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I'm looking for the fastest way to encode a webcam stream that will be viewable in a html5 video tag. I'm using a Pandaboard: http://www.digikey.com/product-highlights/us/en/texas-instruments-pandaboard/686#tabs-2 for the hardware. Can use gstreamer, cvlc, ffmpeg. I'll be using it to drive a robot, so need the least amount of lag in the video stream. Quality doesn't have to be great and it doesn't need audio. Also, this is only for one client so bandwidth isn't an issue. The best solution so far is using ffmpeg with a mpjpeg gives me around 1 sec delay. Anything better?

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ffmpeg.org/trac/ffmpeg/wiki/StreamingGuide#Latency lists some hints/clues for latency with ffmpeg –  rogerdpack Apr 30 '13 at 15:51

3 Answers 3

up vote 9 down vote accepted

I have been asked this many times so I will try and answer this a bit generically and not just for mjpeg. Getting very low delays in a system requires a bit of system engineering effort and also understanding of the components.

Some simple top level tweaks I can think of are:

Ensure the codec is configured for the lowest delay. Codecs will have (especially embedded system codecs) a low delay configuration. Enable it. If you are using H.264 its most useful. Most people don't realize that by standard requirements H.264 decoders need to buffer frames before displaying it. This can be upto 16 for Qcif and upto 5 frames for 720p. That is a lot of delay in getting the first frame out. If you do not use H.264 still ensure you do not have B pictures enabled. This adds delay to getting the first picture out.

Since you are using mjpeg, I don't think this is applicable to you much.

Encoders will also have a rate control delay. (Called init delay or vbv buf size). Set it to the smallest value that gives you acceptable quality. That will also reduce the delay. Think of this as the bitstream buffer between encoder and decoder. If you are using x264 that would be the vbv buffer size.

Some simple other configurations: Use as less I pictures as possible (large intra period). I pictures are huge and add to the delay to send over the network. May not be very visible in systems where end to end delay is in the range of 1 second or more but when you are designing systems that need end to end delay of 100ms or less this and several other aspects come into play. Also ensure you are using a low latency audio codec aac-lc (and not heaac).

In your case to get to lower latencies I would suggest to move out of MJPEG and use atleast mpeg4 without B pictures (Simple profile) or best is H.264 baseline profile (x264 gives a zerolatency option). The simple reason you will get lower latency is that you will get lower bitrate post encoding to send the data out and you can go to full framerate. If you must stick to mjpeg you have close to what you can get without more advanced features support from the codec and system using the open source components as is.

Another aspect is the transmission of the content to the display unit. If you can use udp it will reduce latency quite a lot compared to tcp though it can be lossy at times depending on network conditions. You have mentioned html5 video. I am curious as to how you are doing live streaming to a html5 video tag.

There are other aspects that can also be tweaked which I would put in the advanced category and requires the system engineer to try various things out

What is the network buffering in the OS? The OS also buffers data before sending it out for performance reasons. Tweak this to get a good balance between performance and speed.

Are you using CR or VBR encoding? While CBR is great for low jitter you can also use capped vbr if the codec provides it.

Can your decoder start decoding partial frames? So you don't have to worry about framing the data before providing it to the decoder. Just keep pushing the data to the decoder as soon as possible.

Can you do field encoding? Halves the time from frame encoding before getting the first picture out.

Can you do sliced encoding with callbacks whenever a slice is available to send over the network immediately?

In sub 100 ms latency systems that I have worked in all of the above are used. Some of the features may not be available in open source components but if you really need it and are enthusiastic you could go ahead and implement them.

EDIT: I realize you cannot do a lot of the above for a ipad streaming solution and there are limitations because of hls also to the latency you can achieve. But I hope it will prove useful in other cases when you need any low latency system.

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If you are using HLS, create 1 second chunks for the transport stream. I don't think you will be able to go lower than that irrespective of the above points because the ipad will itself wait for 1 second before it starts playing out the content. If it is not HLS I am curious how you play it out. –  av501 Aug 23 '12 at 6:21

Read the next answer, that suggests "If you are using HLS, create 1 second chunks for the transport stream"

Unreal Media Server can do that, you can get 2.5 seconds latency with HLS

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We had a similar problem, in our case it was necessary to time external events and sync them with the video stream. We tried several solutions but the one described here solved the problem and is extremely low latency:

Github Link

It uses gstreamer transcode to mjpeg which is then sent to a small python streaming server. This has the advantage that it uses the tag instead of so it can be viewed by most modern browsers, including the iPhone.

As you want the <video> tag, a simple solution is to use http-launch. That had the lowest latency of all the solutions we tried so it might work for you. Be warned that ogg/theora will not work on Safari or IE so those wishing to target the Mac or Windows will have to modify the pipe to use MP4 or WebM.

Although we really liked http-launch we needed to write a program because of the external event monitoring. The approach we settled on was to send the video stream to icecast via shout2send. We wrote an application in python, but you can easily experiment using gst-launch-1.0:


gst-launch-1.0 v4l2src \
  ! deinterlace \
  ! video/x-raw, width=640, height=480, format=YUY2 \
  ! videorate ! videoscale ! videoconvert \
  ! theoraenc bitrate=200 ! oggmux \
  ! shout2send sync=true ip=localhost password=xxxx mount=stream \
    streamname=video description="Video Feed"

Don't forget the caveats about only using a browser that supports ogg. We found Chrome to be the best, Firefox kept caching the live stream. <video> needs a switch to identify live streams IMHO.

One final solution looks promising, gst-streaming-server. We simply couldn't find enough documentation to make it worth pursuing. I'd grateful if somebody could ask a stackoverflow question about how it should be used!

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Thank you to the person that changed the Github link for me, appreciated! –  sgbirch May 24 at 12:52
have you considered WebRTC? –  Alex Cohn May 24 at 14:36

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