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• Audio recorded with MediaRecorder on Chrome missing duration

  3 juin 2017, par suppp111

  I am recording audio (oga/vorbis) files with MediaRecorder. When I record these file through Chrome I get problems : I cannot edit the files on ffmpeg and when I try to play them on Firefox it says they are corrupt (they do play fine on Chrome though).

  Looking at their metadata on ffmpeg I get this :

  Input #0, matroska,webm, from '91.oga':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
  
  [STREAM]
  
  index=0
  
  codec_name=opus
  
  codec_long_name=Opus (Opus Interactive Audio Codec)
  
  profile=unknown
  
  codec_type=audio
  
  codec_time_base=1/48000
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  sample_fmt=fltp
  
  sample_rate=48000
  
  channels=1
  
  channel_layout=mono
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=0/0
  
  time_base=1/1000
  
  start_pts=0
  
  start_time=0.000000
  
  duration_ts=N/A
  
  duration=N/A
  
  bit_rate=N/A
  
  max_bit_rate=N/A
  
  bits_per_raw_sample=N/A
  
  nb_frames=N/A
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=1
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  TAG:language=eng
  
  [/STREAM]
  
  [FORMAT]
  
  filename=91.oga
  
  nb_streams=1
  
  nb_programs=0
  
  format_name=matroska,webm
  
  format_long_name=Matroska / WebM
  
  start_time=0.000000
  
  duration=N/A
  
  size=7195
  
  bit_rate=N/A
  
  probe_score=100
  
  TAG:encoder=Chrome

  As you can see there are problems with the duration. I have looked at posts like this :
  How can I add predefined length to audio recorded from MediaRecorder in Chrome ?

  But even trying that, I got errors when trying to chop and merge files.For example when running :

  ffmpeg -f concat  -i 89_inputs.txt -c copy final.oga

  I get a lot of this :

  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57612, current: 1980; changing to 57613. This may result in incorrect timestamps in the output file.
  
  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57613, current: 2041; changing to 57614. This may result in incorrect timestamps in the output file.
  
  DTS -442721849179034176, next:42521 st:0 invalid dropping
  
  PTS -442721849179034176, next:42521 invalid dropping st:0
  
  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57614, current: 2041; changing to 57615. This may result in incorrect timestamps in the output file.
  
  [oga @ 00000000006789c0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
  
  DTS -442721849179031296, next:42521 st:0 invalid dropping
  
  PTS -442721849179031296, next:42521 invalid dropping st:0

  Does anyone know what we need to do to audio files recorded from Chrome for them to be useful ? Or is there a problem with my setup ?

  Recorder js :

  if (navigator.getUserMedia) {
  
    console.log('getUserMedia supported.');
  
  
  
    var constraints = { audio: true };
  
    var chunks = [];
  
  
  
    var onSuccess = function(stream) {
  
      var mediaRecorder = new MediaRecorder(stream);
  
  
  
      record.onclick = function() {
  
        mediaRecorder.start();
  
        console.log(mediaRecorder.state);
  
        console.log("recorder started");
  
        record.style.background = "red";
  
  
  
        stop.disabled = false;
  
        record.disabled = true;
  
  
  
        var aud = document.getElementById("audioClip");
  
        start = aud.currentTime;
  
      }
  
  
  
      stop.onclick = function() {
  
        console.log(mediaRecorder.state);
  
        console.log("Recording request sent.");
  
        mediaRecorder.stop();
  
      }
  
  
  
      mediaRecorder.onstop = function(e) {
  
        console.log("data available after MediaRecorder.stop() called.");
  
  
  
        var audio = document.createElement('audio');
  
        audio.setAttribute('controls', '');
  
        audio.setAttribute('id', 'audioClip');
  
  
  
        audio.controls = true;
  
        var blob = new Blob(chunks, { 'type' : 'audio/ogg; codecs="vorbis"' });
  
        chunks = [];
  
        var audioURL = window.URL.createObjectURL(blob);
  
        audio.src = audioURL;
  
  
  
        sendRecToPost(blob);   // this just send the audio blob to the server by post
  
        console.log("recorder stopped");
  
  
  
      }

• How to modify music pitch in audio of video file and still sound natural

  9 janvier 2020, par Keith Bennett

  I have some karaoke .mp4 video files (legally obtained) for Thai songs, and want to convert the pitch downward to fit my singing range. I’ve gotten most of the way there thanks to https://superuser.com/questions/292833/how-to-change-audio-frequency/1076762#1076762
  using a command line like this :

  ffmpeg -i in.mp4 -af 'asetrate=35280.0,atempo=1.25' out.mp4

  ...but the instruments and human singing voices don’t sound natural at the modified pitch.

  Is there a better way to change the pitch ? I know some commercial products can do this.

  By the way, I wrote a Ruby script to simplify this ffmpeg call ; it’s at https://gist.github.com/keithrbennett/9ba7043792bfb2fcc92d615076a8413f. It enables you to specify a single factor, and modifies both pitch and tempo accordingly.

• Mix 2 mp3 files with FFmpeg for Android

  24 mai 2017, par Thankgod Richard

  I am developing something similar to a karaoke app. I have a sound file(beat) and a recorded file(voice) from android mediarecorder. I am trying to mix the two files to become one file that plays the same time. I have set up ffmpeg provided by writingminds and everything is working fine. I am not familiar with ffmpeg commands so searched for and got few commands that can achieve the task for me. This is the code i got :

    String files = "-i " + currentFile + " -i " + someFile.getAbsolutePath();
  
              String output = mFileName + "/recorded/test.mp3";
  
              String cmd = "ffmpeg "+files+
  
                      " -filter_complex \"aevalsrc=0:d=10[s1];[s1][1:a]concat=n=2:v=0:a=1[ac1];[0:a][ac1]amix[aout]\" -map [aout] -c:a libmp3lame " + output;

  But it not working. this is the error i got :

  04-01 12:09:26.598 21300-21300/com.shixels.thankgodrichard.mixer I/fpeg: ffmpeg version n3.0.1 Copyright (c) 2000-2016 the FFmpeg developers
  
                                                                             built with gcc 4.8 (GCC)
  
                                                                             configuration: --target-os=linux --cross-prefix=/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/bin/arm-linux-androideabi- --arch=arm --cpu=cortex-a8 --enable-runtime-cpudetect --sysroot=/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/sysroot --enable-pic --enable-libx264 --enable-libass --enable-libfreetype --enable-libfribidi --enable-libmp3lame --enable-fontconfig --enable-pthreads --disable-debug --disable-ffserver --enable-version3 --enable-hardcoded-tables --disable-ffplay --disable-ffprobe --enable-gpl --enable-yasm --disable-doc --disable-shared --enable-static --pkg-config=/home/vagrant/SourceCode/ffmpeg-android/ffmpeg-pkg-config --prefix=/home/vagrant/SourceCode/ffmpeg-android/build/armeabi-v7a --extra-cflags='-I/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/include -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow -fstack-protector-all' --extra-ldflags='-L/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/lib -Wl,-z,relro -Wl,-z,now -pie' --extra-libs='-lpng -lexpat -lm' --extra-cxxflags=
  
                                                                             libavutil      55. 17.103 / 55. 17.103
  
                                                                             libavcodec     57. 24.102 / 57. 24.102
  
                                                                             libavformat    57. 25.100 / 57. 25.100
  
                                                                             libavdevice    57.  0.101 / 57.  0.101
  
                                                                             libavfilter     6. 31.100 /  6. 31.100
  
                                                                             libswscale      4.  0.100 /  4.  0.100
  
                                                                             libswresample   2.  0.101 /  2.  0.101
  
                                                                             libpostproc    54.  0.100 / 54.  0.100
  
                                                                           Output #0, mp3, to 'ffmpeg -i /storage/emulated/0/MyStudio/temps/1491044929409.mp3 -i /storage/emulated/0/temp.mp3 -filter_complex "aevalsrc=0:d=10[s1];[s1][1:a]concat=n=2:v=0:a=1[ac1];[0:a][ac1]amix[aout]" -map [aout] -c:a libmp3lame /storage/emulated/0/recorded/test.mp3':
  
                                                                           Output file #0 does not contain any stream
  
  04-01 12:09:26.599 21300-21300/com.shixels.thankgodrichard.mixer I/fpeg: finished

  I read some answers here saying it depends on the version of the ffmpeg your using. I am using the latest ffmpeg for android by writingminds.