GStreamer

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 this page will explain how to build an encoder using its command-line interface.

Introduction to GStreamer

No two use cases for encoding are quite alike. Is your processor fast enough to encode high quality video? Do you want to play your video in DVD players, or is it enough that it works in your version of VLC? Which obscure quirks does your system have?

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 48,000kHz 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 47,999kHz, 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:

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

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/*-gst-launch.PLAYING_READY.dot > 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/*-gst-launch.PLAYING_READY.dot > 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

Determining your video device

See all your video devices by doing:

ls /dev/video*

One of these is the device you want. Most people only have one, or can figure it out by disconnecting devices and rerunning the above command. Otherwise, check the capabilites of each device:

for VIDEO_DEVICE in /dev/video* ; do echo ; echo ; echo $VIDEO_DEVICE ; echo ; v4l2-ctl --device=$VIDEO_DEVICE --list-inputs ; done

Usually you will see e.g. a webcam with a single input and a TV card with multiple inputs. If you're still not sure which one is yours, try each one in turn:

v4l2-ctl --device=<device> --set-input=<whichever-input-you-want-to-use>
gst-launch-0.10 v4l2src do-timestamp=true device=<device> ! autovideosink

(if your source is a VCR, remember to play a video so you know the right one when you see it)

If you like, you can store your device in an environment variable:

VIDEO_DEVICE=<device>

All further examples will use $VIDEO_DEVICE in place of an actual video device

Determining your audio device

See all of our audio devices by doing:

arecord -l

Again, it should be fairly obvious which of these is the right one. Get the device names by doing:

arecord -L | grep ^hw:

If you're not sure which one you want, try each in turn:

gst-launch-0.10 alsasrc do-timestamp=true device=hw:<device> ! autoaudiosink

Again, you should hear your tape playing when you get the right one. Note: always use an ALSA hw device, as they are closest to the hardware. Pulse audio devices and ALSA's plughw devices add extra layers that, while more convenient for most uses, only cause headaches for us.

Optionally set your device in an environment variable:

AUDIO_DEVICE=<device>

All further examples will use $AUDIO_DEVICE in place of an actual audio device

Getting your device capabilities

To find the capabilities of your audio device, do:

gst-launch-0.10 --gst-debug=alsa:5 alsasrc device=$AUDIO_DEVICE ! fakesink 2>&1 | grep 'returning caps'

To find the capabilities of your video device, do:

gst-launch-0.10 --gst-debug=v4l2src:3 v4l2src device=$VIDEO_DEVICE ! fakesink 2>&1 | grep 'probed caps:'

These will display a list of capability ranges. You might find them more readable if you paste them into a text editor and replace all the semicolons with newlines. When you build a pipeline, you will need to specify capabilities based on these ranges. For example, given a video capability range video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)25/1, width=(int)[ 48, 720 ], height=(int)[ 32, 578 ], interlaced=(boolean)true, pixel-aspect-ratio=(fraction)1/1, you might choose a capability video/x-raw-yuv, width=720, height=576 and allow GStreamer to choose sensible defaults for the other values.

Finding the best height

Some devices report a maximum height of 578. A PAL TV signal is 576 lines tall and an NTSC signal is 486 lines, so height=578 won't give you the best picture quality. To confirm this, tune to a non-existent TV channel then take a screenshot of the snow:

gst-launch-0.10 -q v4l2src device=$VIDEO_DEVICE \
	! video/x-raw-yuv,height=578 \
	! imagefreeze \
	! autovideosink

Here's an example of what you might see - notice the blurring in the middle of the picture. Now take a screenshot with the appropriate height for your TV norm:

gst-launch-0.10 -q v4l2src device=$VIDEO_DEVICE \
	! video/x-raw-yuv,height=<appropriate-height> \
	! imagefreeze \
	! autovideosink

Here's an example taken with height=576 - notice the middle of this picture is nice and crisp.

You may want to test this yourself and set your height to whatever looks best.

Measuring your video framerate

As mentioned above, some devices produce slightly too many (or too few) frames per second. To check your system's actual frames per second, start your video source (e.g. a VCR or webcam) then run this command:

gst-launch-0.10 v4l2src ! fpsdisplaysink fps-update-interval=100000

1. Let it run for 100 seconds to get a large enough sample. It should print some statistics in the bottom of the window - write down the number of frames dropped
2. Let it run for another 100 seconds, then write down the new number of frames dropped
3. Calculate (second number) - (first number) - 1 (e.g. 5007 - 2504 - 1 == 2502)
   • You need to subtract one because fpsdisplaysink drops one frame every time it displays the counter
4. That number is exactly one hundred times your framerate, so you should tell GStreamer e.g. framerate=2502/100

Note: VHS framerates can vary within the same file. To get an accurate measure of a VHS recording's framerate, encode to a format that supports variable framerates then retrieve the video's duration and total number of frames. You can then transcode a file with your desired frame rate.

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

Fixing initialisation timing errors

If your hardware doesn't support timestamps, your encoded audio and video might be desynchronised by a fixed amount throughout the video. This offset is based on too many factors to isolate (e.g. a new driver version might increase or decrease the value), so fixing this is a manual process that probably needs to be done every time you encode a file.

Note: using a plughw source can cause initialisation timing errors. If your video and audio are desynchronised from the start, make sure you're using a hw source.

Calculate your desired offset:

1. Record a video using one of the techniques below
2. Open the video in your favourite video player
3. Adjust the A/V sync until it looks right to you - different players put this in different places, for example it's Tools > Track Synchronisation in VLC
4. write down your desired offset

If possible, look for (or create) clapperboard-like events - moments where an obvious visual element occurred at the same moment as an obvious audio moment. A hand clapping or a cup being placed on a table are good examples.

Extract your audio:

gst-launch-0.10 \
	uridecodebin uri="file:///path/to/my.file" \
		! progressreport \
		! audioconvert \
		! audiorate \
		! wavenc \
		! filesink location="/path/to/my.file.wav"

If you have a clapperboard event, you might want to examine the extracted file in an audio editor like Audacity. You should be able to see the exact time of the clap sound in the audio stream, watch the video to isolate the exact frame, and use that information to calculate the precise audio delay.

Use sox to prepend some silence:

sox -S -t wav <( sox -V1 -n -r <bitrate> -c <audio-channels> -t wav - trim 0.0 <delay-in-seconds> ) "/path/to/my.file.wav" "/path/to/my.file.flac"

Mix the new audio and the old video into a new file:

gst-launch-0.10 \
	uridecodebin uri="file:///path/to/my.file" \
		! video/your-video-settings \
		! mux. \
	uridecodebin uri="file:///path/to/my.file.flac" \
		! audioconvert \
		! audiorate \
		! your_preferred_audio_encoder \
		! mux. \
	avimux name=mux \
		! filesink location="/path/to/my.file.new"

Note: you can apply any sox filter this way, like normalising the volume or removing background noise.

A specific solution for measuring your offset

Measuring your offset will probably be the most unique part of your recording solution. Here is one solution you could use when digitising old VHS tapes:

1. Connect a camcorder to your VCR
2. Tune the VCR so it shows the camcorder output when it's not playing
3. Start your GStreamer pipeline
4. Clap your hands in front of the camcorder so you can later measure A/V synchronisation
5. Press play on the VCR
6. When the video has finished recording, split the audio and video tracks as described above
7. Examine the audio with Audacity and identify the precise time of the clap sound
8. Examine the video with avidemux and identify the frame of the clap image

You'll probably need to change every step of the above to match your situation, but hopefully it will provide some inspiration.

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

Choosing formats

When you create a video, you need to choose your audio format (e.g. WAV or MP3), video format (e.g. XviD or MPEG-4) and container format (e.g. AVI or MP4). There's constant work to improve the codecs that create audio/video and the muxers that create containers, and whole new formats are invented fairly regularly, so this page can't recommend any specific formats. Wikipedia's comparisons of audio, video and container formats are a good place to start your research - here are some important things to look for:

• encoding speed - encoders that generate too much CPU load will cause GStreamer to drop frames
• accuracy - some formats are lossless, others throw away information to improve speed and/or reduce file size
• file size - different formats use different amounts of disk space, even with the same accuracy
• compatibility - newer formats usually produce better results but can't be played by older software

Speed and accuracy are usually the most important when encoding, but size and compatibility the most important for playback. So it can make sense to encode with modern, fast, lossless formats then transcode to a format that produces a smaller or more compatible file. For example, as of 2015 it might make sense to encode FLAC audio and x264 video into a Matroska file, then transcode MP3 audio and MPEG-4 video into an AVI file. The transcoded file might be larger or lower quality, but it should play on most software, and you can just delete it and try again if your grandmother's DVD player doesn't like it.

If you use separate encode and transcode pipelines, make sure to do any video processing in the transcode pipeline. Reducing noise, masking out overscan areas etc. usually increases CPU usage and sometimes introduces errors, which are much easier to fix if you don't have to re-record the original.

Cleaning audio

Any analogue recording will contain a certain amount of background noise. Cleaning noise is optional, and you'll always be able to produce a slightly better result if you spend a little longer on it. GStreamer doesn't have built-in noise reduction, so this section will just introduce enough theory to get you started. Audacity's noise reduction effect is a good place to start experimenting.

The major noise sources are:

• your audio codec might throw away sound it thinks you won't hear in order to reduce file size
• your recording system will produce a small, consistent amount of noise based on its various electrical and mechanical components
• VHS format limitations cause static at high and low frequencies, depending on the VCR's settings
• imperfections in tape recording and playback produce noise that differs between recordings and even between scenes

Always use a lossless audio format while cleaning (e.g. WAV or FLAC). Even if you plan to eventually use a format like MP3 that throws information away, the extra noise makes it harder to isolate the noise sources you're actually trying to measure.

The primary means of reducing noise is the frequency-based noise gate, which blocks some frequencies and passes others. High-pass and low-pass filters pass noise above or below a certain frequency, and can be combined into band-pass or even multi-band filters. The rest of this section discusses how to build a series of noise gates for your audio.

Identify noise from your recording system by recording the sound of a paused tape or silent television channel for a few seconds:

gst-launch-0.10 alsasrc device=$AUDIO_DEVICE ! wavenc ! filesink location=baseline.wav

You can use this baseline recording as a noise profile, which your software uses to build a multi-band noise gate. You can then apply that noise gate to any future recordings with the same hardware.

Identify VHS format limitations by searching online for information based on your TV norm (NTSC, PAL or SECAM), your recording quality (normal or Hi-Fi) and your VHS play mode (short- or long-play). Wikipedia's discussion of VHS audio recording is a good place to start. If you're able to find the information, gate your recordings with high-pass and low-pass filters that only allow frequencies within the range your tape actually records. For example, a long-play recording of a PAL tape will produce static below 100Hz and above 4kHz so you should gate your recording with a high-pass filter at 100Hz and a low-pass filter at 4000Hz. If you can't find the information, you can determine it experimentally by trying out different filters to see what sounds right - your system probably produces static below about 10Hz or 100Hz and above about 4kHz or 12kHz, so try high- and low-pass filters in those ranges. If you don't remove this noise source, the next step will do a reasonable job of guessing it for you anyway.

Identify imperfections in recording and playback by watching the video and looking for periods of silence. You only need half a second of background noise to generate a profile, but the number of profiles is up to you. Some people grab one profile for a whole recording, others combine clips into averaged noise profiles, others cut audio into scenes and de-noise each in turn. At a minimum, tapes with multiple recordings should be split up and each one de-noised separately - a tape containing a TV program recorded in LP mode in one VCR followed by a home video recorded in SP in another VCR will produce two very different noise profiles, even if played back all in one go.

It's good to apply filters in the right order (system profile, then VHS limits, then recording profiles), but beyond that noise reduction is very subjective. You can run your audio through as many gates as you like, and even repeat the same filter several times. If you use a noise reduction profile, you can even get different results from different programs (see for example this comparison of sox and Audacity's algorithms). You'll always be able to get a better result if you spend more time on the problem, so you'll need to decide for yourself when the result is good enough.

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=$VIDEO_DEVICE do-timestamp=true norm=PAL-I \
		! queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
		! video/x-raw-yuv,interlaced=true,width=720,height=576 \
		! x264enc interlaced=true pass=quant option-string=qpmin=0:qpmax=0 speed-preset=ultrafast tune=zerolatency byte-stream=true \
		! progressreport update-freq=1 \
		! mux. \
	alsasrc device=$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=2000000000 \
		! 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 5-10GB per hour and it's only supported by relatively recent software. Consider transcoding to a smaller/more compatible format after encoding.

min-index-interval=2000000000 improves seek times by telling the Matroska muxer to create cue data enteries every two seconds (two billion nanoseconds). This increases file size by a few kilobytes an hour.

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.

#!/bin/bash
#
# 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
"

CONFIGURATION='#
# CONFIGURATION FOR GSTREAMER RECORD SCRIPT
# For more information, see http://www.linuxtv.org/wiki/index.php/GStreamer
#

#
# VARIABLES YOU NEED TO EDIT
# 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:
# http://forum.videohelp.com/threads/215570-Sensible-resolution-for-VHS-captures?p=1244415#post1244415
# Run `'"$0"' --caps` to find your supported width, height and bitrate:
SOURCE_WIDTH="${SOURCE_WIDTH:-720}"
SOURCE_HEIGHT="${SOURCE_HEIGHT:-576}"
AUDIO_BITRATE="${AUDIO_BITRATE:-32000}"

# For systems that do not automatically handle audio/video initialisation times:
AUDIO_DELAY="$AUDIO_DELAY"

#
# VARIABLES YOU MIGHT NEED TO EDIT
# 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:
#GST_VERSION=0.10

# Set these to alter the recording quality:
#GST_X264_OPTS="..."
#GST_FLAC_OPTS="..."

# Set these to control the audio/video pipelines:
#GST_QUEUE="..."
#GST_VIDEO_CAPS="..."
#GST_AUDIO_CAPS="..."
#GST_VIDEO_SRC="..."
#GST_AUDIO_SRC="..."

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

# Reducing noise:
#GLOBAL_NOISE_AMOUNT=0.21

#
# VARIABLES SET AUTOMATICALLY
#
# 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
'

#
# CONFIGURATION SECTION
#

CONFIG_SCRIPT="$HOME/.v4l-record-scriptrc"
[ -e "$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_FLAC_OPTS=""
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 "$GST_VERSION" in
	0.10)
		GST_VIDEO_FORMAT="-yuv"
		GST_AUDIO_FORMAT="-int"
		;;
	1.0)
		GST_VIDEO_FORMAT=""
		GST_AUDIO_FORMAT=""
		;;
	*)
		echo "Please specify 'GST_VERSION' of '0.10' or '1.0', not '$GST_VERSION'"
		exit 1
		;;
esac

# `gst-inspect-0.10 <element> | less -i` for a list of properties (e.g. `gst-inspect-0.10 v4l2src | less -i`):
GST_QUEUE="${GST_QUEUE:-queue max-size-buffers=0 max-size-time=0 max-size-bytes=0}"
GST_VIDEO_CAPS="${GST_VIDEO_CAPS:-video/x-raw$GST_VIDEO_FORMAT,interlaced=true,width=$SOURCE_WIDTH,height=$SOURCE_HEIGHT}"
GST_AUDIO_CAPS="${GST_AUDIO_CAPS:-audio/x-raw$GST_AUDIO_FORMAT,depth=16,rate=$AUDIO_BITRATE}"
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"

#
# UTILITY FUNCTIONS
# 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, "
			;;
	esac
}

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

ffmpeg_progress() {

	MESSAGE="$1..."

	echo -n $'\r'"$MESSAGE" >&2
	while IFS== read PARAMETER VALUE
	do
		if [ "$PARAMETER" = out_time_ms ]
		then
			echo -n $'\r'"$( echo -n "$MESSAGE" | tr -c '' ' ' )"$'\r' >&2
			if [ -z "$TOTAL_TIME_MS" -o "$TOTAL_TIME_MS" = 0 ]
			then
				case $SPINNER in
					\-|'') SPINNER=\\ ;;
					\\   ) SPINNER=\| ;;
					\|   ) SPINNER=\/ ;;
					\/   ) SPINNER=\- ;;
				esac
				MESSAGE="$1 $SPINNER"
			else
				if [ -n "$VALUE" -a "$VALUE" != 0 ]
				then
					TIME_REMAINING=$(( ( $(date +%s) - $START_TIME ) * ( $TOTAL_TIME_MS - $VALUE ) / $VALUE ))
					HOURS_REMAINING=$(( $TIME_REMAINING / 3600 ))
					MINUTES_REMAINING=$(( ( $TIME_REMAINING - $HOURS_REMAINING*3600 ) / 60 ))
					SECONDS_REMAINING=$((   $TIME_REMAINING - $HOURS_REMAINING*3600 - $MINUTES_REMAINING*60 ))
					HOURS_REMAINING="$(   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)"
				fi
			fi
			echo -n $'\r'"$MESSAGE" >&2
		elif [ "$PARAMETER" = progress -a "$VALUE" = end ]
		then
			echo -n $'\r'"$( echo -n "$MESSAGE" | tr -c '' ' ' )"$'\r' >&2
			return
		fi
	done

}

# 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 "$1" ]
	then
		echo "$HELP_MESSAGE"
		exit 1
	else
		DIRECTORY="$( readlink -f "$1" )"
		FILE="$DIRECTORY/$( basename "$DIRECTORY" )"
	fi
}

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

#
# MAIN LOOP
#

case "$1" in

	-i|--i|--in|--ini|--init)
		if [ -e "$CONFIG_SCRIPT" ]
		then
			echo "Please delete $CONFIG_SCRIPT if you want to recreate it"
		else
			echo "$CONFIGURATION" > "$CONFIG_SCRIPT"
			echo "Please edit $CONFIG_SCRIPT to match your system"
		fi
	;;

	-p|--p|--pr|--pro|--prof|--profi|--profil|--profile)
		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"
	;;

	-c|--c|--ca|--cap|--caps)
		{
			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
	;;

	-r|--rec|--reco|--recor|--record)

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

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

		date +"%c: started recording $FILE.mkv"
		# trap keyboard interrupt (control-c)
		trap kill_gstreamer 0 SIGHUP SIGINT SIGQUIT SIGABRT SIGKILL SIGALRM SIGSEGV SIGTERM
		kill_gstreamer() { [ -e "/proc/$(< "$FILE.pid" )" ] && kill -s 2 "$(< "$FILE.pid" )" ; }
		sh -c "echo \$\$ > '$FILE.pid' && \
			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
			done
		trap '' 0 SIGHUP SIGINT SIGQUIT SIGABRT SIGKILL SIGALRM SIGSEGV SIGTERM
		echo >&2
		date +"%c: finished recording $FILE.mkv"

		rm -f "$FILE.pid"

		cat <<EOF > "$FILE-remaster.sh"
#!$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.
#
# *** REMASTERING OPTIONS ***

#
# AUDIO DELAY
#
# 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}

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

#
# SEGMENTS
#
# 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 $FILE ) part 2.avi" "00:59:55" "01:00:05"
EOF
		chmod 755 "$FILE-remaster.sh"

		cat <<EOF
To remaster this recording, see $FILE-remaster.sh
EOF
	;;

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

	-m|--rem|--rema|--remas|--remast|--remaste|--remaster)

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

		HAVE_REMASTERED=

		# 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"
			fi
		}
		original() { ORIGINAL="$1" ; }

		# build a segment:
		segment() {
			SEGMENT_FILENAME="$1"
			SEGMENT_START="$2"
			SEGMENT_END="$3"

			if [ -e "$SEGMENT_FILENAME" ]
			then
				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
				fi
			fi

			# Calculate segment:
			if [ -z "$SEGMENT_START" ]
			then
				SEGMENT_START_OPTS=
				SEGMENT_END_OPTS=
			else
				SEGMENT_START_OPTS="-ss $SEGMENT_START"
				SEGMENT_END_OPTS="$(( $( parse_time "$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"
			fi

			AUDIO_FILE="${SEGMENT_FILENAME/\.*/.wav}"

			CURRENT_STAGE=1
			if [ -e "$AUDIO_FILE" ]
			then STAGE_COUNT=2
			else STAGE_COUNT=3
			fi

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

			START_TIME="$( date +%s )"
			while IFS== read PARAMETER VALUE
			do
				if   [ "$PARAMETER" = frame ]
				then FRAME=$VALUE
				else
					[ "$PARAMETER" = out_time_ms ] && OUT_TIME_MS="$VALUE"
					echo $PARAMETER=$VALUE
				fi
				TOTAL_TIME_MS=$OUT_TIME_MS
				FRAMERATE="${FRAME}000000/$OUT_TIME_MS"
			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:
			MESSAGE=
			if ! [ -e "$AUDIO_FILE" ]
			then
				START_TIME="$( 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
						X)
							# no audio delay
							cat
							;;
						-)
							# 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 ))
			fi
			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 \
				$FFMPEG_VIDEO_OPTS $FFMPEG_AUDIO_OPTS \
				"$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"

			HAVE_REMASTERED=true

		}

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

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

	;;

	*)
		echo "$HELP_MESSAGE"

esac

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

#!/bin/bash
 
 targetdirectory="~/videos"
 
 
 # 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
     exit
 fi
 
 # trap keyboard interrupt (control-c)
 trap control_c 0 SIGHUP SIGINT SIGQUIT SIGABRT SIGKILL SIGALRM SIGSEGV SIGTERM
 
 control_c()
 # run if user hits control-c
 {
   cleanup
   exit $?
 }
 
 cleanup()
 {
   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
     else
         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 ""
     fi
 done
 
 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;
 name=${name//\//_}
 name=${name//\"/_}
 name=${name//:/_}
 
 # Falls Name schon vorhanden
 if [[ -e "$targetdirectory/$name.mpg" ]]; then
     nummer=0
     while [[ -e "$targetdirectory/$name.$nummer.mpg" ]]; do
        let nummer=nummer+1
     done
     name=$name.$nummer
 fi
 
 # 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

• Gstreamer project
• FAQ
• Documentation
  • Core plugins
  • FFMpeg plugins
  • Base plugins
  • Good plugins
  • Bad plugins
  • Ugly plugins
  • Entrans plugins
• man gst-launch
• entrans command line tool documentation
• gst-inspect plugin-name

External Links

• GStreamer project page
• GStreamer Wikipedia page
• GEntrans project