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• find the timestamp of a sound sample of an mp3 with linux or python

  23 juin 2020, par cardamom

  I am slowly working on a project which where it would be very useful if the computer could find where in an mp3 file a certain sample occurs. I would restrict this problem to meaning a fairly exact snippet of the audio, not just for example the chorus in a song on a different recording by the same band where it would become more some kind of machine learning problem. Am thinking if it has no noise added and comes from the same file, it should somehow be possible to locate the time at which it occurs without machine learning, just like grep can find the lines in a textfile where a word occurs.

  


  In case you don't have an mp3 lying around, can set up the problem with some music available on the net which is in the public domain, so nobody complains :

  


  curl https://web.archive.org/web/20041019004300/http://www.navyband.navy.mil/anthems/ANTHEMS/United%20Kingdom.mp3 --output godsavethequeen.mp3


  


  It's a minute long :

  


  exiftool godsavethequeen.mp3 | grep Duration
Duration                        : 0:01:03 (approx)


  


  Now cut out a bit between 30 and 33 seconds (the bit which goes la la la la..) :

  


  ffmpeg -ss 30 -to 33 -i godsavethequeen.mp3 gstq_sample.mp3


  


  both files in the folder :

  


  $ ls -la
-rw-r--r-- 1 cardamom cardamom   48736 Jun 23 00:08 gstq_sample.mp3
-rw-r--r-- 1 cardamom cardamom 1007055 Jun 22 23:57 godsavethequeen.mp3


  


  This is what am after :

  


  $ findsoundsample gstq_sample.mp3 godsavethequeen.mp3
start 30 end 33


  


  Am happy if it is a bash script or a python solution, even using some kind of python library. Sometimes if you use the wrong tool, the solution might work but look horrible, so whichever tool is more suitable. This is a one minute mp3, have not thought yet about performance just about getting it done at all, but would like some scalability, eg find ten seconds somewhere in half an hour.

  


• ffmpeg works at command line but I got this error when I use it in nodejs app :

  15 avril 2023, par Ibrahim Ashour

  This is the code :

  


  import fs from 'fs';
import googleTTS from 'google-tts-api';
import https from 'https';
import { exec } from 'child_process';

async function convertTextToVoice(text, fileName) {
  const outputFile = `./audios/${fileName}.mp3`;

  const words = text.split(' ');
  const chunks = [];
  let chunk = '';

  for (const word of words) {
    if (chunk.length + word.length < 200) {
      chunk += ' ' + word;
    } else {
      chunks.push(chunk.trim());
      chunk = word;
    }
  }

  chunks.push(chunk.trim());

  for (let i = 0; i < chunks.length; i++) {
    const chunkFile = `./audios/${fileName}-chunk${i}.mp3`;
    const url = await googleTTS.getAudioUrl(chunks[i], {
      lang: 'ar',
      slow: false,
      host: 'https://translate.google.com',
    });


    try {
      const file = await fs.createWriteStream(chunkFile);
      https.get(url, response => {
        response.pipe(file);
        file.on('finish', () => {
          file.close();
          console.log(`Chunk ${i} created successfully`);
        });
      }).on('error', error => {
        console.error(`Error downloading chunk ${i}: ${error}`);
      });
    } catch (error) {
      console.error(error);
    }
  }

  const chunkFiles = chunks.map((chunk, i) => `./audios/${fileName}-chunk${i}.mp3`);

  // Use ffmpeg to merge the audio files
  const concatCommand = `ffmpeg -i "concat:${chunkFiles.join('|')}" -acodec copy ${outputFile}`;
  try {
    await new Promise((resolve, reject) => {
      exec(concatCommand, (error, stdout, stderr) => {
        if (error) {
          console.error(`Error executing ffmpeg command: ${error}`);
          reject(error);
        } else {
          console.log(`Audio file ${outputFile} created successfully`);
          resolve();
        }
      });
    });

    for (const chunkFile of chunkFiles) {
      await fs.promises.unlink(chunkFile);
      console.log(`${chunkFile} deleted successfully`);
    }
  } catch (error) {
    console.error(error);
  }
}


  


  This is the Error :
./audios/تجربة-chunk0.mp3|./audios/تجربة-chunk1.mp3
ffmpeg -i "concat :./audios/تجربة-chunk0.mp3|./audios/تجربة-chunk1.mp3" -acodec copy ./audios/تجربة.mp3
Error executing ffmpeg command : Error : Command failed : ffmpeg -i "concat :./audios/تجربة-chunk0.mp3|./audios/تجربة-chunk1.mp3" -acodec copy ./audios/تجربة.mp3
ffmpeg version 2023-02-04-git-bdc76f467f-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers
built with gcc 12.2.0 (Rev10, Built by MSYS2 project)
configuration : —enable-gpl —enable-version3 —enable-static —disable-w32threads —disable-autodetect —enable-fontconfig —enable-iconv —enable-gnutls —enable-libxml2 —enable-gmp —enable-bzlib —enable-lzma —enable-libsnappy —enable-zlib —enable-librist —enable-libsrt —enable-libssh —enable-libzmq —enable-avisynth —enable-libbluray —enable-libcaca —enable-sdl2 —enable-libaribb24 —enable-libdav1d —enable-libdavs2 —enable-libuavs3d —enable-libzvbi —enable-librav1e —enable-libsvtav1 —enable-libwebp —enable-libx264 —enable-libx265 —enable-libxavs2 —enable-libxvid —enable-libaom —enable-libjxl —enable-libopenjpeg —enable-libvpx —enable-mediafoundation —enable-libass —enable-frei0r —enable-libfreetype —enable-libfribidi —enable-liblensfun —enable-libvidstab —enable-libvmaf —enable-libzimg —enable-amf —enable-cuda-llvm —enable-cuvid —enable-ffnvcodec —enable-nvdec —enable-nvenc —enable-d3d11va —enable-dxva2 —enable-libvpl —enable-libshaderc —enable-vulkan —enable-libplacebo —enable-opencl —enable-libcdio —enable-libgme —enable-libmodplug —enable-libopenmpt —enable-libopencore-amrwb —enable-libmp3lame —enable-libshine —enable-libtheora —enable-libtwolame —enable-libvo-amrwbenc —enable-libilbc —enable-libgsm —enable-libopencore-amrnb —enable-libopus —enable-libspeex —enable-libvorbis —enable-ladspa —enable-libbs2b —enable-libflite —enable-libmysofa —enable-librubberband —enable-libsoxr —enable-chromaprint
libavutil 57. 44.100 / 57. 44.100
libavcodec 59. 63.100 / 59. 63.100
libavformat 59. 38.100 / 59. 38.100
libavdevice 59. 8.101 / 59. 8.101
libavfilter 8. 56.100 / 8. 56.100
libswscale 6. 8.112 / 6. 8.112
libswresample 4. 9.100 / 4. 9.100
libpostproc 56. 7.100 / 56. 7.100
[mp3 @ 00000177df48a1c0] Format mp3 detected only with low score of 1, misdetection possible !
[mp3 @ 00000177df48a1c0] Failed to read frame size : Could not seek to 1026.
concat :./audios/تجربة-chunk0.mp3|./audios/تجربة-chunk1.mp3 : Invalid argument

  


  Error : Command failed : ffmpeg -i "concat :./audios/تجربة-chunk0.mp3|./audios/تجربة-chunk1.mp3" -acodec copy ./audios/تجربة.mp3
ffmpeg version 2023-02-04-git-bdc76f467f-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers
built with gcc 12.2.0 (Rev10, Built by MSYS2 project)
configuration : —enable-gpl —enable-version3 —enable-static —disable-w32threads —disable-autodetect —enable-fontconfig —enable-iconv —enable-gnutls —enable-libxml2 —enable-gmp —enable-bzlib —enable-lzma —enable-libsnappy —enable-zlib —enable-librist —enable-libsrt —enable-libssh —enable-libzmq —enable-avisynth —enable-libbluray —enable-libcaca —enable-sdl2 —enable-libaribb24 —enable-libdav1d —enable-libdavs2 —enable-libuavs3d —enable-libzvbi —enable-librav1e —enable-libsvtav1 —enable-libwebp —enable-libx264 —enable-libx265 —enable-libxavs2 —enable-libxvid —enable-libaom —enable-libjxl —enable-libopenjpeg —enable-libvpx —enable-mediafoundation —enable-libass —enable-frei0r —enable-libfreetype —enable-libfribidi —enable-liblensfun —enable-libvidstab —enable-libvmaf —enable-libzimg —enable-amf —enable-cuda-llvm —enable-cuvid —enable-ffnvcodec —enable-nvdec —enable-nvenc —enable-d3d11va —enable-dxva2 —enable-libvpl —enable-libshaderc —enable-vulkan —enable-libplacebo —enable-opencl —enable-libcdio —enable-libgme —enable-libmodplug —enable-libopenmpt —enable-libopencore-amrwb —enable-libmp3lame —enable-libshine —enable-libtheora —enable-libtwolame —enable-libvo-amrwbenc —enable-libilbc —enable-libgsm —enable-libopencore-amrnb —enable-libopus —enable-libspeex —enable-libvorbis —enable-ladspa —enable-libbs2b —enable-libflite —enable-libmysofa —enable-librubberband —enable-libsoxr —enable-chromaprint
libavutil 57. 44.100 / 57. 44.100
libavcodec 59. 63.100 / 59. 63.100
libavformat 59. 38.100 / 59. 38.100
libavdevice 59. 8.101 / 59. 8.101
libavfilter 8. 56.100 / 8. 56.100
libswscale 6. 8.112 / 6. 8.112
libswresample 4. 9.100 / 4. 9.100
libpostproc 56. 7.100 / 56. 7.100
[mp3 @ 00000177df48a1c0] Format mp3 detected only with low score of 1, misdetection possible !
[mp3 @ 00000177df48a1c0] Failed to read frame size : Could not seek to 1026.
concat :./audios/تجربة-chunk0.mp3|./audios/تجربة-chunk1.mp3 : Invalid argument

  


  at ChildProcess.exithandler (node:child_process:419:12)
at ChildProcess.emit (node:events:513:28)
at maybeClose (node:internal/child_process:1091:16)
at ChildProcess._handle.onexit (node:internal/child_process:302:5) {


  


  code : 1,
killed : false,
signal : null,
cmd : 'ffmpeg -i "concat :./audios/تجربة-chunk0.mp3|./audios/تجربة-chunk1.mp3" -acodec copy ./audios/تجربة.mp3'
}
Chunk 1 created successfully
Chunk 0 created successfully

  


  I expected the command to work in the nodejs app like in the command line

  


• Synchronize video subtitle with text-to-speech voice

  8 décembre 2015, par Ahmad

  I try to create a video of a text in which the text is narrated by text-to-speech.

  To create the video file, I use the VideoFileWriter of Aforge.Net as the following :

  VideoWriter = new VideoFileWriter();
  
  
  
  VideoWriter.Open(CurVideoFile, (int)(Properties.Settings.Default.VideoWidth),
  
      (int)(Properties.Settings.Default.VideoHeight), 25, VideoCodec.MPEG4, 800000);

  To read aloud the text I use SpeechSynthesizer class and write the output to a wave stream

  AudioStream = new FileStream(CurAudioFile, FileMode.Create);
  
  synth.SetOutputToWaveStream(AudioStream);

  I want to highlight the word is spoken in the video, so I synchronize them by the SpeakProgress event :

      void synth_SpeakProgress(object sender, SpeakProgressEventArgs e)
  
      {
  
  
  
          curAuidoPosition = e.AudioPosition;
  
          using (Graphics g = Graphics.FromImage(Screen))
  
          {
  
               g.DrawString(e.Text,....); 
  
          }                    
  
          VideoWriter.WriteVideoFrame(Screen, curAuidoPosition);
  
      }

  And finally, I merge the video and audio using ffmpeg

  using (Process process = new Process())
  
  {
  
            process.StartInfo.FileName = exe_path;
  
            process.StartInfo.Arguments = string.Format(@"-i ""{0}"" -i ""{1}"" -y -acodec copy -vcodec copy ""{2}""",
  
                                             avi_path, mp3_path, output_file);
  
  ......

  The problem is that for some voices like Microsoft Hazel, Zira and David, the video is not synchronized with the audio, and the audio is much faster than the shown subtitle. In windows 7, it works for Mircrosoft Sam

  How can I synchronize them so that it works for any text-to-speech voices ?