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• iOS Objective C Opus audio stream kAudioFormatOpus — specification / conversion query

  16 avril 2022, par deffodeffo

  I'm working on a React Native voice app wanting to record straight to an Opus stream.

  


  On the iOS side at device level, I'm working in Objective C and I'm using an AVAudioSession recorder with formatID set to kAudioFormatOpus. The recorder captures audio data in the specified format and passes packets for upstreaming to the React Native app.

  


  This is all working well and emitting a stream of audio data when I run my code on an iOS Simulator or real device.

  


  My problem is when I receive the audio stream in the backend, I find I'm unable to reliably decode the Opus stream using ffmpeg. (I wish to decode to PCM.)

  


  Whilst I could post up specific code to show what I'm doing in more detail, my question at this stage is more generic in nature :

  


  Is anyone familiar with the format of Opus audio stream that is generated by iOS AudioSession recorder under audio format kAudioFormatOpus ? Is anyone able to suggest proven conversion techniques ? (e.g. some ffmpeg commands), or else hook me up with some links to format specs so I can figure out what is going on here ?

  


  The Apple Developer documentation contains zero useful information : https://developer.apple.com/documentation/coreaudiotypes/1572096-audio_data_format_identifiers/kaudioformatopus?changes=la__2&language=objc

  


  I have looked all over the internet, but I'm unable to find any useful spec info that corresponds to the stream that AVAudioSession recorder is outputting. I have seen a few posts which say Apple have not fully complied with the Opus spec, but I don't know enough about the proper structure of Opus to ascertain this for myself.

  


  Any help would be very much appreciated.

  


  Thanks all

  


• Convert a file to match another's specification exactly with FFMPEG ?

  25 mars 2022, par user1748217

  I have a series of files from the same source and so of the exact same format in every way that I'm concatenating with FFMPEG

  


  

  • file1.mov

  


  • file2.mov

  


  • file3.mov

  


  


  This is very fast and is working fine however no I want to take optional intro file (from many different source and of many different types) and convert that to match the others before joining.

  


  

  • intro.mp4

  


  


  How do I do this with FFMPEG ?

  


  Does this give me everything I need ?

  


  ffprobe -select_streams a:0 -show_entries \
  stream=codec_name,channels -of default=nw=1:nk=1 -v 0 ./file1.mov

ffprobe -select_streams v:0 -show_entries \
  stream=codec_name,width,height,r_frame_rate,pix_fmt \
  -of default=nw=1:nk=1 -v 0 ./file1.mov


  


  So with that I can just :

  


  ffmpeg -i intro.mp4 \ 
   -c:v h264 -s 1280x720 -pix_fmt yuv420p -framerate 30/1  \ 
   -c:a pcm_s16le -ca 1 intro.mov


  


  and then merge it seamlessly to the rest ?

  


  ffmpeg -f concat -safe 0 -i videos.txt -c copy merged.mov -y


  


  The answer is of course "no", hence the request for your support.
The audio is fine when files 1, 2 & 3 are merged but is too fast when the intro + 1, 2 & 3 are merged. The converted intro file always plays fine on it's own after the conversion and after the merge, but the others play audio too fast after the merge.

  


  What am I missing ?

  


  UPDATE :

  


  So in the end this worked for the intro :

  


  ffmpeg -i intro.mp4 \ 
   -c:v h264 -s 1280x720 -pix_fmt yuv420p -framerate 30 \ 
   -c:a pcm_s16le -ac 1 -b:a 512k -ar 32000 intro.mov -y


  


• FFmpeg trim filter fails to use sexagisimal time specification and the output stream is empty. Is it a bug and is there a fix ?

  29 novembre 2020, par Link-akro

  With ffmpeg the trim filter (or its audio variant atrim) malfunctions whenever i try to write a sexagesimal time specification for the boundaries or duration parameters of the filter : the result is always an empty stream.

  


  According to the documentation i already linked, which was generated on November 26, 2020 so far, the time duration specification should be supported.

  


  


  start, end, and duration are expressed as time duration specifications

  


  


  Quote of the spec.

  


  


  [-][HH :]MM:SS[.m...]

  


  


  


  [-]S+[.m...][s|ms|us]

  


  


  I am working in a Windows_10-64bit and scripting in its CMD.exe command-line, should it matter.

  


  Here is an instance of trimming a video stream out of a media file with hardcoded value for simplicity. The audio is retained unless i use atrim as well.
We may or may not append the setpts filter as recommended in the documentation henceforth setpts=PTS-STARTPTS if we want to shift the stream to start at the beginning of the cut range.

  


      ffmpeg -i sample-counter.mp4 -vf "trim=start=start='1:2':end='1:5'" sample-counter-trimmed.mp4


  


  If i use decimal it works as intended.

  


      ffmpeg -i sample-counter.mp4 -vf "trim=start=start='2':end='5'" sample-counter-trimmed.mp4


  


  This is the banner of my ffmpeg build. We may see it is the latest build by gyan.dev at the moment i post this.

  


      ffmpeg version 4.3.1-2020-11-19-full_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
      built with gcc 10.2.0 (Rev5, Built by MSYS2 project)
      configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-libsnappy --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libzvbi --enable-librav1e --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint


  


  The incorrect stream is always reported as 0kB weight in the output of ffmpeg. I confirmed with ffprobe Movie_Countdown-trim-2-5.mov -show_streams -show_entries format=duration that the incorrect stream is duration zero.

  


  Should i report it as a bug ? Is there some correction or workaround ?

  


  I would rather a solution with the trim filter itself, but if not possible a CMD batch scripting.
We should not need a different filter like select or seeking option like there are tutorials and questions/answers everywhere already. Scripting would be straight-forward in a proper *nix shell+distrib and taught everywhere so it is not worth caring while CMD answers on the other hand are rare so a scripting workaround for CMD would have some microcosmic worth since making one is considerably less straight-forward than a shell with modern mathematics and parsing abilities built-in or packaged.