ffmpeg Cheat Sheet
The following will create a 640x480 sized output video by copying a corresponding window at offset
y=25px from the input video
ffmpeg -i <input> -filter:v "crop=640:480:100:25" <output>
ffmpeg -i <input> -vf scale=640:480 <output>
ffmpeg -i <input> -ss 00:01:45 -t 00:02:35 -vcodec copy -acodec copy <output>ffmpeg -ss 00:00:30 -i orginalfile.mpg -t 00:00:05 -vcodec copy -acodec copy newfile.mpg
Do not recode for rotation but simple add a video metadate field for the rotation angle
ffmpeg -i <input> -c copy -metadata:s:v:0 rotate=90 <output>
For H265 2-pass encoding you need to combine 2 ffmpeg calls. Example from ffmpeg:
ffmpeg -y -i input -c:v libx265 -b:v 2600k -x265-params pass=1 -an -f mp4 /dev/null && \ ffmpeg -i input -c:v libx265 -b:v 2600k -x265-params pass=2 -c:a aac -b:a 128k output.mp4
Combine “-vn” (for no video) with “-acodec copy”. Note that the output file extension must match the audio codec in the input file for “-acodec copy” to work.
ffmpeg -i file.mp4 -vn -acodec copy output.aac
To create a single thumb at 10s
ffmpeg -ss 10 -i <input file> -vframes 1 -vcodec png -an thumb.png
To create thumbnails every n seconds use “-vf fps=1/n” for example
ffmpeg -i <input file> -vf fps=1/60 thumbnails/thumb%03d.png
ffmpeg -i file.mp3 -f ffmetadata metadata.txt
ffmpeg -i file.mp3 -acodec copy -metadata title="<title>" -metadata artist="<artist>" -metadata album="<album>" out.mp3
ffmpeg -i file.aac -acodec mp3 -ar 44100 -ab 128000 output.mp3
Change container from MKV to MP4
ffmpeg -i file.mkv -acodec copy -vcodec copy file.mp4
If you have multiple numbered images image1.jpg, image2.jpg… create a video from them like this
ffmpeg -f image2 -i image%d.jpg video.mp4
ffmpeg -i video.mp4 image%d.jpg
Update: The workaround for the problem doesn’t work for ffmpeg versions more recent than 20.06.2011 as libfaad support was dropped in favour of the now stable native ffmpeg AAC encoder! If you still have a separate compilation of libfaad you can workaround using the “faad” encoder tool as described in the next section. If you are using recent ffmpeg versions to decode a .MOV file you might get the following error:
Stream #0.0(eng): Audio: aac, 48000 Hz, 2 channels, s16 Stream #0.1(eng): Video: h264, yuv420p, 1280x530, PAR 1:1 DAR 128:53, 25 tbr, 25 tbn, 50 tbc Output #0, flv, to 'test.flv': Stream #0.0(eng): Video: flv (hq), yuv420p, 400x164 [PAR 101:102 DAR 050:2091], q=2-31, 300 kb/s, 1k tbn, 25 tbc Stream #0.1(eng): Audio: libmp3lame, 22050 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding [aac @ 0x80727a0]channel element 1.0 is not allocated Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 [...]
The message “Error while decoding stream #0.0” is repeated continuously. The resulting video is either unplayable or has no sound. Still the input video is playable in all standard players (VLC, in Windows…). The reason for the problem as I understood it is that the ffmpeg-builtin AAC codec cannot handle an audio stream stream with index “1.0”. This is documented in various bugs (see ffmpeg issues #800, #871, #999, #1733…). It doesn’t look like this will be handled by ffmpeg very soon. In fact it could well be that they’ll handle it as an invalid input file. Solution: Upgrade to latest ffmpeg and faad library version and add “ -acodec libfaad “ in front of the “-i” switch. This uses the libfaad AAC decoder, which is said to be a bit slower than the ffmpeg-builtin, but which decodes the AAC without complaining. For example:
ffmpeg -acodec libfaad -i input.mov -b 300kbit/s -ar 22050 -o test.flv
The “-acodec” preceding the “-i” option only influences the input audio decoding, not the audio encoding.
When you try to encode with ffmpeg and you end up with such an error
Resampling with input channels greater than 2 unsupported. Can not resample 6 channels @ 48000 Hz to 6 channels @ 48000
you are probably trying to encode from AAC with 5.1 audio to less than 6 channels or different audio sampling rate. There are three solutions:
- As a solution either do not reduce the audio channels and change the audio sampling rate or do convert the audio with faad first.
- Apply one of the available ffmpeg patches to fix the AAC 6 channel issue…
- Split video and audio and convert audio separately.
The third solution can be done as following:
Extract audio with ffmpeg:
ffmpeg -y -i source.avi -acodec copy source.6.aac
Convert audio with faad:
faad -d -o source.2.pcm source.6.aac
Merge video and audio again with ffmpeg:
ffmpeg -y -i source.avi -i source.2.pcm -map 0:0 -map 1:0 -vcodec copy -acodec copy output.avi
Update: As hinted by a fellow commenter the big disadvantage is the quality loss as faad can only convert into PCM 16bit.
This can be done using the “-async” switch of ffmpeg which according to the documentation “Stretches/squeezes” the audio stream to match the timestamps. The parameter takes a numeric value for the samples per seconds to enforce. Example:
ffmpeg -async 25 -i input.mpg <encoding options> -r 25
Try slowly increasing the -async value until audio and video matches.
Case 1: Audio ahead of video: As a special case the “-async” switch auto-corrects the start of the audio stream when passed as “-async 1”. So try running
ffmpeg -async 1 -i input.mpg <encoding options>
Case 2: Audio behind video: Instead of using “-async” you need to use “-vsync” to drop/duplicate frames in the video stream. There are two methods in the manual page “-vsync 1” and “-vsync 2” and an method auto-detection with “-vsync -1”. But using “-map” it is possible to specify the stream to sync against. Interestingly Google shows people using -aync and -vsync together. So it might be worth experimenting a bit to achieve the intended result :-)
If you have a constantly shifted sound/video track that the previous fix doesn’t work with, but you know the time shift that needs to be corrected, then you can easily fix it with one of the following two commands: Case 1: Audio ahead of video:
ffmpeg -i input.flv -itsoffset 00:00:03.0 -i input.flv -vcodec copy -acodec copy -map 0:1 -map 1:0 output_shift3s.flv
Case 2: Audio behind video:
ffmpeg -i input.flv -itsoffset 00:00:03.0 -i input.flv -vcodec copy -acodec copy -map 1:0 -map 0:1 output_shift3s.flv
The difference is in the mapping parameters which specify which of the two supplied input files to map on which output channel. The “-itsoffset” option indicates an offset (3 seconds in the example) for the following input file. The input file is required to have exactly one video channel at position 0 and one audio channel at position 1. I added “-vcodec copy -acodec copy” to avoid reencoding the video and loose quality. These parameters need to be added after the second input file and before the mapping options. Otherwise one runs into mapping errors. Update: Also check the comment of an anonymous user below mentioning that he needed a different mapping with a more recent version of ffmpeg. The commands above were tested using ffmpeg 0.5/0.6
When preparing videos for Apples HTTP streaming for iPad/iPhone you need to split your video into 10s chunks and provide a play list for Quicktime to process. The problem lies with frame exact splitting of arbitrary video input material. Wether you split the file using ffmpeg or the Apple segmenter tool you often end up with
- asynchronous audio in some or all segments
- missing video frames at the start of each segment
- audio glitches between two segements
- missing audio+video between otherwise audio-synchronous consecutive segments
When using the Apple segmenter the only safe way to split files is to convert into an intermediate format which allows frame-exact splitting. As the segmenter only supports transport stream only MPEG-2 TS and MPEG-4 TS do make sense. To allow frame-exact splitting on problematic input files the easiest way is to blow them up to consist only of I-frames. The parameter for this depends on the output video codec. An ffmpeg command line for MPEG-2 TS can look like this:
ffmpeg -i inputfile -vcodec mpeg2video -pix_fmt yuv422p -qscale 1 -qmin 1 -intra outputfile
The relevant piece is the “-intra” switch. For MPEG-4 TS something like the following should work:
ffmpeg -i inputfile -vcodec libx264 -vpre slow -vpre baseline -acodec libfaac -ab 128k -ar 44100 -intra -b 2000k -minrate 2000k -maxrate 2000k outputfile
Note: It is important to watch the resulting muxing overhead which might lower the effective bitrate a lot! The resulting output files should be safe to be passed to the Apple segmenter.
This is a comparison of the performance of different tools for MP4 tagging. Here you can select between a lot of tools from the net, but only a few of them are command line based and available for Unix. The MP4 test file used is 100MB large.
|AtomicParsely||0.6s||AtomicParsley test.mp4 –artist “Test” –genre “Test” –year “1995”|
|mp4box||0.6s||MP4Box -itags Name=Test:Artist=Me:disk=95/100 test.mp4|
|ffmpeg 0.6||0.8s||ffmpeg -i test.mp4 -metadata title=”Test” -metadata artist=”Test” -metadata date=”1995” -acodec copy -vcodec copy test2.mp4|
If you are unlucky you might see the following ffmpeg error message:
Output #0, image2, to 'output.ppm': Stream #0.0: Video: ppm, rgb24, 144x108, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding av_interleaved_write_frame(): I/O error occurred
Usually that means that input file is truncated and/or corrupted. The above error message was produced with a command like this
ffmpeg -v 0 -y -i 'input.flv' -ss 00:00:01 -vframes 1 -an -sameq -vcodec ppm -s 140x100 'output.ppm'
There are several possible reasons for the error message “av_interleaved_write_frame(): I/O error occurred”.
- You are extracting a thumb and forgot to specify to extract a single frame only (-vframes 1)
- You have a broken input file.
- And finally: The target file cannot be written.
The above was caused by problem three. After a lot of trying I found that the target directory did not exist. Quite confusing.
If compilation fails with an error about the numbers of parameters in common/cpu.c you need to check which glibc version is used. Remove the second parameter to sched_getaffinity() if necessary and recompile.
ffmpeg configure fails with:
ERROR: libx264 not found If you think configure made a mistake, make sure you are using the latest version from SVN. If the latest version fails, report the problem to the [email protected] mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.err" produced by configure as this will help solving the problem.
This can be caused by two effects:
- Unintended library is used for linking. Check wether you have different ones installed. Avoid this and uninstall them if possible. If necessary use LD_LIBRARY_PATH or –extra-ldflags to change the search order.
- Incompatible combination of ffmpeg and libx264. Older libx264 provide a method x264_encoder_open which older ffmpeg versions do check for. More recent libx264 add a version number to the method name. Now when you compile a new libx264 against an older ffmpeg the libx264 detection that relies on the symbol name fails. As a workaround you could hack the configure script to check for “x264_encoder_open_78” instead of “x264_encoder_open” (given that 78 is the libx264 version you use).
ffmpeg compilation fails on AMD64 with:
libavcodec/svq3.c: In function 'svq3_decode_slice_header': libavcodec/svq3.c:721: warning: cast discards qualifiers from pointer target type libavcodec/svq3.c:724: warning: cast discards qualifiers from pointer target type libavcodec/svq3.c: In function 'svq3_decode_init': libavcodec/svq3.c:870: warning: dereferencing type-punned pointer will break strict-aliasing rules /tmp/ccSySbTo.s: Assembler messages: /tmp/ccSySbTo.s:10644: Error: suffix or operands invalid for `add' /tmp/ccSySbTo.s:10656: Error: suffix or operands invalid for `add' /tmp/ccSySbTo.s:12294: Error: suffix or operands invalid for `add' /tmp/ccSySbTo.s:12306: Error: suffix or operands invalid for `add' make: *** [libavcodec/h264.o] Error 1
This post explains that this is related to a glibc issue and how to patch it.
ffmpeg compilation fails with:
libavcodec/libx264.c: In function 'encode_nals': libavcodec/libx264.c:60: warning: implicit declaration of function 'x264_nal_encode' libavcodec/libx264.c: In function 'X264_init': libavcodec/libx264.c:169: error: 'x264_param_t' has no member named 'b_bframe_pyramid' make: *** [libavcodec/libx264.o] Error 1
This means you are using incompatible ffmpeg and libx264 versions. Try to upgrade ffmpeg or to downgrade libx264.
/usr/include/linux/videodev.h:55: error: syntax error before "ulong" /usr/include/linux/videodev.h:71: error: syntax error before '}' token
--- configure.ac.080605 2005-06-08 21:56:04.000000000 +1200 +++ configure.ac 2005-06-08 21:56:42.000000000 +1200 @@ -1226,6 +1226,7 @@ AC_CHECK_HEADERS(linux/videodev.h,,, [#ifdef HAVE_SYS_TIME_H #include <sys/time.h> +#include <sys/types.h> #endif #ifdef HAVE_ASM_TYPES_H #include <asm/types.h>
http://www.winehq.org/pipermail/wine-devel/2005-June/037400.html oder Workaround: –disable-demuxer=v4l –disable-muxer=v4l –disable-demuxer=v4l2 –disable-muxer=v4l2
make: *** No rule to make target `libavdevice/libavdevice.so', needed by `all'. Stop.
Problem: GNU make is too old, you need at least v3.81 http://www.mail-archive.com/[email protected]/msg01284.html
make: *** No rule to make target `install-libs', needed by `install'. Stop.
Problem: GNU make is too old, you need at least v3.81 http://ffmpeg.arrozcru.org/forum/viewtopic.php?f=1&t=833