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• Capturing audio data (using javascript) and uploading on a server as MP3

  4 septembre 2018, par Michel

  Following a number of resources on the internet, I am trying to build a simple web page, where I can go to record something (my voice), then make a mp3 file out of the recording and finally upload that file to a server.

  At this point I can do the recording and also play back, but I haven’t gone as far as uploading, it seems like I cannot even make an mp3 file locally.
  Can someone tell me what I am doing wrong, or in the wrong order ?

  Below is all the code I have at this point.

      
  
      
  
      
  
      
  
  
  
  
  
  <div>
  
      <h2>Audio record and playback</h2>
  
      <p>
  
          <button></button></p><h3>Start</h3>
  
          <button disabled="disabled"><h3>Stop</h3></button>
  
          <audio controls="controls"></audio>
  
          <a></a>
  
      
  
  </div>
  
  
  
  <code class="echappe-js">&lt;script&gt;<br />
  
    var player = document.getElementById('player');<br />
  
  <br />
  
    var handleSuccess = function(stream) {<br />
  
      rec = new MediaRecorder(stream);<br />
  
  <br />
  
      rec.ondataavailable = e =&gt; {<br />
  
          audioChunks.push(e.data);<br />
  
          if (rec.state == &quot;inactive&quot;) {<br />
  
              let blob = new Blob(audioChunks,{type:'audio/x-mpeg-3'});<br />
  
              player.src = URL.createObjectURL(blob);<br />
  
              player.controls=true;<br />
  
              player.autoplay=true;<br />
  
              // audioDownload.href = player.src;<br />
  
              // audioDownload.download = 'sound.data';<br />
  
              // audioDownload.innerHTML = 'Download';<br />
  
              mp3Build();<br />
  
          }<br />
  
      }<br />
  
  <br />
  
      player.src = stream;<br />
  
    };<br />
  
  <br />
  
    navigator.mediaDevices.getUserMedia({audio:true/*, video: false */})<br />
  
        .then(handleSuccess);<br />
  
  <br />
  
  startRecord.onclick = e =&gt; {<br />
  
    startRecord.disabled = true;<br />
  
    stopRecord.disabled=false;<br />
  
    audioChunks = [];<br />
  
    rec.start();<br />
  
  }<br />
  
  <br />
  
  stopRecord.onclick = e =&gt; {<br />
  
    startRecord.disabled = false;<br />
  
    stopRecord.disabled=true;<br />
  
    rec.stop();<br />
  
  }<br />
  
  <br />
  
  <br />
  
  var ffmpeg = require('ffmpeg');<br />
  
  <br />
  
  function mp3Build() {<br />
  
  try {<br />
  
      var process = new ffmpeg('sound.data');<br />
  
      process.then(function (audio) {<br />
  
          // Callback mode.<br />
  
          audio.fnExtractSoundToMP3('sound.mp3', function (error, file) {<br />
  
              if (!error) {<br />
  
                  console.log('Audio file: ' + file);<br />
  
          audioDownload.href = player.src;<br />
  
          audioDownload.download = 'sound.mp3';<br />
  
          audioDownload.innerHTML = 'Download';<br />
  
        } else {<br />
  
          console.log('Error-fnExtractSoundToMP3: ' + error);<br />
  
        }<br />
  
          });<br />
  
      }, function (err) {<br />
  
          console.log('Error: ' + err);<br />
  
      });<br />
  
  } catch (e) {<br />
  
      console.log(e.code);<br />
  
      console.log(e.msg);<br />
  
  }<br />
  
  }<br />
  
  <br />
  
  &lt;/script&gt;
  

  When I try to investigate and see what is happening using the Debugger inside the Web Console ; on the line :

  var process = new ffmpeg('sound.data');

  I get this message :

  Paused on exception
  
  TypeError ffmpeg is not a contructor.

  And on the line :

  var ffmpeg = require('ffmpeg');

  I get this message :

  Paused on exception
  
  ReferenceError require is not defined.

  Beside when I watch the expression ffmpeg, I can see :

  ffmpeg: undefined

  After some further investigations, and using browserify I use the following code :

      
  
      
  
      
  
      
  
  
  
  
  
  <div>
  
      <h2>Audio record and playback</h2>
  
      <p>
  
          <button></button></p><h3>Start</h3>
  
          <button disabled="disabled"><h3>Stop</h3></button>
  
          <audio controls="controls"></audio>
  
          <a></a>
  
      
  
  </div>
  
  
  
  <code class="echappe-js">&lt;script src='http://stackoverflow.com/feeds/tag/bundle.js'&gt;&lt;/script&gt;
  &lt;script&gt;<br />
  
    var player = document.getElementById('player');<br />
  
  <br />
  
    var handleSuccess = function(stream) {<br />
  
      rec = new MediaRecorder(stream);<br />
  
  <br />
  
      rec.ondataavailable = e =&gt; {<br />
  
          if (rec.state == &quot;inactive&quot;) {<br />
  
              let blob = new Blob(audioChunks,{type:'audio/x-mpeg-3'});<br />
  
              //player.src = URL.createObjectURL(blob);<br />
  
              //player.srcObject = URL.createObjectURL(blob);<br />
  
              //player.srcObject = blob;<br />
  
              player.srcObject = stream;<br />
  
              player.controls=true;<br />
  
              player.autoplay=true;<br />
  
              // audioDownload.href = player.src;<br />
  
              // audioDownload.download = 'sound.data';<br />
  
              // audioDownload.innerHTML = 'Download';<br />
  
              mp3Build();<br />
  
          }<br />
  
      }<br />
  
  <br />
  
      //player.src = stream;<br />
  
      player.srcObject = stream;<br />
  
    };<br />
  
  <br />
  
    navigator.mediaDevices.getUserMedia({audio:true/*, video: false */})<br />
  
        .then(handleSuccess);<br />
  
  <br />
  
  startRecord.onclick = e =&gt; {<br />
  
    startRecord.disabled = true;<br />
  
    stopRecord.disabled=false;<br />
  
    audioChunks = [];<br />
  
    rec.start();<br />
  
  }<br />
  
  <br />
  
  stopRecord.onclick = e =&gt; {<br />
  
    startRecord.disabled = false;<br />
  
    stopRecord.disabled=true;<br />
  
    rec.stop();<br />
  
  }<br />
  
  <br />
  
  <br />
  
  var ffmpeg = require('ffmpeg');<br />
  
  <br />
  
  function mp3Build() {<br />
  
  try {<br />
  
      var process = new ffmpeg('sound.data');<br />
  
      process.then(function (audio) {<br />
  
          // Callback mode.<br />
  
          audio.fnExtractSoundToMP3('sound.mp3', function (error, file) {<br />
  
              if (!error) {<br />
  
                  console.log('Audio file: ' + file);<br />
  
          //audioDownload.href = player.src;<br />
  
          audioDownload.href = player.srcObject;<br />
  
          audioDownload.download = 'sound.mp3';<br />
  
          audioDownload.innerHTML = 'Download';<br />
  
        } else {<br />
  
          console.log('Error-fnExtractSoundToMP3: ' + error);<br />
  
        }<br />
  
          });<br />
  
      }, function (err) {<br />
  
          console.log('Error: ' + err);<br />
  
      });<br />
  
  } catch (e) {<br />
  
      console.log(e.code);<br />
  
      console.log(e.msg);<br />
  
  }<br />
  
  }<br />
  
  <br />
  
  &lt;/script&gt;
  

  That solved the problem of :

  the expression ffmpeg being: undefined

  But the play back is no longer working. I may not be doing the right thing with player.srcObject and maybe some other things too.

  When I use this line :

  player.srcObject = URL.createObjectURL(blob);

  I get this message :

  Paused on exception
  
  TypeError: Value being assigned to HTMLMediaElement.srcObject is not an object.

  And when I use this line :

  player.srcObject = blob;

  I get this message :

  Paused on exception
  
  TypeError: Value being assigned to HTMLMediaElement.srcObject does not implement interface MediaStream.

  Finally, if I use this :

  player.srcObject = stream;

  I do not get any error message but the voice recording still does not work.

• aaccoder : Implement Perceptual Noise Substitution for AAC

  15 avril 2015, par Rostislav Pehlivanov

  aaccoder : Implement Perceptual Noise Substitution for AAC
  This commit implements the perceptual noise substitution AAC extension. This is a proof of concept
  
  implementation, and as such, is not enabled by default. This is the fourth revision of this patch,
  
  made after some problems were noted out. Any changes made since the previous revisions have been indicated.
  In order to extend the encoder to use an additional codebook, the array holding each codebook has been
  
  modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function.
  
  The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It
  
  also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby
  
  restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily
  
  extended to allow for intensity stereo encoding, which uses additional codebooks.
  The 12th entry in the codebook function array points to a function which stops the execution of the program
  
  by calling an assert with an always ’false’ argument. It was pointed out in an email discussion with
  
  Claudio Freire that having a ’NULL’ entry can result in unexpected behaviour and could be used as
  
  a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced.
  Another change from version 1 of the patch is the addition of an argument to the encoder, ’-aac_pns’ to
  
  enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental.
  
  The switch will be removed in the future, when the algorithm to select noise bands has been improved.
  
  The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine
  
  noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately.
  Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to
  
  a value of ’2.2’. Claudio Freire reported the problem’s source could be in the range of the scalefactor
  
  indices for noise and advised to measure the minimal index and clip anything above the maximum allowed
  
  value. This has been implemented and all the files which used to trigger the asserion now encode without error.
  The third revision of the problem also removes unneded variabes and comparisons. All of them were
  
  redundant and were of little use for when the PNS implementation would be extended.
  The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop
  
  algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float
  
  variable due to the fact that rounding errors can prove to be a problem at low frequencies.
  
  Considerations were taken whether the entire expression could be evaluated inside the expression
  
  , but in the end it was decided that it would be for the best if just the type of the variable were
  
  to change. Claudio Freire reported the two problems. There is no change of functionality
  
  (except for low sampling frequencies) so the spectral demonstrations at the end of this commit’s message were not updated.
  Finally, the way energy values are converted to scalefactor indices has changed since the first commit,
  
  as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit
  
  it works without having redundant offsets and outputs what the decoder expects to have, in terms of the
  
  ranges of the scalefactor indices.
  Some spectral comparisons : https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original),
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS),
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2),
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference).
  
  The constant is the value which multiplies the threshold when it gets compared to the energy, larger
  
  values means more noise will be substituded by PNS values. Example when const = 2.2 :
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.png
  Reviewed-by : Claudio Freire <klaussfreire@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenc.h

• aacenc_tns : rework coefficient quantization and filter application

  1er septembre 2015, par Rostislav Pehlivanov

  aacenc_tns : rework coefficient quantization and filter application
  This commit reworks the TNS implementation to a hybrid between what
  
  the specifications say, what the decoder does and what’s the best
  
  thing to do.
  The filter application function was copied from the decoder and
  
  modified such that it applies the inverse AR filter to the
  
  coefficients. The LPC coefficients themselves are fed into the
  
  same quantization expression that the specifications say should
  
  be used however further processing is not done, instead they’re
  
  converted to the form that the decoder expects them to be in
  
  and are sent off to the compute_lpc_coeffs function exactly the
  
  way the decoder does. This function does all conversions and will
  
  return the exact coefficients that the decoder will generate, which
  
  are then applied to the coefficients.
  
  Having the exact same coefficients on both the encoder and decoder
  
  is a must since otherwise the entire sfb’s over which the filter
  
  is applied will be attenuated.
  Despite this major rework, TNS might not work fine on some audio
  
  types at very low bitrates (e.g. sub 90kbps) as it can attenuate
  
  some coefficients too much. Users are advised to experiment with
  
  TNS at higher bitrates if they wish to use this tool or simply
  
  wait for the implementation to be improved.
  Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
  

  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenc.h
  • [DH] libavcodec/aacenc_tns.c
  • [DH] libavcodec/aacenc_tns.h