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• How to save/encode recorded raw PCM Data as AAC/MP4 format file in Android

  28 janvier 2015, par INVISIBLE

  i want to save recorder pcm data as aac/mp4 format file.
  i am using AudioRecord class for recording audio in android. i have success fully saved it as wave file by adding a wave header to raw data. but dont know how to save it as aac/mp4 file, because aac/mp4 format is compressed then wave.
  the methods i am using for saving pcm data as wave is pasted below.

  recorder = new AudioRecord(MediaRecorder.AudioSource.MIC,
  
              SavedSampleRate, RECORDER_CHANNELS, RECORDER_AUDIO_ENCODING,
  
              bufferSize);
  
  recorder.startRecording();
  
  isRecording = true;
  
  
  
  isRecording = true;
  
  
  
  recordingThread = new Thread(new Runnable() {
  
      @Override
  
      public void run() {
  
          writeAudioDataToFile();
  
      }
  
  }, "AudioRecorder Thread");
  
  
  
  recordingThread.start();

  second

  private void writeAudioDataToFile() {
  
  
  
      byte data[] = new byte[bufferSize];
  
      // short sData[] = new short[bufferSize];
  
      String filename = getTempFilename();
  
      FileOutputStream os = null;
  
  
  
      try {
  
          os = new FileOutputStream(filename);
  
      } catch (Exception e) {
  
          e.printStackTrace();
  
      }
  
  
  
      int read = 0;
  
  
  
      if (null != os) {
  
          while (isRecording) {
  
              double sum = 0;
  
              read = recorder.read(data, 0, bufferSize);
  
  
  
              if (AudioRecord.ERROR_INVALID_OPERATION != read) {
  
                  try {
  
  
  
                      synchronized (this) {
  
  
  
  
  
                          // Necessary in order to convert negative shorts!
  
                          final int USHORT_MASK = (1 << 16) - 1;
  
  
  
                          final ByteBuffer buf = ByteBuffer.wrap(data).order(
  
                                  ByteOrder.LITTLE_ENDIAN);
  
  
  
                          final ByteBuffer newBuf = ByteBuffer.allocate(
  
                                  data.length).order(ByteOrder.LITTLE_ENDIAN);
  
  
  
                          int sample;
  
  
  
                          while (buf.hasRemaining()) {
  
  
  
  
  
  
  
                              short shortSample = buf.getShort();
  
                              sample = (int) shortSample & USHORT_MASK;
  
  
  
  
  
  
  
                              sample = sample * db_value_global;
  
                              sample = mRmsFilterSetting.filter
  
                                      .apply((((int) 0) | shortSample)
  
                                              * db_value_global);
  
  
  
  
  
  
  
                              newBuf.putShort((short) sample);
  
                          }
  
  
  
                          data = newBuf.array();
  
  
  
                          os.write(data);
  
  
  
  
  
  
  
  
  
  
  
                      }
  
  
  
                  } catch (Exception e) {
  
                      e.printStackTrace();
  
                  }
  
              }
  
          }
  
  
  
          try {
  
              os.close();
  
          } catch (Exception e) {
  
              e.printStackTrace();
  
          }
  
      }
  
  }

  and finally saving it as

  private void copyWaveFile(ArrayList<string> inFilename, String outFilename) {
  
      FileInputStream[] in = null;
  
      FileOutputStream out = null;
  
      long totalAudioLen = 0;
  
      long totalDataLen = totalAudioLen + 36;
  
      long longSampleRate = SavedSampleRate;
  
      int channels = 2;
  
      long byteRate = RECORDER_BPP * SavedSampleRate * channels / 8;
  
  
  
      byte[] data = new byte[bufferSize];
  
  
  
      try {
  
          out = new FileOutputStream(outFilename);
  
  
  
          in = new FileInputStream[inFilename.size()];
  
  
  
          for (int i = 0; i &lt; in.length; i++) {
  
              in[i] = new FileInputStream(inFilename.get(i));
  
              totalAudioLen += in[i].getChannel().size();
  
          }
  
  
  
          totalDataLen = totalAudioLen + 36;
  
  
  
          WriteWaveFileHeader(out, totalAudioLen, totalDataLen,
  
                  longSampleRate, channels, byteRate);
  
  
  
          for (int i = 0; i &lt; in.length; i++) {
  
              while (in[i].read(data) != -1) {
  
                  out.write(data);
  
              }
  
  
  
              in[i].close();
  
          }
  
  
  
          out.close();
  
      } catch (Exception e) {
  
          e.printStackTrace();
  
      } 
  
  }
  
  
  
  
  
  
  
  private void WriteWaveFileHeader(FileOutputStream out, long totalAudioLen,
  
          long totalDataLen, long longSampleRate, int channels, long byteRate)
  
          throws IOException {
  
  
  
      byte[] header = new byte[44];
  
  
  
      header[0] = 'R'; // RIFF/WAVE header
  
      header[1] = 'I';
  
      header[2] = 'F';
  
      header[3] = 'F';
  
      header[4] = (byte) (totalDataLen &amp; 0xff);
  
      header[5] = (byte) ((totalDataLen >> 8) &amp; 0xff);
  
      header[6] = (byte) ((totalDataLen >> 16) &amp; 0xff);
  
      header[7] = (byte) ((totalDataLen >> 24) &amp; 0xff);
  
      header[8] = 'W';
  
      header[9] = 'A';
  
      header[10] = 'V';
  
      header[11] = 'E';
  
      header[12] = 'f'; // 'fmt ' chunk
  
      header[13] = 'm';
  
      header[14] = 't';
  
      header[15] = ' ';
  
      header[16] = 16; // 4 bytes: size of 'fmt ' chunk
  
      header[17] = 0;
  
      header[18] = 0;
  
      header[19] = 0;
  
      header[20] = 1; // format = 1
  
      header[21] = 0;
  
      header[22] = (byte) channels;
  
      header[23] = 0;
  
      header[24] = (byte) (longSampleRate &amp; 0xff);
  
      header[25] = (byte) ((longSampleRate >> 8) &amp; 0xff);
  
      header[26] = (byte) ((longSampleRate >> 16) &amp; 0xff);
  
      header[27] = (byte) ((longSampleRate >> 24) &amp; 0xff);
  
      header[28] = (byte) (byteRate &amp; 0xff);
  
      header[29] = (byte) ((byteRate >> 8) &amp; 0xff);
  
      header[30] = (byte) ((byteRate >> 16) &amp; 0xff);
  
      header[31] = (byte) ((byteRate >> 24) &amp; 0xff);
  
      header[32] = (byte) (2 * 16 / 8); // block align
  
      header[33] = 0;
  
      header[34] = RECORDER_BPP; // bits per sample
  
      header[35] = 0;
  
      header[36] = 'd';
  
      header[37] = 'a';
  
      header[38] = 't';
  
      header[39] = 'a';
  
      header[40] = (byte) (totalAudioLen &amp; 0xff);
  
      header[41] = (byte) ((totalAudioLen >> 8) &amp; 0xff);
  
      header[42] = (byte) ((totalAudioLen >> 16) &amp; 0xff);
  
      header[43] = (byte) ((totalAudioLen >> 24) &amp; 0xff);
  
  
  
      out.write(header, 0, 44);
  
  }
  
  </string>

  in this piece of code i am recording small PCM files with AudioRecord and saving them as wave file by adding wave header.

  is there any step by step tutorial for how to save pcm data as mp4/aac file.

  thanks in advance.

• Redux : how to add subtitles to mp4 using ffmpeg ?

  25 novembre 2015, par stachyra

  I have a tiny little example mp4 video (which unfortunately stack overflow won’t allow me to embed directly inside my question). Following the advice previously offered in this answer, I’d like to add a subtitle to the video, which will flash on the screen for 2 seconds, using an .srt file that looks like this :

  1
  
  00:00:02,000 --> 00:00:04,000
  
  Test subtitle

  As provided in the link above, I issue a command that looks like this :

  ffmpeg -i small.mp4 -f srt -i small.srt -c:v copy -c:a copy -c:s mov_text small_subtitles.mp4 >& err.log

  And the err.log file looks like this :

  ffmpeg version N-76950-g401c93d-tessus Copyright (c) 2000-2015 the FFmpeg developers
  
  built with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn)
  
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --as=yasm --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzmq --enable-version3 --disable-ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb
  
    libavutil      55.  7.100 / 55.  7.100
  
    libavcodec     57. 15.100 / 57. 15.100
  
    libavformat    57. 17.100 / 57. 17.100
  
    libavdevice    57.  0.100 / 57.  0.100
  
    libavfilter     6. 15.100 /  6. 15.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'small.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf56.40.101
  
    Duration: 00:00:10.02, start: 0.000000, bitrate: 23 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 188x112 [SAR 1:1 DAR 47:28], 17 kb/s, 60 fps, 60 tbr, 15360 tbn, 120 tbc (default)
  
  Metadata:
  
    handler_name    : VideoHandler
  
  Input #1, srt, from 'small.srt':
  
    Duration: N/A, bitrate: N/A
  
  Stream #1:0: Subtitle: subrip
  
  [mp4 @ 0x7f8492848000] Codec for stream 0 does not use global headers but container format requires global headers
  
  Output #0, mp4, to 'small_subtitles.mp4':
  
    Metadata:
  
  major_brand     : isom
  
  minor_version   : 512
  
  compatible_brands: isomiso2avc1mp41
  
  encoder         : Lavf57.17.100
  
  Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 188x112 [SAR 1:1 DAR 47:28], q=2-31, 17 kb/s, 60 fps, 60 tbr, 15360 tbn, 15360 tbc (default)
  
  Metadata:
  
    handler_name    : VideoHandler
  
  Stream #0:1: Subtitle: mov_text ([8][0][0][0] / 0x0008) (default)
  
  Metadata:
  
    encoder         : Lavc57.15.100 mov_text
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #1:0 -> #0:1 (subrip (srt) -> mov_text (native))
  
  Press [q] to stop, [?] for help
  
  frame=  601 fps=0.0 q=-1.0 Lsize=      29kB time=00:00:09.96 bitrate=  24.0kbits/s    
  
  video:21kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 39.561928%

  The resulting output, small_subtitles.mp4, does not seem to have the expected subtitle present.

  Furthermore, the error message,

  [mp4 @ 0x7f8492848000] Codec for stream 0 does not use global headers but container format requires global headers

  is highlighted in bold colors when I run this at the command line without capturing the err.log file (which makes me think it must be a significant message), although I am unable to reproduce that special print formatting easily here.

  I’ve also tried this method, but with the .avi replaced by .mp4, and it seems to just fail silently—no obvious error message, but also no subtitle in the resulting video either.

  Anybody have an idea how to fix this ? In particular, what does the business about global headers mean ?

  I’m working on OS X and am open to hearing answers that would use other open source tools as well.

• How to detect two identical audio/video files with different volume level ?

  3 juillet 2017, par Marina Rappoport

  I’m working on program that could compare 2 video files and show difference.
  I compare audio track of files using SOX and FFMPEG :

  1. invert one of the files (sox)
  2. merge other file and invert version of first (sox)
  3. detect silence (ffmpeg)

  But if two file differs only by volume level - all audio track will be detected as non-silent ranges.

  How to understand that 2 files have the same audio track, but with different volume level ?

  I tried to change sound level via sox : sox -v 1.1 input.wav output.wav
  And then compare statistical information (-n stat).
  It works fine. Result of division parameters audio2/audio1 :

      Samples read 1.00;
  
      Length (seconds) 1.00;
  
      Scaled by 1.00;
  
      Maximum amplitude 1.10;
  
      Minimum amplitude 1.10;
  
      Midline amplitude 1.10;
  
      Mean norm 1.10;
  
      Mean amplitude 1.00;
  
      RMS amplitude 1.10;
  
      Maximum delta 1.10;
  
      Mean delta 1.10;
  
      RMS delta 1.10;
  
      Rough frequency 1.00;
  
      Volume adjustment 1/1.10;

  BUT ! When I tried ffmpeg to change volume of video : ffmpeg -i input.mp4 -vcodec copy -af "volume=10dB" output.mp4 (or volume=volume=0.5) and than compared sox audio statistic : I can’t find any patterns...

  Samples read    1.00
  
  Length (seconds)    1.00
  
  Scaled by   1.00
  
  Maximum amplitude   0.71
  
  Minimum amplitude   0.64
  
  Midline amplitude   -2401.73
  
  Mean    norm    0.34
  
  Mean    amplitude   0.50
  
  RMS     amplitude   0.36
  
  Maximum delta   0.37
  
  Mean    delta   0.34
  
  RMS     delta   0.36
  
  Rough   frequency   0.99
  
  Volume adjustment   0.71

  I will be grateful for any ideas and help.