Published 2022-01-23. Last modified 2023-11-28.
Time to read: 5 minutes.

This page is part of the av_studio collection, categorized under Media, OBS Studio.

When I set up a 4K camera on location and let it roll while I play a set, I get really large video files. I also get large files when using OBS Studio; my usual setup records video from Cam Link 4K, and audio using RME TotalMix. I need to be able to trim video files so the portions before and after the good stuff are discarded.

Lots of online conversations revolve around trimming video files. Dozens of PC and Mac programs exist to do that task, mostly low quality and / or bothersome to use. Other solutions are overkill, for example Adobe Premiere Pro and DaVinci Resolve.

In this post I present a command-line program that can dramatically reduce file size, while preserving quality, and properly trimming to specified time periods.

Although the program is written as a Ruby gem, it installs a command called trim. No programming knowledge is required in order to use it.

What Is FFmpeg?

ffmpeg is a powerful command-line media converter. It is not particularly user-friendly, but it works very well if you put in the time to learn it. The program described in this article provides a easy means of running ffmpeg for those who want to trim media.

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

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

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

 – From FFmpeg documentation

DJV

Need a way to figure out the start and stop times to trim a video? Djv is an excellent video player for this purpose. It is one of the very few video players than can step through a video frame-by-frame, forwards and backwards.

• Mac, Windows, Linux
• Allows frame-by-frame stepping
• Displays the current time reliabily
• F/OSS
• High quality

Installing the Trim Command

You need a working Ruby environment to install this program. I describe how to set that up here.

The trim command is provided by the media_trim Ruby gem. Install it like this:

Shell

$ gem install media_trim

Help Message

The help message is:

Shell

$ trim -h
media_trim - Trim an audio or video file using ffmpeg

- Works with all formats supported by ffmpeg.
- Seeks to the nearest frame positions by re-encoding the media.
- Reduces file size produced by OBS Studio by over 80 percent.
- Can be used as a Ruby gem.
- Installs the 'trim' command.

When run as a command, output files are named by adding a 'trim.' prefix to the media file name, e.g. 'dir/trim.file.ext'.
By default, the trim command does not overwrite pre-existing output files.
When trimming is complete, the trim command displays the trimmed file, unless the -q option is specified

Command-line Usage:
  trim [OPTIONS] dir/file.ext start [[to|for] end]

- The start and end timecodes have the format [HH:[MM:]]SS[.XXX]
  Note that decimal seconds may be specified, and frames may not;
  this is consistent with how ffmpeg parses timecodes.
- end defaults to the end of audio/video file

OPTIONS are:
  -d Enable debug output
  -f Overwrite output file if present
  -v Verbose output
  -V Do not view the trimmed file when complete.

Examples:
  # Crop demo/demo.mp4 from 15.0 seconds to the end of the video, save to demo/trim.demo.mp4:
  trim demo/demo.mp4 15

  # Crop dir/file.mkv from 3 minutes, 25 seconds to 9 minutes, 35 seconds, save to demo/trim.demo.mp4:
  trim demo/demo.mp4 3:25 9:35

  # Same as the previous example, using optional 'to' syntax:
  trim demo/demo.mp4 3:25 to 9:35

  # Save as the previous example, but specify the duration instead of the end time by using the for keyword:
  trim demo/demo.mp4 3:25 for 6:10

Need a way to figure out the start and stop times to trim a video?
DJV is an excellent video viewer https://darbyjohnston.github.io/DJV/
- allows frame-by-frame stepping
- displays the current time reliabily
- F/OSS
- Mac, Windows, Linux
- High quality 

Example Usage

Following is a sample usage of trim, which extracts the portion of VideoFile.mkv from 00:00:25.000 to 00:02:52.000, and writes the extracted portion to a new file called VideoFile.trim.mkv:

Shell

$ trim VideoFile.mkv 25 2:52
Trimming '/mnt/f/work/VideoFile.mp4' from 00:25 to 2:52 (duration 2:27)
Trim took 0:03 

Now we have these files:

Shell

$ ls -AlF
total 1546136
-rw-r--r-- 1 mslinn mslinn   30579665 Jan 23 13:03 'trim.VideoFile.mkv'
-rwxrwxrwx 1 mslinn mslinn  180988555 Jan 22 18:31 'VideoFile.mkv' 

Notes to Programmers

For any programmers who might read this:

• Some of the ffmpeg options this script uses are not available in older versions.
• The most important thing to know about options that might be passed to ffmpeg is that arbitrary start and end times can be specified accurately when trimming if the video stream is re-encoded. To force a video re-encoding, simply do not specify the -vcodec copy option.
• We discard extra (non-essential) streams from video files. See Selecting streams with the -map option.

Sony A7iii Media Files

This section provides background for how the media_trim program operates. Most people do not need to read it.

I wanted to trim a video file created by my Sony A7iii camera, so only the portion from 0:51 through 2:45 was extracted.

FFprobe

First let’s use ffprobe to examine the streams within the media file.

FFprobe gathers information from multimedia streams and prints it in human- and machine-readable fashion.

For example, it can be used to check the format of the container used by a multimedia stream and the format and type of each media stream contained in it.

 – From FFprobe documentation

Shell

$ ffprobe myvideo.mp4
ffprobe version 5.1.2-3ubuntu1 Copyright (c) 2007-2022 the FFmpeg developers
  built with gcc 12 (Ubuntu 12.2.0-14ubuntu2)
  configuration: --prefix=/usr --extra-version=3ubuntu1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libglslang --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librist --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --disable-sndio --enable-libjxl --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-libplacebo --enable-librav1e --enable-shared
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x560a973df5c0] st: 0 edit list: 1 Missing key frame while searching for timestamp: 1001
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x560a973df5c0] st: 0 edit list 1 Cannot find an index entry before timestamp: 1001.
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'myvideo.mp4':
  Metadata:
    major_brand     : XAVC
    minor_version   : 16785407
    compatible_brands: XAVCmp42iso2
    creation_time   : 2023-01-05T00:52:24.000000Z
  Duration: 00:10:58.16, start: 0.000000, bitrate: 51445 kb/s
  Stream #0:0[0x1](und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709/bt709/iec61966-2-4, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 49370 kb/s, 59.94 fps, 59.94 tbr, 60k tbn (default)
    Metadata:
      creation_time   : 2023-01-05T00:52:24.000000Z
      handler_name    : Video Media Handler
      vendor_id       : [0][0][0][0]
      encoder         : AVC Coding
  Stream #0:1[0x2](und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2 channels, s16, 1536 kb/s (default)
    Metadata:
      creation_time   : 2023-01-05T00:52:24.000000Z
      handler_name    : Sound Media Handler
      vendor_id       : [0][0][0][0]
  Stream #0:2[0x3](und): Data: none (rtmd / 0x646D7472), 491 kb/s (default)
    Metadata:
      creation_time   : 2023-01-05T00:52:24.000000Z
      handler_name    : Timed Metadata Media Handler
      timecode        : 03:52:30:26
Unsupported codec with id 0 for input stream 2 

FFprobe shows us that myvideo.mp4 has 3 streams, numbered from origin zero:

0. MPEG-4 video stream – encoded using the H.264 high profile, which is the most commonly used H.264 profile), with 1920x1080 resolution, recorded at 50 Mb/s, and 59.94 fps. This stream was recoded to the same format using FFmpeg’s default stream handling. The recoding process computed keyframes for the new start and points after trimming.
1. CD-quality audio stream – pcm_s16be (16-bit WAV format), at 48 kHz, in stereo, with a bit rate of 1536 kb/s. In contrast, 24 bits is commonly used in 2023 for streaming audio. WAV encoding is uncompressed, and so it is disallowed by strict mp4 compliance.
   
   Because FFmpeg follows string mp4 compliance by default, FFmpeg will not copy audio encoded in WAV format. However, the audio stream can be transcoded to a compressed format, so it can be included in the output file. You could relax FFmpeg’s strict mp4 compliance by providing the -strict experimental option to FFmpeg. Instead, I decided to compress the audio using AAC because this results in a smaller file that sounds just as good.
2. Data stream – normally ignored by ffmpeg when creating output. ffprobe shows an error for this stream (Unsupported codec with id 0 for input stream 2), but we do not need this stream, so the trim script will also ignore it.

Transcoding Audio

Audio is best transcoded to a lossless compressed format that is supported by the MP4 container, like aac or Opus.

The hierarchy of audio encoder quality has been reported as: libopus > libvorbis >= libfdk_aac > libmp3lame >= eac3/ac3 > aac > libtwolame > vorbis > mp2 > wmav2/wmav1.

Ffmpeg’s AAC transcoder gives poor results for bit rates less than 128k. Opus gives a better result, however some hardware devices may not have support. My 3-year-old TCL TV does not support Opus, for example.

FFMpeg supports 5 AAC encoders, including the regular AAC encoder, and the Fraunhofer FDK AAC codec. The latter requires that you compile FFmpeg because of licensing issues. This article will not get into that.

Regular AAC codec

Following is the incantation used by the trim script. It yields regular an mp4 with a video stream and an AAC stereo stream.

Shell

$ ffmpeg -y -ss 51 -to 2:45 -i input.mp4 -acodec aac output.mp4

Both the video and audio streams are recoded. The video stream selected from the input file is the highest resolution video stream in that file. Non-essential streams for regular playback are ignored.

😁

The resulting file is a balance of the highest quality audio and video streams, with the smallest overall file size that can usually be expected to play on most devices.

Fraunhofer FDK AAC codec

You could go one better and build FFmpeg with the Fraunhofer FDK AAC codec, then invoke it with:

Shell

$ ffmpeg -y -ss 51 -to 2:45 -i input.mp4 -c:a libfdk_aac output.mp4

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