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GStreamer is a toolkit for building audio- and video-processing pipelines. A pipeline might stream video from a file to a network, or add an echo to a recording, or (most interesting to us) capture the output of a Video4Linux device. Gstreamer is most often used to power graphical applications such as Totem, but can also be used directly from the command-line. This page will explain why GStreamer is better than the alternatives, and how to build an encoder using its command-line interface.

Before reading this page, please read V4L capturing for background information about the capturing process.

Introduction to GStreamer

No two use cases for encoding are quite alike. What's your preferred workflow? Is your processor fast enough to encode high quality video in real-time? Do you have enough disk space to store the raw video then process it after the fact? Do you want to play your video in DVD players, or is it enough that it works in your version of VLC? How will you work around your system's obscure quirks?

Use GStreamer if you want the best video quality possible with your hardware, and don't mind spending a weekend browsing the Internet for information.

Avoid GStreamer if you just want something quick-and-dirty, or can't stand programs with bad documentation and unhelpful error messages.

Why is GStreamer better at encoding?

GStreamer isn't as easy to use as mplayer, and doesn't have as advanced editing functionality as ffmpeg. But it has superior support for synchronising audio and video in disturbed sources such as VHS tapes. If you specify your input is (say) 25 frames per second video and 48,000Hz audio, most tools will synchronise audio and video simply by writing 1 video frame, 1,920 audio frames, 1 video frame and so on. There are at least three ways this can cause errors:

  • initialisation timing: audio and video desynchronised by a certain amount from the first frame, usually caused by audio and video devices taking different amounts of time to initialise. For example, the first audio frame might be delivered to GStreamer 0.01 seconds after it was requested, but the first video frame might not be delivered until 0.7 seconds after it was requested, causing all video to be 0.6 seconds behind the audio
    • mencoder's -delay option solves this by delaying the audio
  • failure to encode: frames that desynchronise gradually over time, usually caused by audio and video shifting relative each other when frames are dropped. For example if your CPU is not fast enough and sometimes drops a video frame, after 25 dropped frames the video will be one second ahead of the audio
    • mencoder's -harddup option solves this by duplicating other frames to fill in the gaps
  • source frame rate: frames that aren't delivered at the advertised rate, usually caused by inaccurate clocks in the source hardware. For example, a low-cost webcam might deliver 25.01 video frames per second and 47,999Hz, causing your audio and video to drift apart by a second or so per hour
    • video tapes are especially problematic here - if you've ever seen a VCR struggle during those few seconds between two recordings on a tape, you've seen them adjusting the tape speed to accurately track the source. Frame counts can vary enough during these periods to instantly desynchronise audio and video
    • mencoder has no solution for this problem

GStreamer solves these problems by attaching a timestamp to each incoming frame based on the time GStreamer receives the frame. It can then mux the sources back together accurately using these timestamps, either by using a format that supports variable framerates or by duplicating frames to fill in the blanks:

  1. If you choose a container format that supports timestamps (e.g. Matroska), timestamps are automatically written to the file and used to vary the playback speed
  2. If you choose a container format that does not support timestamps (e.g. AVI), you must duplicate other frames to fill in the gaps by adding the videorate and audiorate plugins to the end of the relevant pipelines

To get accurate timestamps, specify the do-timestamp=true option for all your sources. This will ensure accurate timestamps are retrieved from the driver where possible. Sadly, many v4l2 drivers don't support timestamps - GStreamer will add timestamps for these drivers to stop audio and video drifting apart, but you will need to fix the initialisation timing yourself (discussed below).

Once you've encoded your video with GStreamer, you might want to transcode it with ffmpeg's superior editing features.

Getting GStreamer

GStreamer, its most common plugins and tools are available through your distribution's package manager. Most Linux distributions include both the legacy 0.10 and modern 1.0 release series - each has bugs that stop them from working on some hardware, and this page focuses mostly on the legacy 0.10 series because it happened to work with my TV card. Converting the commands below to work with 1.0 is mostly just search-and-replace work (e.g. changing instances of ff to av because of the switch from ffmpeg to libavcodec). See the porting guide for more.

Other plugins are also available, such as entrans (used in some examples below). Google might help you find packages for your distribution, otherwise you'll need to download and compile them yourself.

Using GStreamer with gst-launch

gst-launch is the standard command-line interface to GStreamer. Here's the simplest pipline you can build:

gst-launch-0.10 fakesrc ! fakesink

This connects a single (fake) source to a single (fake) sink using the 0.10 series of GStreamer:

Very simple pipeline

GStreamer can build all kinds of pipelines, but you probably want to build one that looks something like this:

Idealised pipeline example

To get a list of elements that can go in a GStreamer pipeline, do:

gst-inspect-0.10 | less

Pass an element name to gst-inspect-0.10 for detailed information. For example:

gst-inspect-0.10 fakesrc
gst-inspect-0.10 fakesink

If you install Graphviz, you can build graphs like the above yourself:

mkdir gst-visualisations
GST_DEBUG_DUMP_DOT_DIR=gst-visualisations gst-launch-0.10 fakesrc ! fakesink
dot -Tpng gst-visualisations/* > my-pipeline.png

To get graphs of the example pipelines below, prepend GST_DEBUG_DUMP_DOT_DIR=gst-visualisations to the gst-launch command. Run this command to generate a PNG version of GStreamer's most interesting stage:

dot -Tpng gst-visualisations/* > my-pipeline.png

Remember to empty the gst-visualisations directory between runs.

Using GStreamer with entrans

gst-launch is the main command-line interface to GStreamer, available by default. But entrans is a bit smarter:

  • it provides partly-automated composition of GStreamer pipelines
  • it allows you to cut streams, for example to capture for a predefined duration. That ensures headers are written correctly, which is not always the case if you close gst-launch by pressing Ctrl+C. To use this feature one has to insert a dam element after the first queue of each part of the pipeline

Common caputuring issues and their solutions

Reducing Jerkiness

If motion that should appear smooth instead stops and starts, try the following:

Check for muxer issues. Some muxers need big chunks of data, which can cause one stream to pause while it waits for the other to fill up. Change your pipeline to pipe your audio and video directly to their own filesinks - if the separate files don't judder, the muxer is the problem.

  • If the muxer is at fault, add ! queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 immediately before each stream goes to the muxer
    • queues have hard-coded maximum sizes - you can chain queues together if you need more buffering than one buffer can hold

Check your CPU load. When GStreamer uses 100% CPU, it may need to drop frames to keep up.

  • If frames are dropped occasionally when CPU usage spikes to 100%, add a (larger) buffer to help smooth things out.
    • this can be a source's internal buffer (e.g. v4l2src queue-size=16 or alsasrc buffer-time=2000000), or it can be an extra buffering step in your pipeline (! queue max-size-buffers=0 max-size-time=0 max-size-bytes=0)
  • If frames are dropped when other processes have high CPU load, consider using nice to make sure encoding gets CPU priority
  • If frames are dropped regularly, use a different codec, change the parameters, lower the resolution, or otherwise choose a less resource-intensive solution

As a general rule, you should try increasing buffers first - if it doesn't work, it will just increase the pipeline's latency a bit. Be careful with nice, as it can slow down or even halt your computer.

Check for incorrect timestamps. If your video driver works by filling up an internal buffer then passing a cluster of frames without timestamps, GStreamer will think these should all have (nearly) the same timestamp. Make sure you have a videorate element in your pipeline, then add silent=false to it. If it reports many framedrops and framecopies even when the CPU load is low, the driver is probably at fault.

  • videorate on its own will actually make this problem worse by picking one frame and replacing all the others with it. Instead install entrans and add its stamp element between v4l2src and queue (e.g. v4l2src do-timestamp=true ! stamp sync-margin=2 sync-interval=5 ! videorate ! queue)
    • stamp intelligently guesses timestamps if drivers don't support timestamping. Its sync- options drop or copy frames to get a nearly-constant framerate. Using videorate as well does no harm and can solve some remaining problems

Avoiding pitfalls with video noise

If your video contains periods of video noise (snow), you may need to deal with some extra issues:

  • Most devices send an EndOfStream signal if the input signal quality drops too low, causing GStreamer to finish capturing. To prevent the device from sending EOS, set num-buffers=-1 on the v4l2src element.
  • The stamp plugin gets confused by periods of snow, causing it to generate faulty timestamps and framedropping. stamp will recover normal behaviour when the break is over, but will probably leave the buffer full of weirdly-stamped frames. stamp only drops one weirdly-stamped frame each sync-interval, so it can take several minutes until everything works fine again. To solve this problem, set leaky=2 on each queue element to allow dropping old frames
  • Periods of noise (snow, bad signal etc.) are hard to encode. Variable bitrate encoders will often drive up the bitrate during the noise then down afterwards to maintain the average bitrate. To minimise the issues, specify a minimum and maximum bitrate in your encoder
  • Snow at the start of a recording is just plain ugly. To get black input instead from a VCR, use the remote control to change the input source before you start recording

Working around bugs in GStreamer

Sadly, you will probably run into several GStreamer bugs while creating a pipeline. Worse, the debugging information usually isn't enough to diagnose problems. Debugging usually goes something like this:

  1. see that gst-launch is failing to initialise, and gives no useful error message
  2. try similar pipelines - add, change and remove large parts until you find something that works
    • if possible, start with the working version and make one change at a time until you identify a single change that triggers the problem
  3. run the simplest failing pipeline with --gst-debug=3 (or higher if necessary)
  4. find an error message that looks relevant, search the Internet for information about it
  5. try more variations based on what you learnt, until you eventually find something that works

For example, here's the basic outline of how I diagnosed the need to use format=UYVY (minus several wrong turns and silly mistakes):

  1. Made a simple pipeline to encode video in GStreamer 1.0:
    gst-launch-1.0 v4l2src device=$VIDEO_DEVICE ! video/x-raw,interlaced=true,width=720,height=576 ! autovideosink
    • this opened a window, but never played any video frames into it
  2. Tried the same pipeline in GStreamer 0.10:
    gst-launch-0.10 v4l2src device=$VIDEO_DEVICE ! video/x-raw-yuv,interlaced=true,width=720,height=576 ! autovideosink
    • this worked, proving GStreamer is capable of processing video
  3. Reran the failing command with --gst-debug=3:
    gst-launch-1.0 --gst-debug=3 v4l2src device=$VIDEO_DEVICE ! video/x-raw,interlaced=true,width=720,height=576 ! autovideosink
    • this produced some error messages that didn't mean anything to me
  4. Searched on Google for "gst_video_frame_map_id: failed to map video frame plane 1" (with quotes)
  5. Read through that thread
    • Most of it went over my head, but I understood it was a GStreamer bug related to the YUV format
  6. Read the v4l2src description for supported formats:
    gst-inspect-1.0 v4l2src | less
  7. Tried every possible format, made a note of which ones worked:
        echo FORMAT: $FORMAT
        gst-launch-1.0 v4l2src device=$VIDEO_DEVICE ! video/x-raw,format=$FORMAT,interlaced=true,width=720,height=576 ! autovideosink
    • YUY2 and UYVY worked, the others failed to initialise or opened a window but failed to play video
  8. Searched on Google for information about YUY2 and UYVY, as well as YUV9 and YUV12, which seemed to be related
  9. compared the output of the formats that worked to the GStreamer 0.10 baseline:
    gst-launch-1.0  v4l2src device=$VIDEO_DEVICE ! video/x-raw, format=YUY2, interlaced=true, width=720, height=576 ! videoconvert ! x264enc ! matroskamux ! filesink location=1-yuy2.mkv
    gst-launch-1.0  v4l2src device=$VIDEO_DEVICE ! video/x-raw, format=UYVY, interlaced=true, width=720, height=576 ! videoconvert ! x264enc ! matroskamux ! filesink location=1-uyvy.mkv
    gst-launch-0.10 v4l2src device=$VIDEO_DEVICE ! video/x-raw-yuv,          interlaced=true, width=720, height=576                ! x264enc ! matroskamux ! filesink location=10.mkv
    • killed each after 5 seconds then played them side-by-side - they looked similar to my eye, but the 1.0 files played at 720x629 resolution
  10. Re-read the v4l2src description:
    gst-inspect-1.0 v4l2src | less
    • found the pixel-aspect-ratio setting
  11. arbitrarily chose format=UYVY and tried again with a defined pixel aspect ratio:
    gst-launch-1.0  v4l2src device=$VIDEO_DEVICE pixel-aspect-ratio=1 \
        ! video/x-raw, format=UYVY, interlaced=true, width=720, height=576 \
        ! videoconvert \
        ! x264enc \
        ! matroskamux \
        ! filesink location=1-uyvy.mkv
    • it worked!

Sample pipelines

At some point, you will probably need to build your own GStreamer pipeline. This section contains examples to give you the basic idea.

Note: for consistency and ease of copy/pasting, all filenames in this section are of the form test-$( date --iso-8601=seconds ) - your shell should automatically convert this to e.g. test-2010-11-12T13:14:15+1600.avi

Record raw video only

A simple pipeline that initialises one video source, sets the video format, muxes it into a file format, then saves it to a file:

gst-launch-0.10 \
    v4l2src do-timestamp=true device=$VIDEO_DEVICE \
        ! video/x-raw-yuv,width=640,height=480 \
        ! avimux
        ! filesink location=test-$( date --iso-8601=seconds ).avi

tcprobe says this video-only file uses the I420 codec and gives the framerate as correct NTSC:

$ tcprobe -i test-*.avi
[tcprobe] RIFF data, AVI video
[avilib] V: 29.970 fps, codec=I420, frames=315, width=640, height=480
[tcprobe] summary for test-(date).avi, (*) = not default, 0 = not detected
import frame size: -g 640x480 [720x576] (*)
       frame rate: -f 29.970 [25.000] frc=4 (*)
   no audio track: use "null" import module for audio
           length: 315 frames, frame_time=33 msec, duration=0:00:10.510

The files will play in mplayer, using the codec [raw] RAW Uncompressed Video.

Record to ogg theora

Here is a more complex example that initialises two sources - one video and audio:

gst-launch-0.10 \
    v4l2src do-timestamp=true device=$VIDEO_DEVICE \
        ! video/x-raw-yuv,width=640,height=480,framerate=\(fraction\)30000/1001 \
        ! ffmpegcolorspace \
        ! theoraenc \
        ! queue \
        ! mux. \
    alsasrc do-timestamp=true device=$AUDIO_DEVICE \
        ! audio/x-raw-int,channels=2,rate=32000,depth=16 \
        ! audioconvert \
        ! vorbisenc \
        ! mux. \
    oggmux name=mux \
        ! filesink location=test-$( date --iso-8601=seconds ).ogg

Each source is encoded and piped into a muxer that builds an ogg-formatted data stream. The stream is then saved to test-$( date --iso-8601=seconds ).ogg. Note the required workaround to get sound on a saa7134 card, which is set at 32000Hz (cf. bug). However, I was still unable to get sound output, though mplayer claimed there was sound -- the video is good quality:

VIDEO:  [theo]  640x480  24bpp  29.970 fps    0.0 kbps ( 0.0 kbyte/s)
Selected video codec: [theora] vfm: theora (Theora (free, reworked VP3))
AUDIO: 32000 Hz, 2 ch, s16le, 112.0 kbit/10.94% (ratio: 14000->128000)
Selected audio codec: [ffvorbis] afm: ffmpeg (FFmpeg Vorbis decoder)

Record to mpeg4

This is similar to the above, but generates an AVI file with streams encoded using AVI-compatible encoders:

gst-launch-0.10 \
    v4l2src do-timestamp=true device=$VIDEO_DEVICE \
        ! video/x-raw-yuv,width=640,height=480,framerate=\(fraction\)30000/1001 \
        ! ffmpegcolorspace \
        ! ffenc_mpeg4 \
        ! queue \
        ! mux. \
    alsasrc do-timestamp=true device=$AUDIO_DEVICE \
        ! audio/x-raw-int,channels=2,rate=32000,depth=16 \
        ! audioconvert \
        ! lame \
        ! mux. \
    avimux name=mux \
        ! filesink location=test-$( date --iso-8601=seconds ).avi

I get a file out of this that plays in mplayer, with blocky video and no sound. Avidemux cannot open the file.

High quality video

gst-launch-0.10 -q -e \
    v4l2src device=$VIDEO_DEVICE do-timestamp=true norm=PAL-I \
        ! queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! video/x-raw-yuv,interlaced=true,width=720,height=576 \
        ! x264enc interlaced=true pass=quant option-string=quantizer=0 speed-preset=ultrafast byte-stream=true \
        ! progressreport update-freq=1 \
        ! mux. \
    alsasrc device=$AUDIO_DEVICE do-timestamp=true \
        ! queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! audio/x-raw-int,depth=16,rate=32000 \
        ! flacenc \
        ! mux. \
    matroskamux name=mux min-index-interval=1000000000 \
        ! filesink location=test-$( date --iso-8601=seconds ).mkv

This creates a file with 720x576 video and 32kHz audio, using FLAC audio and x264 video in lossless mode, muxed into in a Matroska container. It doesn't need much CPU and should be a faithful representation of the source, but the files can be 30GB per hour and it's not widely supported. Consider transcoding to a smaller/more compatible format after encoding.

Some things to note about this pipeline:

  • quantizer=0 sets the video codec to lossless mode. Anything up to quantizer=18 should not lose information visible to the human eye, and will produce much smaller files
  • speed-preset=ultrafast sets the video codec to prioritise speed at the cost of file size. speed-preset=veryslow is the opposite, and there are various presets in between
  • The Matroska format supports variable framerates, which can be useful for VHS tapes that might not deliver the same number of frames each second
  • min-index-interval=1000000000 improves seek times by telling the Matroska muxer to create one cue data entry per second of playback. Cue data is a few kilobytes per hour, added to the end of the file when encoding completes. If you try to watch your Matroska video while it's being recorded, it will take a long time to skip forward/back because the cue data hasn't been written yet

GStreamer 1.0: record from a bad analog signal to MJPEG video and RAW mono audio

stamp is not available in GStreamer 1.0, cogcolorspace and ffmpegcolorspace have been replaced by videoconvert:

gst-launch-1.0 \
    v4l2src do-timestamp=true device=$VIDEO_DEVICE do-timestamp=true \
        ! 'video/x-raw,format=(string)YV12,width=(int)720,height=(int)576' \
        ! videorate \
        ! 'video/x-raw,format=(string)YV12,framerate=25/1' \
        ! videoconvert \
        ! 'video/x-raw,format=(string)YV12,width=(int)720,height=(int)576' \
        ! jpegenc \
        ! queue \
        ! mux. \
    alsasrc do-timestamp=true device=$AUDIO_DEVICE \
        ! 'audio/x-raw,format=(string)S16LE,rate=(int)48000,channels=(int)2' \
        ! audiorate \
        ! audioresample \
        ! 'audio/x-raw,rate=(int)44100' \
        ! audioconvert \
        ! 'audio/x-raw,channels=(int)1' \
        ! queue \
        ! mux. \
    avimux name=mux ! filesink location=test-$( date --iso-8601=seconds ).avi

As stated above, it is best to use both audiorate and videorate: you problably use the same chip to capture both audio stream and video stream so the audio part is subject to disturbance as well.

View pictures from a webcam

Here are some miscellaneous examples for viewing webcam video:

gst-launch-0.10 \
    v4l2src do-timestamp=true use-fixed-fps=false \
        ! video/x-raw-yuv,format=\(fourcc\)UYVY,width=320,height=240 \
        ! ffmpegcolorspace \
        ! autovideosink
gst-launch-0.10 \
    v4lsrc do-timestamp=true autoprobe-fps=false device=$VIDEO_DEVICE \
        ! "video/x-raw-yuv,format=(fourcc)I420,width=160,height=120,framerate=10" \
        ! autovideosink

Entrans: Record to DVD-compliant MPEG2

entrans -s cut-time -c 0-180 -v -x '.*caps' --dam -- --raw \
    v4l2src queue-size=16 do-timestamp=true device=$VIDEO_DEVICE norm=PAL-BG num-buffers=-1 \
        ! stamp silent=false progress=0 sync-margin=2 sync-interval=5 \
        ! queue silent=false leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! dam \
        ! cogcolorspace \
        ! videorate silent=false \
        ! 'video/x-raw-yuv,width=720,height=576,framerate=25/1,interlaced=true,aspect-ratio=4/3' \
        ! queue silent=false leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! ffenc_mpeg2video rc-buffer-size=1500000 rc-max-rate=7000000 rc-min-rate=3500000 bitrate=4000000 max-key-interval=15 pass=pass1 \
        ! queue silent=false leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! mux. \
    pulsesrc buffer-time=2000000 do-timestamp=true \
        ! queue silent=false leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! dam \
        ! audioconvert \
        ! audiorate silent=false \
        ! audio/x-raw-int,rate=48000,channels=2,depth=16 \
        ! queue silent=false max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! ffenc_mp2 bitrate=192000 \
        ! queue silent=false leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
        ! mux. \
    ffmux_mpeg name=mux \
        ! filesink location=test-$( date --iso-8601=seconds ).mpg

This captures 3 minutes (180 seconds, see first line of the command) to test-$( date --iso-8601=seconds ).mpg and even works for bad input signals.

  • I wasn't able to figure out how to produce a mpeg with ac3-sound as neither ffmux_mpeg nor mpegpsmux support ac3 streams at the moment. mplex does but I wasn't able to get it working as one needs very big buffers to prevent the pipeline from stalling and at least my GStreamer build didn't allow for such big buffers.
  • The limited buffer size on my system is again the reason why I had to add a third queue element to the middle of the audio as well as of the video part of the pipeline to prevent jerking.
  • In many HOWTOs you find ffmpegcolorspace instead of cogcolorspace. You can even use this but cogcolorspace is much faster.
  • It seems to be important that the video/x-raw-yuv,width=720,height=576,framerate=25/1,interlaced=true,aspect-ratio=4/3-statement is after videorate as videorate seems to drop the aspect-ratio-metadata otherwise resulting in files with aspect-ratio 1 in theis headers. Those files are probably played back warped and programs like dvdauthor complain.

Ready-made scripts

Although no two use cases are the same, it can be useful to see scripts used by other people. These can fill in blanks and provide inspiration for your own work.

Bash script to record video tapes with GStreamer (work-in-progress)

Note: as of August 2015, this script is still being fine-tuned. Come back in a month or two to see the final version.

This example encapsulates a whole workflow - encoding with GStreamer, transcoding with ffmpeg and opportunities to edit the audio by hand. The default GStreamer command is similar to this, and by default ffmpeg converts it to MPEG4 video and MP3 audio in an AVI container.

The script has been designed so most people should only need to edit the config file, and even includes a more usable version of the commands from getting your device capabilities. In general, you should first run the script with --init to create the config file, then edit that file by hand with help from --caps and --profile, then record with --record and transcode with a generated remaster script.

Search the script for CMD to find the interesting bits. Although the script is quite complex, most of it is just fluff to improve progress information etc.

# Encode a video using either the 0.1 or 1.0 series of GStreamer
# (each has bugs that break encoding on different cards)
# Also uses `v4l2-ctl` (from the v4l-utils package) to set the input source,
# and `ffmpeg` to remaster the file
# Approximate system requirements for maximum quality settings:
# * about 5-10GB disk space for every hour of the initial recording
# * about 4-8GB disk space for every hour of remastered recordings
# * 1.5GHz processor

HELP_MESSAGE="Usage: $0 --init
       $0 --caps
       $0 --profile
       $0 --record <directory>
       $0 --kill <directory> <timeout>
       $0 --remaster <remaster-script>

Record a video into a directory (one directory per video).

    --init     create an initial ~/.v4l-record-scriptrc
               please edit this file before your first recording

    --caps     show audio and video capabilities for your device

    --profile  update ~/.v4l-record-scriptrc with your system's noise profile
               pause a tape or tune to a silent channel for the best profile

    --record   create a faithful recording in the specified directory

    --kill     stop the recording in <directory> after <timeout>
               see \`man sleep\` for details about allowed time formats

    --remaster create remastered recordings based on the initial recording

# For more information, see

# Every system and every use case is slightly different.
# Here are the things you will probably need to change:

# Set these based on your hardware/location:
VIDEO_DEVICE=${VIDEO_DEVICE:-/dev/video0} # `ls /dev/video*` for a list
AUDIO_DEVICE=${AUDIO_DEVICE:-hw:CARD=SAA7134,DEV=0} # `arecord -L` for a list
NORM=${NORM:-PAL} # (search Wikipedia for the exact norm in your country)

VIDEO_INPUT="${VIDEO_INPUT:-1}" # composite input - `v4l2-ctl --device=$VIDEO_DEVICE --list-inputs` for a list

# PAL video is approximately 720x576 resolution.  VHS tapes have about half the horizontal quality, but this post convinced me to encode at 720x576 anyway:
# Run `'"$0"' --caps` to find your supported width, height and bitrate:

# For systems that do not automatically handle audio/video initialisation times:

# These are defined in the script, but you can override them here if you need non-default values:

# set this to 1.0 to use the more recent version of GStreamer:

# Set these to alter the recording quality:

# Set these to control the audio/video pipelines:

# ffmpeg has better remastering tools:
#FFMPEG_DENOISE_OPTS="..." # edit depending on your tape quality

# Reducing noise:

# Once you have set the above, record a silent source (e.g. a paused tape or silent TV channel)
# then call '"$0"' --profile to build the global noise profile


[ -e "$CONFIG_SCRIPT" ] && source "$CONFIG_SCRIPT"
source <( echo "$CONFIGURATION" )

GST_VERSION="${GST_VERSION:-0.10}" # or 1.0

# `gst-inspect` has more information here too:
GST_X264_OPTS="interlaced=true pass=quant option-string=qpmin=0:qpmax=0 speed-preset=ultrafast tune=zerolatency byte-stream=true"
GST_MKV_OPTS="min-index-interval=2000000000" # also known as "cue data", this makes seeking faster

# this doesn't really matter, and isn't required in 1.0:
case "$GST_VERSION" in
        echo "Please specify 'GST_VERSION' of '0.10' or '1.0', not '$GST_VERSION'"
        exit 1

# `gst-inspect-0.10 <element> | less -i` for a list of properties (e.g. `gst-inspect-0.10 v4l2src | less -i`):
GST_QUEUE="${GST_QUEUE:-queue max-size-buffers=0 max-size-time=0 max-size-bytes=0}"
GST_VIDEO_SRC="${GST_VIDEO_SRC:-v4l2src device=$VIDEO_DEVICE do-timestamp=true norm=$NORM ! $GST_QUEUE ! $GST_VIDEO_CAPS}"
GST_AUDIO_SRC="${GST_AUDIO_SRC:-alsasrc device=$AUDIO_DEVICE do-timestamp=true            ! $GST_QUEUE ! $GST_AUDIO_CAPS}"

# `ffmpeg -h full` for more information:
FFMPEG_DENOISE_OPTS="hqdn3d=luma_spatial=6:2:luma_tmp=20" # based on an old VHS tape, with recordings in LP mode
FFMPEG_VIDEO_OPTS="${FFMPEG_VIDEO_OPTS:--flags +ilme+ildct -c:v mpeg4 -q:v 3 -vf il=d,$FFMPEG_DENOISE_OPTS,il=i,crop=(iw-10):(ih-14):3:0,pad=iw:ih:(ow-iw)/2:(oh-ih)/2}"
FFMPEG_AUDIO_OPTS="${FFMPEG_AUDIO_OPTS:--c:a libmp3lame -b:a 256k}" # note: for some reason, ffmpeg desyncs audio and video if "-q:a" is used instead of "-b:a"

# You should only need to edit these if you're making significant changes to the way the script works

pluralise() {
    case "$1" in
        ""|0) return
        1) echo "$1 $2, "
        *) echo "$1 ${2}s, "

gst_progress() {
    START_TIME="$( date +%s )"
    while read HEAD TAIL
        if [ "$HEAD" = "progressreport0" ]
            NOW_TIME="$( date +%s )"
            echo -n $'\r'"$( echo -n "$MESSAGE" | tr -c '' ' ' )"$'\r'
            MESSAGE="$( echo "$TAIL" | {
                progress_message "" "$START_TIME" "$TOTAL" "$PROCESSED"
                echo "$MESSAGE"
            echo "$PROGRESS_NEWLINE$HEAD $TAIL" >&2
            echo "$MESSAGE" >&2
    echo -n $'\r'"$( echo -n "$MESSAGE" | tr -c '' ' ' )"$'\r' >&2

ffmpeg_progress() {


    echo -n $'\r'"$MESSAGE" >&2
    while IFS== read PARAMETER VALUE
        if [ "$PARAMETER" = out_time_ms ]
            echo -n $'\r'"$( echo -n "$MESSAGE" | tr -c '' ' ' )"$'\r' >&2
            if [ -z "$TOTAL_TIME_MS" -o "$TOTAL_TIME_MS" = 0 ]
                case $SPINNER in
                    \-|'') SPINNER=\\ ;;
                    \\   ) SPINNER=\| ;;
                    \|   ) SPINNER=\/ ;;
                    \/   ) SPINNER=\- ;;
                MESSAGE="$1 $SPINNER"
                if [ -n "$VALUE" -a "$VALUE" != 0 ]
                    TIME_REMAINING=$(( ( $(date +%s) - $START_TIME ) * ( $TOTAL_TIME_MS - $VALUE ) / $VALUE ))
                    HOURS_REMAINING=$(( $TIME_REMAINING / 3600 ))
                    MINUTES_REMAINING=$(( ( $TIME_REMAINING - $HOURS_REMAINING*3600 ) / 60 ))
                    HOURS_REMAINING="$(   pluralise   $HOURS_REMAINING hour   )"
                    MINUTES_REMAINING="$( pluralise $MINUTES_REMAINING minute )"
                    SECONDS_REMAINING="$( pluralise $SECONDS_REMAINING second )"
                    MESSAGE_REMAINING="$( echo "$HOURS_REMAINING$MINUTES_REMAINING$SECONDS_REMAINING" | sed -e 's/, $//' -e 's/\(.*\),/\1 and/' )"
                    MESSAGE="$1 $(( 100 * VALUE / TOTAL_TIME_MS ))% ETA: $( date +%X -d "$TIME_REMAINING seconds" ) (about $MESSAGE_REMAINING)"
            echo -n $'\r'"$MESSAGE" >&2
        elif [ "$PARAMETER" = progress -a "$VALUE" = end ]
            echo -n $'\r'"$( echo -n "$MESSAGE" | tr -c '' ' ' )"$'\r' >&2


# convert 00:00:00.000 to a count in milliseconds
parse_time() {
    echo "$(( $(date -d "1970-01-01T${1}Z" +%s )*1000 + $( echo "$1" | sed -e 's/.*\.\([0-9]\)$/\100/' -e 's/.*\.\([0-9][0-9]\)$/\10/' -e 's/.*\.\([0-9][0-9][0-9]\)$/\1/' -e '/^[0-9][0-9][0-9]$/! s/.*/0/' ) ))"

# get the full name of the script's directory
set_directory() {
    if [ -z "$1" ]
        echo "$HELP_MESSAGE"
        exit 1
        DIRECTORY="$( readlink -f "$1" )"
        FILE="$DIRECTORY/$( basename "$DIRECTORY" )"

# actual commands that do something interesting:
CMD_FFMPEG="ffmpeg -loglevel 23 -nostdin"
CMD_SOX="nice -n +20 sox"


case "$1" in

        if [ -e "$CONFIG_SCRIPT" ]
            echo "Please delete $CONFIG_SCRIPT if you want to recreate it"
            echo "$CONFIGURATION" > "$CONFIG_SCRIPT"
            echo "Please edit $CONFIG_SCRIPT to match your system"

        sed -i "$CONFIG_SCRIPT" -e '/^GLOBAL_NOISE_PROFILE=.*/d'
        echo "GLOBAL_NOISE_PROFILE='$( '$CMD_GST' -q alsasrc device="$AUDIO_DEVICE" ! wavenc ! fdsink | sox -t wav - -n trim 0 1 noiseprof | tr '\n' '\t' )'" >> "$CONFIG_SCRIPT"
        echo "Updated $CONFIG_SCRIPT with global noise profile"

            echo 'Audio capabilities:' >&2
            "$CMD_GST" --gst-debug=alsa:5 alsasrc device=$AUDIO_DEVICE ! fakesink 2> >( sed -ne '/returning caps\|src caps/ { s/.*\( returning caps \| src caps \)/\t/ ; s/; /\n\t/g ; p }' | sort >&2 ) | head -1 >/dev/null
            sleep 0.1
            echo 'Video capabilities:' >&2
            "$CMD_GST" --gst-debug=v4l2:5,v4l2src:3 v4l2src device=$VIDEO_DEVICE ! fakesink 2> >( sed -ne '/probed caps:\|src caps/ { s/.*\(probed caps:\|src caps\) /\t/ ; s/; /\n\t/g ; p }' | sort >&2 ) | head -1 >/dev/null
        } 2>&1


        # Build a pipeline with sources being encoded as MPEG4 video and FLAC audio, then being muxed into a Matroska container.
        # FLAC and Matroska are used during encoding to ensure we don't lose much data between passes

        set_directory "$2"
        mkdir -p -- "$DIRECTORY" || exit

        if [ -e '$' ]
            echo "Already recording a video in this directory"
        if [ -e "$FILE.mkv" ]
            echo "Please delete the old $FILE.mkv before making a new recording"
            exit 1

        [ -n "$VIDEO_INPUT" ] && v4l2-ctl --device="$VIDEO_DEVICE" --set-input $VIDEO_INPUT > >( grep -v '^Video input set to' )

        date +"%c: started recording $FILE.mkv"
        # trap keyboard interrupt (control-c)
        kill_gstreamer() { [ -e "/proc/$(< "$" )" ] && kill -s 2 "$(< "$" )" ; }
        sh -c "echo \$\$ > '$' && \
            exec $CMD_GST -q -e \
                $GST_VIDEO_SRC ! x264enc $GST_X264_OPTS ! progressreport update-freq=1 ! mux. \
                $GST_AUDIO_SRC ! flacenc $GST_FLAC_OPTS ! mux. \
                matroskamux name=mux $GST_MKV_OPTS ! filesink location='$FILE.mkv'" \
            2> >( grep -v 'Source ID [0-9]* was not found when attempting to remove it' ) \
            | \
            while read FROM TIME REMAINDER
            do [ "$FROM" = progressreport0 ] && echo -n $'\r'"$( date +"%c: recorded ${TIME:1:8} - press ctrl+c to finish" )" >&2
        echo >&2
        date +"%c: finished recording $FILE.mkv"

        rm -f "$"

        cat <<EOF > "$"
#!$0 --remaster
# The original $( basename $FILE ).mkv accurately represents the source.
# If you would like to get rid of imperfections in the source (e.g.
# splitting it into segments), edit then run this file.

# To add a period of silence at the beginning of the video, watch the .mkv
# file and decide how much silence you want.
# If you want to add a delay, set this variable to the duration in seconds
# (can be fractional):
audio_delay ${AUDIO_DELAY:-0.0}

# This is the original file to be remastered:
original "$( basename $FILE ).mkv"

# You can split a video into one or more files.  To create a remastered
# segment, add a line like this:
# segment "name of output file.avi" "start time" "end time"
# "start time"/"end time" is optional, and specifies the part of the file
# that will be used for the segment

# Here are some examples - remove the leading '#' to make one work:

# remaster the whole file in one go:
# segment "$( basename $FILE ).avi"

# split into two parts just over and hour:
# segment "$( basename $FILE ) part 1.avi" "00:00:00" "01:00:05"
# segment "$( basename $FILE ) part 2.avi" "00:59:55" "01:00:05"
        chmod 755 "$"

        cat <<EOF
To remaster this recording, see $

        set_directory "$2"
        if [ -e "$" ]
            if [ -n "$3" ]
                date +"Will \`kill -INT $(< "$" )\` at %X..." -d "+$( echo "$3" | sed -e 's/h/ hour/' -e 's/m/ minute/' -e 's/^\([0-9]*\)s\?$/\1 second/' )" \
                    && sleep "$3" \
                    || exit 0
            kill -s 2 "$(< "$" )" \
                && echo "Ran \`kill -INT $(< "$" )\` at %"
            echo "Cannot kill - not recording in $DIRECTORY"


        # we use ffmpeg and sox here, as they have better remastering tools and GStreamer doesn't offer any particular advantages


        # so people that don't understand shell scripts don't have to learn about variables:
        audio_delay() {
            if [[ "$1" =~ ^[0.]*$ ]]
            then AUDIO_DELAY=
            else AUDIO_DELAY="$1"
        original() { ORIGINAL="$1" ; }

        # build a segment:
        segment() {

            if [ -e "$SEGMENT_FILENAME" ]
                read -p "Are you sure you want to delete the old $SEGMENT_FILENAME (y/N)? "
                if [ "$REPLY" = "y" ]
                then rm -f "$SEGMENT_FILENAME"
                else return

            # Calculate segment:
            if [ -z "$SEGMENT_START" ]
                SEGMENT_END_OPTS="$(( $( parse_time "$SEGMENT_END" ) - $( parse_time "$SEGMENT_START" ) ))";
                TOTAL_TIME_MS="${SEGMENT_END_OPTS}000" # initial estimate, will calculate more accurately later
                SEGMENT_END_OPTS="-t $( echo "$SEGMENT_END_OPTS" | sed -e s/\\\([0-9][0-9][0-9]\\\)$/.\\\1/ )000"


            if [ -e "$AUDIO_FILE" ]
            then STAGE_COUNT=2
            else STAGE_COUNT=3

            [ -e "$AUDIO_FILE" ] || echo "Edit audio file $AUDIO_FILE and rerun to include hand-crafted audio"

            START_TIME="$( date +%s )"
            while IFS== read PARAMETER VALUE
                if   [ "$PARAMETER" = frame ]
                then FRAME=$VALUE
                    [ "$PARAMETER" = out_time_ms ] && OUT_TIME_MS="$VALUE"
                    echo $PARAMETER=$VALUE
            done < <( $CMD_FFMPEG $SEGMENT_START_OPTS -i "$ORIGINAL" $SEGMENT_END_OPTS -vcodec copy -an -f null /dev/null -progress /dev/stdout < /dev/null ) \
                 > >( ffmpeg_progress "$SEGMENT_FILENAME: $CURRENT_STAGE/$STAGE_COUNT calculating framerate" )
            CURRENT_STAGE=$(( CURRENT_STAGE + 1 ))

            # Build audio file for segment:
            if ! [ -e "$AUDIO_FILE" ]
                START_TIME="$( date +%s )"
                # Step one: extract audio
                $CMD_FFMPEG -y -progress >( ffmpeg_progress "$SEGMENT_FILENAME: extracting audio" ) $SEGMENT_START_OPTS -i "$ORIGINAL" $SEGMENT_END_OPTS -vn -f wav >(
                    case "${AUDIO_DELAY:0:1}X" in # Step two: shift the audio according to the audio delay
                            # no audio delay
                            # negative audio delay - trim start
                            $CMD_SOX -V1 -t wav - -t wav - trim 0 "${AUDIO_DELAY:1}"
                            # positive audio delay - prepend silence
                            $CMD_SOX -t wav <( $CMD_SOX -n -r "$AUDIO_BITRATE" -c 2 -t wav - trim 0.0 "$AUDIO_DELAY" ) -t wav -
                    esac | \
                    if [ -z "$GLOBAL_NOISE_PROFILE" ] # Step three: denoise based on the global noise profile, then normalise audio levels
                    then $CMD_SOX -t wav - "$AUDIO_FILE"                                                                                          norm -1
                    else $CMD_SOX -t wav - "$AUDIO_FILE" noisered <( echo "$GLOBAL_NOISE_PROFILE" | tr '\t' '\n' ) "${GLOBAL_NOISE_AMOUNT:-0.21}" norm -1
                    fi 2> >( grep -vF 'sox WARN wav: Premature EOF on .wav input file' )
                ) < /dev/null
                CURRENT_STAGE=$(( CURRENT_STAGE + 1 ))
            echo -n $'\r'"$( echo -n "$MESSAGE" | tr -c '' ' ' )"$'\r' >&2

            # Build video file for segment:
            START_TIME="$( date +%s )"
            $CMD_FFMPEG \
                -progress file://>( ffmpeg_progress "$SEGMENT_FILENAME: $CURRENT_STAGE/$STAGE_COUNT creating video" ) \
                $SEGMENT_START_OPTS -i "$ORIGINAL" \
                -i "$AUDIO_FILE" \
                -map 1:0 -map 0:1 \
                -r "$FRAMERATE" \
                $SEGMENT_END_OPTS \
                "$SEGMENT_FILENAME" \
                < /dev/null
            sleep 0.1 # quick-and-dirty way to ensure ffmpeg_progress finishes before we print the next line
            echo "$SEGMENT_FILENAME saved"



        SCRIPT_FILE="$( readlink -f "$2" )"
        cd "$( dirname "$SCRIPT_FILE" )"
        source "$SCRIPT_FILE"

        if [ -z "$HAVE_REMASTERED" ]
        then echo "Please specify at least one segment"


        echo "$HELP_MESSAGE"


This script generates a video in two passes: first it records and builds statistics, then lets you analyse the output, then builds an optimised final version.

Bash script to record video tapes with entrans

 # Test ob doppelt geöffnet
 if [[ -e "~/.lock_shutdown.digitalisieren" ]]; then
     echo ""
     echo ""
     echo "Capturing already running. It is impossible to capture to tapes simultaneously. Hit a key to abort."
     read -n 1
 # trap keyboard interrupt (control-c)
 # run if user hits control-c
   exit $?
   rm ~/.lock_shutdown.digitalisieren
   return $?
 touch "~/.lock_shutdown.digitalisieren"
 echo ""
 echo ""
 echo "Please enter the length of the tape in minutes and press ENTER. (Press Ctrl+C to abort.)"
 echo ""
 while read -e laenge; do
     if [[ $laenge == [0-9]* ]]; then
         break 2
         echo ""
         echo ""
         echo "That's not a number."
         echo "Please enter the length of the tape in minutes and press ENTER. (Press Ctrl+C to abort.)"
         echo ""
 let laenge=laenge+10  # Sicherheitsaufschlag, falls Band doch länger
 let laenge=laenge*60
 echo ""
 echo ""
 echo "Please type in the description of the tape."
 echo "Don't forget to rewind the tape?"
 echo "Hit ENTER to start capturing. Press Ctrl+C to abort."
 echo ""
 read -e name;
 # Falls Name schon vorhanden
 if [[ -e "$targetdirectory/$name.mpg" ]]; then
     while [[ -e "$targetdirectory/$name.$nummer.mpg" ]]; do
        let nummer=nummer+1
 # Audioeinstellungen setzen: unmuten, Regler
 amixer -D pulse cset name='Capture Switch' 1 >& /dev/null      # Aufnahme-Kanal einschalten
 amixer -D pulse cset name='Capture Volume' 20724 >& /dev/null  # Aufnahme-Pegel einstellen
 # Videoinput auswählen und Karte einstellen
 v4l2-ctl --set-input 3 >& /dev/null
 v4l2-ctl -c saturation=80 >& /dev/null
 v4l2-ctl -c brightness=130 >& /dev/null
 let ende=$(date +%s)+laenge
 echo ""
 echo "Working"
 echo "Capturing will be finished at "$(date -d @$ende +%H.%M)"."
 echo ""
 echo "Press Ctrl+C to finish capturing now."
 nice -n -10 entrans -s cut-time -c 0-$laenge -m --dam -- --raw \
 v4l2src queue-size=16 do-timestamp=true device=$VIDEO_DEVICE norm=PAL-BG num-buffers=-1 ! stamp sync-margin=2 sync-interval=5 silent=false progress=0 ! \
    queue leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 ! dam ! \
    cogcolorspace ! videorate ! \
    'video/x-raw-yuv,width=720,height=576,framerate=25/1,interlaced=true,aspect-ratio=4/3' ! \
    queue leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 ! \
    ffenc_mpeg2video rc-buffer-size=1500000 rc-max-rate=7000000 rc-min-rate=3500000 bitrate=4000000 max-key-interval=15 pass=pass1 ! \
    queue leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 ! mux. \
 pulsesrc buffer-time=2000000 do-timestamp=true ! \
    queue leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 ! dam ! \
    audioconvert ! audiorate ! \
    audio/x-raw-int,rate=48000,channels=2,depth=16 ! \
    queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 ! \
    ffenc_mp2 bitrate=192000 ! \
    queue leaky=2 max-size-buffers=0 max-size-time=0 max-size-bytes=0 ! mux. \
 ffmux_mpeg name=mux ! filesink location=\"$targetdirectory/$name.mpg\" >& /dev/null
 echo "Finished Capturing"
 rm ~/.lock_shutdown.digitalisieren

The script uses a command line similar to this to produce a DVD compliant MPEG2 file.

  • The script aborts if another instance is already running.
  • If not it asks for the length of the tape and its description
  • It records to description.mpg or if this file already exists to description.0.mpg and so on for the given time plus 10 minutes. The target-directory has to be specified in the beginning of the script.
  • As setting of the inputs and settings of the capture device is only partly possible via GStreamer other tools are used.
  • Adjust the settings to match your input sources, the recording volume, capturing saturation and so on.

Further documentation resources

External Links