On Fri, Oct 18, 2013 at 10:36:17PM +0100, James Stone wrote: You can import the video in Ardour3, this will give you a video timeline in Ardour's editor (i.e. a 'track' consisting of images) as well as synced video playback in a separate window. Within A3 you can do with the audio whatever you want, and when done you can export the whole to a new video file. I used this a few weeks ago to edit and mix the audio for a 1.5 hour documentary, and it worked rather well. You may need some iterations when setting things up (getting to know the quirks of the import and export dialogs) but otherwise it works well. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

On Fri, 2013-10-18 at 22:36 +0100, James Stone wrote: I don't know research about it, but there are recommendations for video and film. "The accuracy of A/V synchronisation at each stage should lie within the range of Audio 5 ms early (soundbefore picture) to 15 ms late (sound after picture). Limits of the relative timing of the sound component of a television programme relative to the corresponding picture component Sound before picture Sound after picture ≤ 40 ms ≤ 60 ms" - http://tech.ebu.ch/docs/r/r037.pdf "For television applications, audio should lead video by no more than 15 milliseconds and audio should lag video by no more than 45 milliseconds.For film, acceptable lip sync is considered to be no more than 22 milliseconds in either direction." - https://en.wikipedia.org/wiki/Audio_to_video_synchronization#Recommendations http://en.wikipedia.org/wiki/Lip_flap#Recommendations Related: http://en.wikipedia.org/wiki/Clapperboard

On Fri, 18 Oct 2013, James Stone wrote: The same player might have non-sync video/audio depending on the audio output path. One would expect the player to compensate for the delays cause by decoding, but then the audio may go straight to the alsa device or through pulse which adds latency or through jack (where you can set the latency so one time might have a different delay than another) or through gs first then through who knows what. Some of the sound servers may have variable latency depending on cpu load too. That is not even touching the video chain (outside of my knowledge), but you get the idea. On any reasonably fast computer one can set up an audio/video chain that will work in sync, but change any one part of that chain and sync may be no more. So if you are looking from a distribution POV, you have no control of what the audience will play your video on. If you are using it in an institution, then all the machines may be the same as well as the OS/SW and so there is a better chance. Keeping the decoding work as low as possible and using sample rates and frame rates that match the sound serve/audio IF and work as close to the GPU setup as possible reduces things like resampling/resizing and reduces cpu load and brings different speed HW closer to matching. For example, if your audio is using opus as an codec, then 48k sample rate in the sound server/audio IF will result in no resample to match the codec ouput. But if the audio card is set to 44.1k then the audio must be resampled. However, many of the videos available on the Inet have audio at 44.1k and so running the audio IF at 48k would mean a resample step for them. So it would seem it would be impossible to make sure that audio and video are always in sync on every machine. The recording and encoding sw can shift the audio with respect to the video. Then check it out on the machine/OS that most of your target audience uses. I don't know how the windows audio server works, however, the intel HDA audio interface assumes 48k (or 96k or 192k... though even at these speeds it seems the bus runs at 48k with just more parallel data). So I would guess the default windows setup would be 48k. I don't really know though. -- Len Ovens www.ovenwerks.net