![]() video files (multiple audio streams are supported). # Extracting audio from flv, avi, mov etc. # Informative and resizable UI suited even for netbooks. # Grabbing of multi-channel Audio CDs to the desired format at once. # Converting to many formats at once using "Multiple" output mode. ![]() # "Glue" input files to one large audio file and create CUE sheet. # Built-in Metadata editor with Cover Art support. # Support of embedded CUE sheets (for FLAC, WavPack and APE files). # Parallel conversion by utilizing power of multi-core CPUs. # Works on XP,Vista,Windows7 32/64 bit versions. Var binaryData = fs.readFileSync('output.wav') Ĭonnection.Xrecode III is converter and audio-grabber which allows you to convert from mp3, mp2 ,wma, cda, ogg, flac, ape, cue, ac3, wv, mpc, cue ,tta, tak, wav, dts, m4a, m4b, mp4, ra, rm, aac, avi, mpg, vob, mkv, flv, swf, mov, ofr, wmv, divx, m4v, spx, 3gp, 3g2, m2v, m4v, ts, m2ts, adts, shn, tak, xm, mod, s3m, it, mtm, umx to m4a, alac, ape, flac, mp3, mp4 (using NeroAAC), ogg, raw, wav, wma, WavPack, mpc, mp2, Speex, ofr, ac3 and Shorten formats. I'm writing this example using npm websocket package. Since nexmo will send you 640 bytes every 20ms, you can just send back 640 bytes at same time. write when you get audio from nexmo (this is the one I will show).start an async task that runs every 20ms and writes 640 bytes of data to websocket.Anything more than that will be discarded run a loop to write write all the audio to the websocket but in chunks of 640 bytes.īut this has an issue, Nexmo will buffer only first 20s of audio.In your code, when you send audio back, you need to stream it as chunks of 640 bytes, not the entire file data in one shot. This converts your filename.mp3 to output.wav which will be Linear PCM 16-bit in 16K samplerate Install ffmpeg on your system and run this commandįfmpeg -i filename.mp3 -ar 16000 -sample_fmt s16 output.wav This means a 20ms/0.02s frame of audio will be equivalent to (32000*0.2) = 640 bytesĬonvert mp3 to wav. So an audio chunk of 1 second will contain bytes = (16000 * 2) = 32000 bytes How you send this audio data to them and how they send it to you: in chunks of audio data worth 20ms framesīased on the audio format, if you choose "16bit Linear PCM with sample rate of 16K" implies:.The format of audio data, which is "Linear PCM 16-bit, with either a 8kHz or a 16kHz sample rate".Linear PCM 16-bit, with either a 8kHz or a 16kHz sample rate, and a I have the file in the project folder but i dont know how to send it via websocket, i looked for how to do so but i dident find anything. I am trying to send the file trough the web socket but i dont know how to do so, " Linear PCM 16-bit, with either a 8kHz or a 16kHz sample rate, and a 20ms frame size " So i want to convert the mp3 file to the same format so i could send it to via ws.send().Īfter making my audio file at the right format which is: Its works (the server echoing the phone call ) Now, if i send the pcm data from the stream (the raw audio from phone call) Some background, the stream is a phone call (via vonage api) so ny ws connected to phone call and hear the user input, and then after some logic on my server i want to play to the user a mp3 file that is a local file in my server, via ws.send(). Ws.send(mp3File) <= stream back the audio file Here is what i want to do: let mp3File = // the 16-bit pcm file I am trying to send an audio file through a websocket, and I realised that in order to do so i need to convert the mp3 file to a Linear PCM 16-bit code, but i cant find a way to do so.
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