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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.

• Streaming android to windows

  13 juin 2017, par iYehuda

  I’m writing an app that enables controlling android devices from windows machines.
  Major part of controlling the device is viewing it’s screen. Currently, my android app (Java code) captures the screen on a fixed rate, compresses it (JPEG) and sends it, while the windows side (C# code) receives buffers of data, each for frame, decompresses them and displays the last decompressed frame.

  Two issues came up from this solution :

  1. Compression of a single image takes 0.3 seconds, which limits me to low FPS streaming with single thread for compressing. I made a thread pool for compressing captured frames, and it damages the app performance.

  2. The compression is not optimal. The screen can be idle for a while and a continuous transmission of the same frame would be done. Usage of streaming/encoding format would be handful and can ease the network traffic.

  I searched for encoding APIs such as MediaCodec and third party libraries such as ffmpeg. All those libraries encode videos and write them to files (maybe I misunderstood them ?).

  What API can I use for streaming my screen and follow these requirements :

  • Fast encoding / non blocking API
  • Outputs raw binary data for each frame. The data must be sent immediately
  • Can be embedded into my existing applicative protocol (protocol buffers based)
  • Available on C# (Windows) and Java or C++ (Android)

• AAC's converted from CAFs for HTTP Live Streaming

  13 mars 2014, par user2901994

  I have several AAC files that were converted from CAF files, for use in HTTP Live Streaming. The stream works, however there is a small gap between each AAC file. It is my understanding that this gap is caused by the "Priming" and "Remainder" frames that are attached to AAC files when they are transcoded from CAFs.

  My question is, is there any way to remove this gap ? Or use FFMpeg to wrap the files, (possibly in m4a ?) so that audio players (VLC, JWPlayer) will understand to skip the gap ?