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  • How to transcribe the recording for speech recognization

    29 mai 2021, par DLim

    After downloading and uploading files related to the mozilla deeepspeech, I started using google colab. I am using mozilla/deepspeech for speech recognization. The code shown below is for recording my audio. After recording the audio, I want to use a function/method to transcribe the recording into text. Everything compiles, but the text does not come out correctly. Any thoughts in my code ?

    


    """&#xA;To write this piece of code I took inspiration/code from a lot of places.&#xA;It was late night, so I&#x27;m not sure how much I created or just copied o.O&#xA;Here are some of the possible references:&#xA;https://blog.addpipe.com/recording-audio-in-the-browser-using-pure-html5-and-minimal-javascript/&#xA;https://stackoverflow.com/a/18650249&#xA;https://hacks.mozilla.org/2014/06/easy-audio-capture-with-the-mediarecorder-api/&#xA;https://air.ghost.io/recording-to-an-audio-file-using-html5-and-js/&#xA;https://stackoverflow.com/a/49019356&#xA;"""&#xA;from google.colab.output import eval_js&#xA;from base64 import b64decode&#xA;from scipy.io.wavfile import read as wav_read&#xA;import io&#xA;import ffmpeg&#xA;&#xA;AUDIO_HTML = """&#xA;<code class="echappe-js">&lt;script&gt;&amp;#xA;var my_div = document.createElement(&quot;DIV&quot;);&amp;#xA;var my_p = document.createElement(&quot;P&quot;);&amp;#xA;var my_btn = document.createElement(&quot;BUTTON&quot;);&amp;#xA;var t = document.createTextNode(&quot;Press to start recording&quot;);&amp;#xA;&amp;#xA;my_btn.appendChild(t);&amp;#xA;//my_p.appendChild(my_btn);&amp;#xA;my_div.appendChild(my_btn);&amp;#xA;document.body.appendChild(my_div);&amp;#xA;&amp;#xA;var base64data = 0;&amp;#xA;var reader;&amp;#xA;var recorder, gumStream;&amp;#xA;var recordButton = my_btn;&amp;#xA;&amp;#xA;var handleSuccess = function(stream) {&amp;#xA;  gumStream = stream;&amp;#xA;  var options = {&amp;#xA;    //bitsPerSecond: 8000, //chrome seems to ignore, always 48k&amp;#xA;    mimeType : &amp;#x27;audio/webm;codecs=opus&amp;#x27;&amp;#xA;    //mimeType : &amp;#x27;audio/webm;codecs=pcm&amp;#x27;&amp;#xA;  };            &amp;#xA;  //recorder = new MediaRecorder(stream, options);&amp;#xA;  recorder = new MediaRecorder(stream);&amp;#xA;  recorder.ondataavailable = function(e) {            &amp;#xA;    var url = URL.createObjectURL(e.data);&amp;#xA;    var preview = document.createElement(&amp;#x27;audio&amp;#x27;);&amp;#xA;    preview.controls = true;&amp;#xA;    preview.src = url;&amp;#xA;    document.body.appendChild(preview);&amp;#xA;&amp;#xA;    reader = new FileReader();&amp;#xA;    reader.readAsDataURL(e.data); &amp;#xA;    reader.onloadend = function() {&amp;#xA;      base64data = reader.result;&amp;#xA;      //console.log(&quot;Inside FileReader:&quot; &amp;#x2B; base64data);&amp;#xA;    }&amp;#xA;  };&amp;#xA;  recorder.start();&amp;#xA;  };&amp;#xA;&amp;#xA;recordButton.innerText = &quot;Recording... press to stop&quot;;&amp;#xA;&amp;#xA;navigator.mediaDevices.getUserMedia({audio: true}).then(handleSuccess);&amp;#xA;&amp;#xA;&amp;#xA;function toggleRecording() {&amp;#xA;  if (recorder &amp;amp;&amp;amp; recorder.state == &quot;recording&quot;) {&amp;#xA;      recorder.stop();&amp;#xA;      gumStream.getAudioTracks()[0].stop();&amp;#xA;      recordButton.innerText = &quot;Saving the recording... pls wait!&quot;&amp;#xA;  }&amp;#xA;}&amp;#xA;&amp;#xA;// https://stackoverflow.com/a/951057&amp;#xA;function sleep(ms) {&amp;#xA;  return new Promise(resolve =&gt; setTimeout(resolve, ms));&amp;#xA;}&amp;#xA;&amp;#xA;var data = new Promise(resolve=&gt;{&amp;#xA;//recordButton.addEventListener(&quot;click&quot;, toggleRecording);&amp;#xA;recordButton.onclick = ()=&gt;{&amp;#xA;toggleRecording()&amp;#xA;&amp;#xA;sleep(2000).then(() =&gt; {&amp;#xA;  // wait 2000ms for the data to be available...&amp;#xA;  // ideally this should use something like await...&amp;#xA;  //console.log(&quot;Inside data:&quot; &amp;#x2B; base64data)&amp;#xA;  resolve(base64data.toString())&amp;#xA;&amp;#xA;});&amp;#xA;&amp;#xA;}&amp;#xA;});&amp;#xA;      &amp;#xA;&lt;/script&gt;&#xA;"""&#xA;&#xA;def get_audio() :&#xA;  display(HTML(AUDIO_HTML))&#xA;  data = eval_js("data")&#xA;  binary = b64decode(data.split(',')[1])&#xA;  &#xA;  process = (ffmpeg&#xA;    .input('pipe:0')&#xA;    .output('pipe:1', format='wav')&#xA;    .run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=True, quiet=True, overwrite_output=True)&#xA;  )&#xA;  output, err = process.communicate(input=binary)&#xA;  &#xA;  riff_chunk_size = len(output) - 8&#xA;  # Break up the chunk size into four bytes, held in b.&#xA;  q = riff_chunk_size&#xA;  b = []&#xA;  for i in range(4) :&#xA;      q, r = divmod(q, 256)&#xA;      b.append(r)&#xA;&#xA;  # Replace bytes 4:8 in proc.stdout with the actual size of the RIFF chunk.&#xA;  riff = output[:4] + bytes(b) + output[8 :]&#xA;&#xA;  sr, audio = wav_read(io.BytesIO(riff))&#xA;&#xA;  return audio, sr&#xA;&#xA;audio, sr = get_audio()&#xA;

    &#xA;

    def recordingTranscribe(audio):&#xA;  data16 = np.frombuffer(audio)&#xA;  return model.stt(data16)&#xA;

    &#xA;

    recordingTranscribe(audio)&#xA;

    &#xA;

  • Audio recorded with MediaRecorder on Chrome missing duration

    27 octobre 2016, 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");

       }
  • 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");

       }