignore case and search for the phrase "signature"
@@ -2487,13 +2487,13 @@

Resize to width

-

convert input_file.ext -resize 750 output_file.ext

+

convert input_file.ext -resize 750 output_file.ext

This script will also convert the file format, if the output has a different file extension than the input.

convert
starts the command
-
-i input_file.ext
path and name of the input file
+
-i input_file.ext
path and name of the input file
-resize 750
resizes the image to 750 pixels wide, retaining aspect ratio
-
output_file.ext
path and name of the output file
+
output_file.ext
path and name of the output file
@@ -2508,13 +2508,13 @@
Change the above data-target field, the hover-over description, the button text, and the below div ID

*****Longer title*****

-

ffmpeg -i input_file *****code goes here***** output_file

+

ffmpeg -i input_file *****code goes here***** output_file

This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info! This is all about info!

ffmpeg
starts the command
-
-i input file
path, name and extension of the input file
+
-i input file
path, name and extension of the input file
*****parameter*****
*****comments*****
-
output file
path, name and extension of the output file
+
output file
path, name and extension of the output file
-->