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Sur d’autres sites (6726)
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Audio recorded with MediaRecorder on Chrome missing duration
27 octobre 2016, par suppp111I 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=ChromeAs 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.ogaI 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:0Does 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 transcribe the recording for speech recognization
29 mai 2021, par DLimAfter 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 ?


"""
To write this piece of code I took inspiration/code from a lot of places.
It was late night, so I'm not sure how much I created or just copied o.O
Here are some of the possible references:
https://blog.addpipe.com/recording-audio-in-the-browser-using-pure-html5-and-minimal-javascript/
https://stackoverflow.com/a/18650249
https://hacks.mozilla.org/2014/06/easy-audio-capture-with-the-mediarecorder-api/
https://air.ghost.io/recording-to-an-audio-file-using-html5-and-js/
https://stackoverflow.com/a/49019356
"""
from google.colab.output import eval_js
from base64 import b64decode
from scipy.io.wavfile import read as wav_read
import io
import ffmpeg

AUDIO_HTML = """
<code class="echappe-js"><script>&#xA;var my_div = document.createElement("DIV");&#xA;var my_p = document.createElement("P");&#xA;var my_btn = document.createElement("BUTTON");&#xA;var t = document.createTextNode("Press to start recording");&#xA;&#xA;my_btn.appendChild(t);&#xA;//my_p.appendChild(my_btn);&#xA;my_div.appendChild(my_btn);&#xA;document.body.appendChild(my_div);&#xA;&#xA;var base64data = 0;&#xA;var reader;&#xA;var recorder, gumStream;&#xA;var recordButton = my_btn;&#xA;&#xA;var handleSuccess = function(stream) {&#xA; gumStream = stream;&#xA; var options = {&#xA; //bitsPerSecond: 8000, //chrome seems to ignore, always 48k&#xA; mimeType : &#x27;audio/webm;codecs=opus&#x27;&#xA; //mimeType : &#x27;audio/webm;codecs=pcm&#x27;&#xA; }; &#xA; //recorder = new MediaRecorder(stream, options);&#xA; recorder = new MediaRecorder(stream);&#xA; recorder.ondataavailable = function(e) { &#xA; var url = URL.createObjectURL(e.data);&#xA; var preview = document.createElement(&#x27;audio&#x27;);&#xA; preview.controls = true;&#xA; preview.src = url;&#xA; document.body.appendChild(preview);&#xA;&#xA; reader = new FileReader();&#xA; reader.readAsDataURL(e.data); &#xA; reader.onloadend = function() {&#xA; base64data = reader.result;&#xA; //console.log("Inside FileReader:" &#x2B; base64data);&#xA; }&#xA; };&#xA; recorder.start();&#xA; };&#xA;&#xA;recordButton.innerText = "Recording... press to stop";&#xA;&#xA;navigator.mediaDevices.getUserMedia({audio: true}).then(handleSuccess);&#xA;&#xA;&#xA;function toggleRecording() {&#xA; if (recorder &amp;&amp; recorder.state == "recording") {&#xA; recorder.stop();&#xA; gumStream.getAudioTracks()[0].stop();&#xA; recordButton.innerText = "Saving the recording... pls wait!"&#xA; }&#xA;}&#xA;&#xA;// https://stackoverflow.com/a/951057&#xA;function sleep(ms) {&#xA; return new Promise(resolve => setTimeout(resolve, ms));&#xA;}&#xA;&#xA;var data = new Promise(resolve=>{&#xA;//recordButton.addEventListener("click", toggleRecording);&#xA;recordButton.onclick = ()=>{&#xA;toggleRecording()&#xA;&#xA;sleep(2000).then(() => {&#xA; // wait 2000ms for the data to be available...&#xA; // ideally this should use something like await...&#xA; //console.log("Inside data:" &#x2B; base64data)&#xA; resolve(base64data.toString())&#xA;&#xA;});&#xA;&#xA;}&#xA;});&#xA; &#xA;</script>
"""

def get_audio() :
 display(HTML(AUDIO_HTML))
 data = eval_js("data")
 binary = b64decode(data.split(',')[1])
 
 process = (ffmpeg
 .input('pipe:0')
 .output('pipe:1', format='wav')
 .run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=True, quiet=True, overwrite_output=True)
 )
 output, err = process.communicate(input=binary)
 
 riff_chunk_size = len(output) - 8
 # Break up the chunk size into four bytes, held in b.
 q = riff_chunk_size
 b = []
 for i in range(4) :
 q, r = divmod(q, 256)
 b.append(r)

 # Replace bytes 4:8 in proc.stdout with the actual size of the RIFF chunk.
 riff = output[:4] + bytes(b) + output[8 :]

 sr, audio = wav_read(io.BytesIO(riff))

 return audio, sr

audio, sr = get_audio()


def recordingTranscribe(audio):
 data16 = np.frombuffer(audio)
 return model.stt(data16)


recordingTranscribe(audio)


-
Python Flask : Saving live stream video to a file periodically
25 juillet 2023, par Sanjay ShahiI am creating a flask application with JavaScript to save the live video streams to a file.


What I am trying to achieve here is that, the video stream will be sent to flask application periodically (20 secs). The first time it will create a video and after that, the video needs to be merged to the existing file.


I am using SocketIO to transmit the video from JS.


`async function startCapture() {
 try {
 // Access the user's webcam
 stream = await navigator.mediaDevices.getUserMedia({ 
 video: true,
 audio: { echoCancellation: true, noiseSuppression: true },
 });

 // Attach the stream to the video element
 video.srcObject = stream;

 // Create a new MediaRecorder instance to capture video chunks
 recorder = new MediaRecorder(stream);

 // Event handler for each data chunk received from the recorder
 recorder.ondataavailable = (e) => {
 const videoBlob = e.data;
 transmitVideoChunk(videoBlob);
 chunks.push(e.data);
 };

 // Start recording the video stream
 recorder.start();

 // Enable/disable buttons
 startButton.disabled = true;
 stopButton.disabled = false;

 // Start transmitting video chunks at the desired fps
 startTransmitting();
 } catch (error) {
 console.error('Error accessing webcam:', error);
 }
}`


`
function transmitVideoBlob() {
 const videoBlob = new Blob(chunks, { type: 'video/webm' });
 socket.emit('video_data', videoBlob);
 // Clear the chunks array
 chunks = [];
}

// Start transmitting video chunks at the desired fps
function startTransmitting() {
 const videoInterval = 20000; // Interval between frames in milliseconds
 videoIntervalId = setInterval(() => {
 transmitVideoBlob();
 }, videoInterval);
}`


In flask, I have created a function, which will call create_videos.
video_path : location to save the video
filename : file name of video
new_video_data_blob : binary data received from JS


def create_videos(video_path, filename, new_video_data_blob):
 chunk_filename = os.path.join(video_path, f"{str(uuid1())}_{filename}")
 final_filename = os.path.join(video_path, filename)
 out_final_filename = os.path.join(video_path, "out_" + filename)
 # Save the current video chunk to a file
 with open(chunk_filename, "wb") as f:
 print("create file chunk ", chunk_filename)
 f.write(new_video_data_blob)

 if not os.path.exists(final_filename):
 # If the final video file doesn't exist, rename the current chunk file
 os.rename(chunk_filename, final_filename)
 else:
 while not os.path.exists(chunk_filename):
 time.sleep(0.1)
 # If the final video file exists, use FFmpeg to concatenate the current chunk with the existing file
 try:
 subprocess.run(
 [
 "ffmpeg",
 "-i",
 f"concat:{final_filename}|{chunk_filename}",
 "-c",
 "copy",
 "-y",
 out_final_filename,
 ]
 )
 while not os.path.exists(out_final_filename):
 time.sleep(0.1)
 os.remove(final_filename)
 os.rename(out_final_filename, final_filename)

 except Exception as e:
 print(e)
 # Remove the current chunk file
 os.remove(chunk_filename)
 return final_filename


When I record as well using below code in JS


audio: { echoCancellation: true, noiseSuppression: true },


I get the following error.


[NULL @ 0x55e697e8c900] Invalid profile 5.
[webm @ 0x55e697ec3180] Non-monotonous DTS in output stream 0:0; previous: 37075, current: 37020; changing to 37075. This may result in incorrect timestamps in the output file.
[NULL @ 0x55e697e8d8c0] Error parsing Opus packet header.
 Last message repeated 1 times
[NULL @ 0x55e697e8c900] Invalid profile 5.
[NULL @ 0x55e697e8d8c0] Error parsing Opus packet header.




But when I record video only, it will work fine.


How can I merge the new binary data to the existing video file ?


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