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• FFmpeg - 2 combined video's (splitscreen) removes sound

  10 juillet 2014, par Joey

  I am stuck on this problem where i am trying to put 2 video’s next to each other, like splitscreen. Found the command to do this here on StackOverflow but it removes my sound. I can’t figure out how to keep the sound from the 2 video’s.

  The command i use :

  exec("ffmpeg -i ".$video1." -i ".$video2." -filter_complex '[0:v]pad=iw*2:ih[int];[int][1:v]overlay=W/2:0[vid]' -map [vid] -c:v libx264 -crf 23 /tmp/output_file.flv", $output, $return);

  The output is exactly how i want but i want the two sound streams too.

  EDIT :

  For anyone having the same problem, i ended up doing this in 3 steps :

  # Combine the two audio streams into 1 temp file
  
  exec("ffmpeg -i ".$video1." -i ".$video2." -filter_complex amix=inputs=2:duration=first:dropout_transition=2 ".$outputSound, $output, $return);
  
  
  
  # Set the two video's as splitscreen
  
  exec("ffmpeg -i ".$video1." -i ".$video2." -filter_complex '[0:v]pad=iw*2:ih[int];[int][1:v]overlay=W/2:0[vid]' -map [vid] -c:v libx264 -crf 23 ".$outputVideo, $output, $return);
  
  
  
  # Combine merged audio file with splitscreen video
  
  exec("ffmpeg -i ".$outputVideo." -i ".$outputSound." -map 0 -map 1 -codec copy -shortest ".$outputCombined, $output, $return);

  Solved !

• samplerate conversion function fails to produce an audible sound but only small pieces of audio

  2 juillet 2014, par user3749290

  playmp3() using libmpg123

  if (isPaused==0 && mpg123_read(mh, buffer, buffer_size, &done) == MPG123_OK)
  
  {
  
      char * resBuffer=&buffer[0]; //22100=0,5s
  
      buffer = resample(resBuffer,22050,22050); // I think the result is 1/2 of audio speed
  
      if((ao_play(dev, (char*)buffer, done)==0)){
  
          return 1;
  
  }

  resample() Using avcodec from ffmpeg

  #define LENGTH_MS 500       // how many milliseconds of speech to store 0,5s:x=1:44100 x=22050 sample to store
  
  #define RATE 44100      // the sampling rate (input)
  
  #define FORMAT PA_SAMPLE_S16NE  // sample size: 8 or 16 bits
  
  #define CHANNELS 2      // 1 = mono 2 = stereo
  
  
  
  struct AVResampleContext* audio_cntx = 0;
  
  //(LENGTH_MS*RATE*16*CHANNELS)/8000
  
  
  
      void resample(char in_buffer[],int out_rate,int nsamples,char out_buffer[])
  
      {
  
          //char out_buffer[ sizeof( in_buffer ) * 4];
  
          audio_cntx = av_resample_init( out_rate, //out rate
  
              RATE, //in rate
  
              16, //filter length
  
              10, //phase count
  
              0, //linear FIR filter
  
              1.0 ); //cutoff frequency
  
          assert( audio_cntx && "Failed to create resampling context!");
  
          int samples_consumed;
  
          //*out_buffer = malloc(sizeof(in_buffer));
  
          int samples_output = av_resample( audio_cntx, //resample context
  
              (short*)out_buffer, //buffout
  
              (short*)in_buffer,  //buffin
  
              &samples_consumed,  //&consumed
  
              nsamples,       //nb_samples
  
              sizeof(out_buffer)/2,//lenout sizeof(out_buffer)/2
  
              0);//is_last
  
          assert( samples_output > 0 && "Error calling av_resample()!" );
  
          av_resample_close( audio_cntx );    
  
      }

  When I run this code, the application part, the problem is that I hear the sound jerky, why ?
  The size of the array I think is right, I calculated considering that in the second half should be 22050 samples from store.

• speex decoding make wrong sound (FFmpeg on iOS)

  2 juillet 2014, par user3796700

  I’m trying to use FFmpeg on iOS to play live streams.
  One with NellyMoser, as below : (success)

  avformat_open_input(&formatContext, "rtmp://my/nellymoser/stream/url", NULL, &options);

  Now I tried to play the same stream but encoded in Speex format.
  So I followed some tutorials, compiled "ogg.a, speex.a, speexdsp.a" for iOS ;
  Then re-compiled FFmpeg, linking to those three .a files.

  However the output is wrong :

  avformat_open_input(&formatContext, "rtmp://my/speex/stream/url", NULL, &options); 

  The output sound is 2x faster than normal. It seems like only half of the data is decoded.

  Does anyone have tried similar things before ?
  I’ve stocked several days, really need help here..
  Thanks !

  As for reference, here is how I compile FFmpeg :

  ./configure \
  
  --enable-libspeex \
  
  --disable-doc \
  
  --disable-ffmpeg \
  
  --disable-ffserver \
  
  --enable-cross-compile \
  
  --arch=armv7 \
  
  --target-os=darwin \
  
  --cc=clang \
  
  --as='gas-preprocessor/gas-preprocessor.pl clang' \
  
  --sysroot=/Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS7.1.sdk \
  
  --cpu=cortex-a8 \
  
  --extra-cflags='-arch armv7 -I ../speex/armv7/include -I ../libogg/armv7/include' \
  
  --extra-ldflags='-arch armv7 -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS7.1.sdk' \
  
  --extra-ldflags='-L ../speex/armv7/lib -lspeexdsp -lspeex -L ../libogg/armv7/lib -logg' \
  
  --enable-pic \
  
  --prefix=/Users/chienlo/Desktop/speexLibrary/ffmpeg-2.2.4/armv7