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• Converting AAC stream to DASH MP4 with high fragment length precision

  5 mars 2017, par vdudouyt

  For my HTML5 project I need to create a fragmented MP4 file with a single audio stream (no video), each fragment of which has a duration of exactly 0.1 second.

  Accordingly to ffmpeg docs, you can accomplish that by passing a value in microseconds with ’-frag_duration’ - which I found to be working and playable with HTML5 MediaSource API :

  $ ffmpeg -y -i input.aac -c:a libfdk_aac -b:a 64k -level:v 13 -r 25 -strict experimental -movflags empty_moov+default_base_moof -frag_duration 100000 output.mp4

  As we have a 210 second audio split up by 0.1s fragments, I expect that in output.mp4 we’d have 2100 fragments, hence 2100 moof atoms. But, upon inspecting it I’ve figured out that we only have 1811 moof atoms - which means that some (or maybe even all) fragments are bigger than expected :

  $ python ~/git/mp4viewer/src/showboxes.py output.mp4 |grep moof|wc -l
  
  1811

  Could anybody tell me what’s wrong, and how could I accomplish what I want ?

  Right now my assumption is that during an encoding I have AAC frame length which is not a multiple of 0.1s, hence ffmpeg has no chance to produce the fragments that are strictly equal to 0.1s but I’m not sure. If somebody can confirm that - and let me know a way to explicitly set AAC frame_size in FFMPEG (I couldn’t find anything like that in the docs), or completely disprove this - it would be also highly appreciated.

• Create dynamic audio broadcast stream (node, ffmpeg, ..?)

  24 mars 2021, par TheCrunkiee

  I have coded a videoboard. Like a soundboard but with video. You go to one URL that's just a black screen and another one which has a list of different videos (sender). When you click one of these videos it plays on the black screen (receiver). If you play 2 different videos at the same time both videos are shown next to each other on the receiver. That's working fine for several months now. It just creates multiple html video-elements with multiple source-tags (x265 mp4 and vp9 webm).

  


  I recently made a discord bot which takes the webm, extracts the opus stream and plays its sound in the voice channel where the bot is connected. This has one disadvantage : It can only play one sound at a time. It happens a lot that there are multiple videos/sounds playing at the same time so this a bit of a bummer.

  


  So I thought I should create a audiostream on the server which hosts the videoboard and just connect the bot to that stream. But I have no clue how to do this. All I know is that it's very likely going to involve ffmpeg.

  


  What would be the best option here ? What I think I would need is basically an infinite silence stream and the possibility to add a audio file onto that stream at any point which will play simultaneously with other audio files that were added before and have not ended payback yet. How is that possible ? Somehow with m3u8 playlist-files or via rtsp protocol ?

  


  Thanks :)

  


• WebSRT and HTML5 media accessibility

  9 septembre 2010, par silvia

  On 23rd July, Ian Hickson, the HTML5 editor, posted an update to the WHATWG mailing list introducing the first draft of a platform for accessibility for the HTML5 <video> element. The platform provides for captions, subtitles, audio descriptions, chapter markers and similar (...)