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• set MediaRecorder to record 1 frame every N seconds

  19 août 2022, par The Blind Hawk

  Summary

  


  I have a version of my code already working on Chrome and Edge, but I need some fixes for it to work on Safari.
  
My objective is to record around 25 minutes and download a timelapse version of the recording.
  
final product requirements :

  


  speed: 3fps
length: ~25s

(I need to record one frame every 20 seconds for 25 mins)


  


  this.secondStream settings :

  


  this.secondStream = await navigator.mediaDevices.getUserMedia({
    audio: false,
    video: {width: 430, height: 430, facingMode: "user"}
});


  


  My code for IOS so far :

  


          startIOSVideoRecording: function() {
            console.log("setting up recorder");
            var self = this;
            this.data = [];

            if (MediaRecorder.isTypeSupported('video/mp4')) {
                // IOS does not support webm, so I will be using mp4
                var options = {mimeType: 'video/mp4', videoBitsPerSecond : 1000000};
            } else {
                console.log("ERROR: mp4 is not supported, trying to default to webm");
                var options = {mimeType: 'video/webm'};
            }
            console.log("options settings:");
            console.log(options);

            this.recorder = new MediaRecorder(this.secondStream, options);

            this.recorder.ondataavailable = function(evt) {
                if (evt.data && evt.data.size > 0) {
                    self.data.push(evt.data);
                    console.log('chunk size: ' + evt.data.size);
                }
            }

            this.recorder.onstop = function(evt) {
                console.log('recorder stopping');
                var blob = new Blob(self.data, {type: "video/mp4"});
                self.download(blob, "mp4");
                self.sendMail(videoBlob);
            }

            console.log("finished setup, starting")
            this.recorder.start(1200);

            function sleep(ms) { return new Promise(resolve => setTimeout(resolve, ms));}

            async function looper() {
                // I am trying to pick one second every 20 more or less
                await sleep(500);
                self.recorder.pause();
                await sleep(18000);
                self.recorder.resume();
                looper();
            }
            looper();
        },


  


  Issues

  


  Only one call to getUserMedia()

  


  I am already using this.secondstream elsewhere, and I need the settings to stay as they are for the other functionality.
  
On Chrome and Edge, I could just call getUserMedia() again with different settings, and the issue would be solved, but on IOS calling getUserMedia() a second time kills the first stream.
  
The settings that I was planning to use (works for Chrome and Edge) :

  


  navigator.mediaDevices.getUserMedia({
    audio: false,
    video: { 
        width: 360, height: 240, facingMode: "user", 
        frameRate: { min:0, ideal: 0.05, max:0.1 } 
    },
}


  


  The timelapse library I am using does not support mp4 (ffmpeg as alternative ?)

  


  I am forced to use mp4 on IOS apparently, but this does not allow me to use the library I was relying on so I need an alternative.
  
I am thinking of using ffmpeg but cannot find any documentation to make it interact with the blob before the download.
  
I do not want to edit the video after downloading it, but I want to be able to download the already edited version, so no terminal commands.

  


  MediaRecorder pause and resume are not ideal

  


  On Chrome and Edge I would keep one frame every 20 seconds by setting the frameRate to 0.05, but this does not seem to work on IOS for two reasons.
  
First one is related to the first issue of not being able to change the settings of getUserMedia() without destroying the initial stream in the first place.
  
And even after changing the settings, It seems that setting the frame rate below 1 is not supported on IOS. Maybe I wrote something else wrong, but I was not able to open the downloaded file.
  
Therefore I tried relying on pausing and resuming the MediaRecorder, but this brings forth another two issues :
  
I am currently saving 1 second every 20 seconds and not 1 frame every 20 seconds, and I cannot find any workarounds.
  
Pause and Resume take a little bit of time, making the code unreliable, as I sometimes pick 2/20 seconds instead of 1/20, and I have no reliability that the loop is actually running every 20 seconds (might be 18 might be 25).

  


• Does the Remote I/O audio unit set the number of channels in the buffer ?

  25 novembre 2013, par awfulcode

  I'm using kxmovie (it's a ffmpeg-based video player) as a base for an app and I'm trying to figure out how the RemoteI/O unit works on iOS when the only thing connected to a device is headphones and we're playing a track with more than 2 channels (say a surround 6 track channel). It seems like it is going with the output channel setting and the buffer only has 2 channels. Is this because of Core Audio's pull structure ? And if so, what's happening to the other channels in the track ? Are they being downmixed or simply ignored ?

  The code for the render callback connected to the remoteio unit is here :

  - (BOOL) renderFrames: (UInt32) numFrames
  
                 ioData: (AudioBufferList *) ioData
  
  {
  
      NSLog(@"Number of channels in buffer: %lu",ioData->mNumberBuffers);
  
  
  
      for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
  
          memset(ioData->mBuffers[iBuffer].mData, 0, ioData->mBuffers[iBuffer].mDataByteSize);
  
      }
  
  
  
  
  
      if (_playing && _outputBlock ) {
  
  
  
          // Collect data to render from the callbacks
  
          _outputBlock(_outData, numFrames, _numOutputChannels);
  
  
  
          // Put the rendered data into the output buffer
  
          if (_numBytesPerSample == 4) // then we've already got floats
  
          {
  
              float zero = 0.0;
  
  
  
              for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
  
  
  
                  int thisNumChannels = ioData->mBuffers[iBuffer].mNumberChannels;
  
  
  
                  for (int iChannel = 0; iChannel < thisNumChannels; ++iChannel) {
  
                      vDSP_vsadd(_outData+iChannel, _numOutputChannels, &zero, (float *)ioData->mBuffers[iBuffer].mData, thisNumChannels, numFrames);
  
                  }
  
              }
  
          }
  
          else if (_numBytesPerSample == 2) // then we need to convert SInt16 -> Float (and also scale)
  
          {
  
              float scale = (float)INT16_MAX;
  
              vDSP_vsmul(_outData, 1, &scale, _outData, 1, numFrames*_numOutputChannels);
  
  
  
              for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
  
  
  
                  int thisNumChannels = ioData->mBuffers[iBuffer].mNumberChannels;
  
  
  
                  for (int iChannel = 0; iChannel < thisNumChannels; ++iChannel) {
  
                      vDSP_vfix16(_outData+iChannel, _numOutputChannels, (SInt16 *)ioData->mBuffers[iBuffer].mData+iChannel, thisNumChannels, numFrames);
  
                  }
  
              }
  
  
  
          }        
  
      }
  
  
  
      return noErr;
  
  }

  Thanks !

  edit : Here's the code for the ASBD (_ouputFormat). It's getting its values straight from the remoteio. You can also check the whole method file here.

  if (checkError(AudioUnitGetProperty(_audioUnit,
  
                                      kAudioUnitProperty_StreamFormat,
  
                                      kAudioUnitScope_Input,
  
                                      0,
  
                                      &_outputFormat,
  
                                      &size),
  
                 "Couldn't get the hardware output stream format"))
  
      return NO;
  
  
  
  
  
  _outputFormat.mSampleRate = _samplingRate;
  
  if (checkError(AudioUnitSetProperty(_audioUnit,
  
                                      kAudioUnitProperty_StreamFormat,
  
                                      kAudioUnitScope_Input,
  
                                      0,
  
                                      &_outputFormat,
  
                                      size),
  
                 "Couldn't set the hardware output stream format")) {
  
  
  
      // just warning
  
  }
  
  
  
  _numBytesPerSample = _outputFormat.mBitsPerChannel / 8;
  
  _numOutputChannels = _outputFormat.mChannelsPerFrame;
  
  
  
  NSLog(@"Current output bytes per sample: %ld", _numBytesPerSample);
  
  NSLog(@"Current output num channels: %ld", _numOutputChannels);
  
  
  
  // Slap a render callback on the unit
  
  AURenderCallbackStruct callbackStruct;
  
  callbackStruct.inputProc = renderCallback;
  
  callbackStruct.inputProcRefCon = (__bridge void *)(self);
  
  
  
  if (checkError(AudioUnitSetProperty(_audioUnit,
  
                                      kAudioUnitProperty_SetRenderCallback,
  
                                      kAudioUnitScope_Input,
  
                                      0,
  
                                      &callbackStruct,
  
                                      sizeof(callbackStruct)),
  
                 "Couldn't set the render callback on the audio unit"))
  
      return NO;

• Audio clicks / cracklings with ffmpeg audio recording on mac os

  15 février 2023, par Specimen

  I'm having audio clicks on my ffmpeg audio recordings. If I record with OBS for example, the audio comes out just fine. This is what I put in the terminal :

  


  ffmpeg -f avfoundation -i ":0" test.mp3


  


  where 0 is my Soundflower audio device, which I found using ffmpeg -f avfoundation -list_devices true -i "", which returns :

  


  AVFoundation audio devices:
[0] Soundflower (2ch)


  


  Soundflower, my speakers, and the ffmpeg recording are all set at 48kHz.
There is another thread that states that this may be an issue with ffmpeg version 4.3, and to try to downgrade to 4.2 ; I tried to google how to downgrade on brew, but didn't find anything.