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• Merge video without sound and audio with offset using ffmeg

  17 août 2022, par Stratocaster

  I need to merge the video (which has no audio) and the audio file with offset (audio should start at 9 seconds). I use this :

  


  ffmpeg -i video_without_sound.mp4 -i audio_file.mp3 -c:v copy -filter_complex "[1]adelay=9000|9000[s1];[s0:a][s1]amix=2[a]" -map 0:v -map "[a]" -c:a aac output.mp4


  


  And I get an error :

  


  Stream specifier 's0:a' in filtergraph description [1]adelay=9000|9000[s1];[s0:a][s1]amix=2[a] matches no streams.


  


  I think problem is that the video files have no sound and no audio streams. How to merge such files with offset ?

  


• AVAssetWriter creating mp4 with no sound in last 50msec

  12 août 2015, par Joseph K

  I’m working on a project involving live streaming from the iPhone’s camera.

  To minimize loss during AVAssetWriter finishWriting, I use an array of 2 asset writers and swap them whenever I need to create an mp4 fragment out of the recorded buffers.

  Code responsible for capturing Audio & Video sample buffers

  func captureOutput(captureOutput: AVCaptureOutput!, didOutputSampleBuffer sampleBuffer: CMSampleBuffer!, fromConnection connection: AVCaptureConnection!) {
  
  
  
      if CMSampleBufferDataIsReady(sampleBuffer) <= 0 {
  
              println("Skipped sample because it was not ready")
  
              return
  
          }
  
  
  
      if captureOutput == audioOutput  {
  
  
  
          if audioWriterBuffers[0].readyForMoreMediaData() {
  
              if !writers[0].appendAudio(sampleBuffer) { println("Failed to append: \(recordingPackages[0].name). Error: \(recordingPackages[0].outputWriter.error.localizedDescription)") }
  
              else {
  
                  writtenAudioFrames++
  
  
  
                  if writtenAudioFrames == framesPerFragment {
  
                      writeFragment()
  
                  }
  
              }
  
          }
  
          else {
  
              println("Skipped audio sample; it is not ready.")
  
          }
  
      }
  
  
  
      else if captureOutput == videoOutput {
  
          //Video sample buffer
  
          if videoWriterBuffers[0].readyForMoreMediaData() {
  
              //Call startSessionAtSourceTime if needed
  
              //Append sample buffer with a source time
  
          }
  
      }
  
  }

  Code responsible for the writing and swapping

  func writeFragment() {
  
      writtenAudioFrames = 0
  
  
  
      swap(&writers[0], &writers[1])
  
      if !writers[0].startWriting() {println( "Failed to start OTHER writer writing") }
  
      else { startTime  = CFAbsoluteTimeGetCurrent() }
  
  
  
      audioWriterBuffers[0].markAsFinished()
  
      videoWriterBuffers[0].markAsFinished()
  
  
  
      writers[1].outputWriter.finishWritingWithCompletionHandler { () -> Void in
  
          println("Finish Package record Writing, now Resetting")
  
          //
  
          // Handle written MP4 fragment code 
  
          //
  
  
  
          //Reset Writer
  
          //Basically reallocate it as a new AVAssetWriter with a given URL and MPEG4 file Type and add inputs to it
  
          self.resetWriter()
  
      }

  The issue at hand

  The written MP4 fragments are being sent over to a local sandbox server to be analyzed.

  When MP4 fragments are stitched together using FFMpeg, there is a noticeable glitch in sound due to the fact that there is not audio at the last 50msec of every fragment.

  My audio AVAssetWriterInput’s settings are the following :

  static let audioSettings: [NSObject : AnyObject]! =
  
  [
  
      AVFormatIDKey : NSNumber(integer: kAudioFormatMPEG4AAC),
  
      AVNumberOfChannelsKey : NSNumber(integer: 1),
  
      AVSampleRateKey : NSNumber(int: 44100),
  
      AVEncoderBitRateKey : NSNumber(int: 64000),
  
  ]

  As such, I encode 44 audio sample buffers every second. They are all being successfully appended.

  Further resources

  Here’s a waveform display of the audio stream after concatenating the mp4 fragments

  

   !! Note that my fragments are about 2secs in length.
   !! Note that I’m focusing on audio since video frames are extremely smooth when jumping from one fragment to another.

  Any idea as to what is causing this ? I can provide further code or info if needed.

• FFmpeg transcoded sound (AAC) stops after half video time

  17 août 2015, par TheSHEEEP

  I have a strange problem in my C/C++ FFmpeg transcoder, which takes an input MP4 (varying input codecs) and produces and output MP4 (x264, baseline & AAC LC @44100 sample rate with libfdk_aac) :

  The resulting mp4 video has fine images (x264) and the audio (AAC LC) works fine as well, but is only played until exactly the half of the video.

  The audio is not slowed down, not stretched and doesn’t stutter. It just stops right in the middle of the video.

  One hint may be that the input file has a sample rate of 22050 and 22050/44100 is 0.5, but I really don’t get why this would make the sound just stop after half the time. I’d expect such an error leading to sound being at the wrong speed. Everything works just fine if I don’t try to enforce 44100 and instead just use the incoming sample_rate.

  Another guess would be that the pts calculation doesn’t work. But the audio sounds just fine (until it stops) and I do exactly the same for the video part, where it works flawlessly. "Exactly", as in the same code, but "audio"-variables replaced with "video"-variables.

  FFmpeg reports no errors during the whole process. I also flush the decoders/encoders/interleaved_writing after all the package reading from the input is done. It works well for the video so I doubt there is much wrong with my general approach.

  Here are the functions of my code (stripped off the error handling & other class stuff) :

  AudioCodecContext Setup

  outContext->_audioCodec = avcodec_find_encoder(outContext->_audioTargetCodecID);
  
  outContext->_audioStream = 
  
          avformat_new_stream(outContext->_formatContext, outContext->_audioCodec);
  
  outContext->_audioCodecContext = outContext->_audioStream->codec;
  
  outContext->_audioCodecContext->channels = 2;
  
  outContext->_audioCodecContext->channel_layout = av_get_default_channel_layout(2);
  
  outContext->_audioCodecContext->sample_rate = 44100;
  
  outContext->_audioCodecContext->sample_fmt = outContext->_audioCodec->sample_fmts[0];
  
  outContext->_audioCodecContext->bit_rate = 128000;
  
  outContext->_audioCodecContext->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
  
  outContext->_audioCodecContext->time_base = 
  
          (AVRational){1, outContext->_audioCodecContext->sample_rate};
  
  outContext->_audioStream->time_base = (AVRational){1, outContext->_audioCodecContext->sample_rate};
  
  int retVal = avcodec_open2(outContext->_audioCodecContext, outContext->_audioCodec, NULL);

  Resampler Setup

  outContext->_audioResamplerContext = 
  
          swr_alloc_set_opts( NULL, outContext->_audioCodecContext->channel_layout,
  
                              outContext->_audioCodecContext->sample_fmt,
  
                              outContext->_audioCodecContext->sample_rate,
  
                              _inputContext._audioCodecContext->channel_layout,
  
                              _inputContext._audioCodecContext->sample_fmt,
  
                              _inputContext._audioCodecContext->sample_rate,
  
                              0, NULL);
  
  int retVal = swr_init(outContext->_audioResamplerContext);

  Decoding

  decodedBytes = avcodec_decode_audio4(   _inputContext._audioCodecContext, 
  
                                          _inputContext._audioTempFrame, 
  
                                          &p_gotAudioFrame, &_inputContext._currentPacket);

  Converting (only if decoding produced a frame, of course)

  int retVal = swr_convert(   outContext->_audioResamplerContext, 
  
                              outContext->_audioConvertedFrame->data, 
  
                              outContext->_audioConvertedFrame->nb_samples, 
  
                              (const uint8_t**)_inputContext._audioTempFrame->data, 
  
                              _inputContext._audioTempFrame->nb_samples);

  Encoding (only if decoding produced a frame, of course)

  outContext->_audioConvertedFrame->pts = 
  
          av_frame_get_best_effort_timestamp(_inputContext._audioTempFrame);
  
  
  
  // Init the new packet
  
  av_init_packet(&outContext->_audioPacket);
  
  outContext->_audioPacket.data = NULL;
  
  outContext->_audioPacket.size = 0;
  
  
  
  // Encode
  
  int retVal = avcodec_encode_audio2( outContext->_audioCodecContext, 
  
                                      &outContext->_audioPacket, 
  
                                      outContext->_audioConvertedFrame,
  
                                      &p_gotPacket);
  
  
  
  
  
  // Set pts/dts time stamps for writing interleaved
  
  av_packet_rescale_ts(   &outContext->_audioPacket, 
  
                          outContext->_audioCodecContext->time_base,
  
                          outContext->_audioStream->time_base);
  
  outContext->_audioPacket.stream_index = outContext->_audioStream->index;

  Writing (only if encoding produced a packet, of course)

  int retVal = av_interleaved_write_frame(outContext->_formatContext, &outContext->_audioPacket);

  I am quite out of ideas about what would cause such a behaviour.