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• ffmpeg how to ignore initial empty audio frames when decoding to loop a sound

  1er décembre 2020, par cs guy

  I am trying to loop a ogg sound file. The goal is to make a loopable audio interface for my mobile app.

  


  I decode the given ogg file into a buffer and that buffer is sent to audio card for playing. All good until it the audio finishes (end of file). When it finishes I use av_seek_frame(avFormatContext, streamInfoIndex, 0, AVSEEK_FLAG_FRAME); to basically loop back to beginning. And continue decoding into writing to the same buffer. At first sight I thought this would give me perfect loops. One problem I had was, the decoder in the end gives me extra empty frames. So I ignored them by keeping track of how many samples are decoded :

  


  durationInMillis = avFormatContext->duration * 1000;
numOfTotalSamples =
                (uint64_t) avFormatContext->duration *
                (uint64_t) pLocalCodecParameters->sample_rate *
                (uint64_t) pLocalCodecParameters->channels /
                (uint64_t) AV_TIME_BASE;


  


  When the threshold is reached I ignore the frames sent by the codec. I thought this was it and ran some test. I recorded 5 minutes of my app and in the end I compared the results in FL studio by customly adding the same sound clip several times to match the length of my audio recording :

  


  Here it is after 5 minutes :

  


  

  


  In the first loops the difference is very low I thought it was working and I used this for several days until I tested this on 5 minute recording. As the looping approached to 5 minutes mark the difference got very huge. My code is not looping the audio correctly. I suspect that the codec is adding 1 or 2 empty frames at the very beginning in each loop caused by av_seek_frame knowing that a frame can contain up several audio samples. These probably accumulate and cause the mismatch.

  


  My question is : how can I drop the empty frames that is sent by codec while decoding so that I can create a perfect loop of the audio ?

  


  My code is below here. Please be aware that I deleted lots of if checks that was inteded for safety to make it more readable in the code below, these removed checks are always false so it doesnt matter for the reader.

  


  helper.cpp

  


  int32_t&#xA;outputAudioFrame(AVCodecContext *avCodecContext, AVFrame *avResampledDecFrame, int32_t &amp;ret,&#xA;                 LockFreeQueue<float> *&amp;buffer, int8_t *&amp;mediaLoadPointer,&#xA;                 AVFrame *avDecoderFrame, SwrContext *swrContext,&#xA;                 std::atomic_bool *&amp;signalExitFuture,&#xA;                 uint64_t &amp;currentNumSamples, uint64_t &amp;numOfTotalSamples) {&#xA;    // resampling is done here but its boiler code so I removed it.&#xA;    auto *floatArrPtr = (float *) (avResampledDecFrame->data[0]);&#xA;&#xA;    int32_t numOfSamples = avResampledDecFrame->nb_samples * avResampledDecFrame->channels;&#xA;&#xA;    for (int32_t i = 0; i &lt; numOfSamples; i&#x2B;&#x2B;) {&#xA;        if (currentNumSamples == numOfTotalSamples) {&#xA;            break;&#xA;        }&#xA;&#xA;        buffer->push(*floatArrPtr);&#xA;        currentNumSamples&#x2B;&#x2B;;&#xA;        floatArrPtr&#x2B;&#x2B;;&#xA;    }&#xA;&#xA;    return 0;&#xA;}&#xA;&#xA;&#xA;&#xA;int32_t decode(int32_t &amp;ret, AVCodecContext *avCodecContext, AVPacket *avPacket,&#xA;               LockFreeQueue<float> *&amp;buffer,&#xA;               AVFrame *avDecoderFrame,&#xA;               AVFrame *avResampledDecFrame,&#xA;               std::atomic_bool *&amp;signalExitFuture,&#xA;               int8_t *&amp;mediaLoadPointer, SwrContext *swrContext,&#xA;               uint64_t &amp;currentNumSamples, uint64_t &amp;numOfTotalSamples) {&#xA;   &#xA;    ret = avcodec_send_packet(avCodecContext, avPacket);&#xA;    if (ret &lt; 0) {&#xA;        LOGE("decode: Error submitting a packet for decoding %s", av_err2str(ret));&#xA;        return ret;&#xA;    }&#xA;&#xA;    // get all the available frames from the decoder&#xA;    while (ret >= 0) {&#xA;&#xA;        // submit the packet to the decoder&#xA;        ret = avcodec_receive_frame(avCodecContext, avDecoderFrame);&#xA;        if (ret &lt; 0) {&#xA;            // those two return values are special and mean there is no output&#xA;            // frame available, but there were no errors during decoding&#xA;            if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)) {&#xA;                //LOGD("avcodec_receive_frame returned special %s", av_err2str(ret));&#xA;                return 0;&#xA;            }&#xA;&#xA;            LOGE("avcodec_receive_frame Error during decoding %s", av_err2str(ret));&#xA;            return ret;&#xA;        }&#xA;&#xA;        ret = outputAudioFrame(avCodecContext, avResampledDecFrame, ret, buffer,&#xA;                               mediaLoadPointer, avDecoderFrame, swrContext, signalExitFuture,&#xA;                               currentNumSamples, numOfTotalSamples);&#xA;&#xA;        av_frame_unref(avDecoderFrame);&#xA;        av_frame_unref(avResampledDecFrame);&#xA;&#xA;        if (ret &lt; 0)&#xA;            return ret;&#xA;    }&#xA;&#xA;    return 0;&#xA;}&#xA;</float></float>

  


  Main.cpp

  


  while (!*signalExitFuture) {
            while ((ret = av_read_frame(avFormatContext, avPacket)) >= 0) {

                ret = decode(ret, avCodecContext, avPacket, buffer, avDecoderFrame,
                             avResampledDecFrame, signalExitFuture,
                             mediaLoadPointer, swrContext,
                             currentNumSamples, numOfTotalSamples);

                // The packet must be freed with av_packet_unref() when it is no longer needed.
                av_packet_unref(avPacket);

                if (ret < 0) {
                    LOGE("Error! %s", av_err2str(ret));

                    goto cleanup;
                }
            }

            if (ret == AVERROR_EOF) {

                ret = av_seek_frame(avFormatContext, streamInfoIndex, 0, AVSEEK_FLAG_FRAME);

                currentNumSamples = 0;
                avcodec_flush_buffers(avCodecContext);
            }
        }


  


• opencv c api video writer query

  28 janvier 2015, par monika

  I am working on opencv. I have written a c code using video writer that converts the video which is coming from camera. But, the output that i am getting containing no audio. So my question is how to compress the audio as well. What library i can use for it. I have googled it and came to know opencv_ffmpeg can be used but i dont know how to use. So please can u help me in resolvimg this.

  My code is :

  #include
  
  #include
  
  #include
  
  #include
  
  #include
  
  #include
  
  #include
  
  
  
  int main()
  
  {
  
       int key = 0,j,k,fps,frameH,frameW,height;  
  
       int c=1,d=1;
  
       char buffer[1000];
  
       int count = 0;
  
       IplImage* color=0;
  
       IplImage* ChangedImage=0;
  
       char special[200];
  
       CvCapture* capture=0;
  
        CvVideoWriter* writer;
  
       IplImage* img=0;
  
      //to capture video  
  
      capture = cvCreateCameraCapture(1); 
  
   /* capture = cvCreateFileCapture("F:\Wildlife.wmv");*/
  
     if ( !capture )   
  
          {  
  
          fprintf( stderr, "Cannot open file!\n" ); 
  
     }
  
      img = cvQueryFrame( capture );
  
      frameH = (int) cvGetCaptureProperty(capture, CV_CAP_PROP_FRAME_HEIGHT);
  
      frameW = (int) cvGetCaptureProperty(capture, CV_CAP_PROP_FRAME_WIDTH);
  
      fps = (int) cvGetCaptureProperty(capture, CV_CAP_PROP_FPS);
  
     cvNamedWindow( "video", CV_WINDOW_AUTOSIZE );
  
  // to create a video file     
  
  writer=cvCreateVideoWriter("F:/logitechhighresolutioncompressed.avi",CV_FOURCC('M','J','P','G'),fps,cvSize(frameW, frameH ),1);
  
  if (!writer) {
  
  printf("!!! Failed cvCreateVideoWriter");
  
  return;
  
  }
  
  if( !img )
  
      {
  
            exit;
  
      }
  
      if(img)
  
      {  
  
      /* img = cvQueryFrame( capture );*/
  
       cvShowImage( "video", img );
  
       cvWriteFrame(writer, img);
  
  
  
      /* cvWaitKey(0);*/
  
       /*cvWriteFrame(writer, img);*/
  
       d++;
  
  
  
  
  
       /*if (cvWaitKey(100)== 27) break;*/
  
  
  
  }/*while(img);*/
  
  
  
  
  
     /* mytime1 = time(NULL);
  
      printf(ctime(&mytime1));*/
  
      cvReleaseVideoWriter(&writer);
  
      cvReleaseCapture( &capture ); 
  
      cvDestroyWindow( "video" );
  
      cvWaitKey(0);
  
      return 0; enter code here
  
      system("PAUSE");
  
  
  
  } 

• Embedding Audio File to Video

  19 septembre 2022, par Hamza Naeem

  I had a special .wav file with some encodings in it and i want to insert this into video file. I tried this with moviepy using the following method but it doesn't helped me as the audio get distorted.

  


  import moviepy
from moviepy.editor import VideoFileClip, AudioFileClip, CompositeAudioClip, concatenate_audioclips


if __name__ == "__main__":
audio_file = f'audios/a1.wav'
    video_file = f'videos/v1.mp4'
    audio_clip = AudioFileClip(audio_file)
    video_clip = VideoFileClip(video_file)

    final_audio = CompositeAudioClip([video_clip.audio, audio_clip])
    video_clip.write_videofile(f'results/out.mp4')
    video_clip.close()
    audio_clip.close()


  


  nothing with ffmpeg as :

  


  import ffmpeg

if __name__ == "__main__":
    video = ffmpeg.input(f'videos/v1.mp4').video  # get only video channel
    audio = ffmpeg.input(f'audios/a1.wav').audio  # get only audio channel
    output = ffmpeg.output(video, audio, f'results/output.mp4', vcodec='copy', acodec='aac', strict='experimental')
    ffmpeg.run(output)


  


  although I can extract the perfect mp3 file using pydub, but again inserting this into video distort the audio. it may due to miss match of fps or something else. The pydub stuff is as follows :

  


  import pydub
from pydub import AudioSegment

audio_file = f'audios/a1.wav'
sound = pydub.AudioSegment.from_mp3(audio_file)
sound.export("results/apple.mp3", format="wav")


  


  can anyone here to guide me with the correct method.

  


  Thanks in advance.