From 3a58b6ae828565bb01d7826ed4033381db3e9062 Mon Sep 17 00:00:00 2001 From: kfrn Date: Tue, 3 Oct 2017 01:10:28 +1300 Subject: [PATCH] Style: add spaces in long filtergraphs; only use italics for inputs In filtergraphs that are reasonably long or unwieldy, I've added a space after each comma for readability. In ffmpeg commands, italics should denote user inputs/outputs. Where italics had been used for 'regular' variables (i.e., not an input/output file or a variable for which the user must explicitly choose the value), I've deleted them. These are stylistic changes that, I think, make things more readable and/or more uniform. --- index.html | 90 +++++++++++++++++++++++++++--------------------------- 1 file changed, 45 insertions(+), 45 deletions(-) diff --git a/index.html b/index.html index 4d0244f..1798154 100644 --- a/index.html +++ b/index.html @@ -156,23 +156,23 @@

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
-
-c:v libx264
transcodes video to H.264
-
-pix_fmt yuv420p
sets pixel format to yuv420p for greater compatibility with media players
+
-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

@@ -188,7 +188,7 @@

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 md5_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.
@@ -325,11 +325,11 @@
ffmpeg
starts the command
-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.
+
-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

A couple notes

@@ -517,13 +517,13 @@
-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’)
+
"
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.
-
,
separates filters
-
format=yuv420p
chroma subsampling set to 4:2:0
+
,
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
+
"
end of filtergraph
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).
@@ -676,7 +676,7 @@

Plays a graphical output showing decibel levels of an input file

-

ffplay -f lavfi "amovie='input.mp3',astats=metadata=1:reset=1,adrawgraph=lavfi.astats.Overall.Peak_level:max=0:min=-30.0:size=700x256:bg=Black[out]"

+

ffplay -f lavfi "amovie='input.mp3', astats=metadata=1:reset=1, adrawgraph=lavfi.astats.Overall.Peak_level:max=0:min=-30.0:size=700x256:bg=Black[out]"

ffplay
starts the command
-f lavfi
tells ffmpeg to use the Libavfilter input virtual device
@@ -710,7 +710,7 @@

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 ffmpeg to use the Libavfilter input virtual device
@@ -792,7 +792,7 @@

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
@@ -860,23 +860,23 @@

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 -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.
+
-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.
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
number of times to loop the GIF. A value of -1 will disable looping. Omitting -loop will use the default which will loop infinitely
+
-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:

-
-filter_complex "[0:v]fps=10,scale=500:-1:flags=lanczos[v],[v][1:v]paletteuse"
[0:v]fps=10,scale=500:-1:flags=lanczos[v]: applies the fps and scale filters described above to the first input file (the video).
+
-filter_complex "[0:v]fps=10, scale=500:-1:flags=lanczos[v], [v][1:v]paletteuse"
[0:v]fps=10,scale=500:-1:flags=lanczos[v]: applies the fps and scale filters described above to the first input file (the video).
[v][1:v]paletteuse": applies the paletteuse filter, setting the second input file (the palette) as the reference file.

Simpler GIF creation

@@ -975,7 +975,7 @@
-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. + 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

@@ -1047,13 +1047,13 @@

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
-
-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.
+
-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

@@ -1184,7 +1184,7 @@

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.

@@ -1364,7 +1364,7 @@ foreach ($file in $inputfiles) {

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:

@@ -1466,13 +1466,13 @@ foreach ($file in $inputfiles) {

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 ffmpeg 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
@@ -1496,13 +1496,13 @@ foreach ($file in $inputfiles) {

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 ffmpeg 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
@@ -1565,7 +1565,7 @@ foreach ($file in $inputfiles) {
ffmpeg
starts the command
-f lavfi
tells ffmpeg to use the Libavfilter input virtual device
-i mandelbrot=size=1280x720:rate=25
asks for the mandelbrot test filter as input. Adjusting the size and rate options allows you to choose a specific frame size and framerate.
-
-c:v libx264
transcodes video from rawvideo to H.264. Set -pix_fmt to yuv420p for greater H.264 compatibility with media players.
+
-c:v libx264
transcodes video from rawvideo to H.264. Set -pix_fmt to yuv420p for greater H.264 compatibility with media players.
-t 10
specifies recording time of 10 seconds
output_file
path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
@@ -1588,7 +1588,7 @@ foreach ($file in $inputfiles) {
ffmpeg
starts the command
-f lavfi
tells ffmpeg to use the Libavfilter input virtual device
-i smptebars=size=720x576:rate=25
asks for the smptebars test filter as input. Adjusting the size and rate options allows you to choose a specific frame size and framerate.
-
-c:v prores
transcodes video from rawvideo to Apple ProRes 4:2:2.
+
-c:v prores
transcodes video from rawvideo to Apple ProRes 4:2:2.
-t 10
specifies recording time of 10 seconds
output_file
path, name and extension of the output file. Try different file extensions such as mov or avi.
@@ -1612,7 +1612,7 @@ foreach ($file in $inputfiles) {
-f lavfi
tells ffmpeg to use the libavfilter input virtual device
-i testsrc=size=720x576:rate=25
asks for the testsrc filter pattern as input. Adjusting the size and rate options allows you to choose a specific frame size and framerate.
The different test patterns that can be generated are listed here.
-
-c:v v210
transcodes video from rawvideo to 10-bit Uncompressed Y′CBCR 4:2:2. Alter this setting to set your desired codec.
+
-c:v v210
transcodes video from rawvideo to 10-bit Uncompressed Y′CBCR 4:2:2. Alter this setting to set your desired codec.
-t 10
specifies recording time of 10 seconds
output_file
path, name and extension of the output file. Try different file extensions such as mkv, mov, mp4, or avi.
@@ -1656,7 +1656,7 @@ foreach ($file in $inputfiles) {
ffplay
starts the command
-f lavfi
tells ffmpeg to use the Libavfilter input virtual device
-
-i smptebars=size=640x480
asks for the smptebars filter pattern as input. and sets the VGA (SD) resolution. This generates a colour bars pattern, based on the SMPTE Engineering Guideline EG 1–1990.
+
-i smptebars=size=640x480
asks for the smptebars filter pattern as input and sets the VGA (SD) resolution. This generates a colour bars pattern, based on the SMPTE Engineering Guideline EG 1–1990.

@@ -1728,7 +1728,7 @@ foreach ($file in $inputfiles) {
-i "sine=frequency=1000:sample_rate=48000"
Sets the signal to 1000 Hz, sampling at 48 kHz.
-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.
+
-c:v ffv1
Encodes to FFV1. Alter this setting to set your desired codec.
output_file.wav
path, name and extension of the output file

@@ -1788,7 +1788,7 @@ 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.
+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
@@ -1849,12 +1849,12 @@ e.g.: ffmpeg -f concat -safe 0 -i mylist.txt -c copy output_file
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.
+ 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.

@@ -2118,8 +2118,8 @@ e.g.: ffmpeg -f concat -safe 0 -i mylist.txt -c copy output_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
-
video_output_file
path, name and extension of the video output file
+
-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

@@ -2139,7 +2139,7 @@ e.g.: ffmpeg -f concat -safe 0 -i mylist.txt -c copy output_file
ffmpeg
starts the command
-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.
+
-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