webrtcvad | webrtcvad provides node.js bindings | Runtime Evironment library
kandi X-RAY | webrtcvad Summary
kandi X-RAY | webrtcvad Summary
webrtcvad is a cross-platform, native node.js addon for detecting speech in raw audio. webrtcvad provides node.js bindings for the native WebRTC voice activity detection library.
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webrtcvad Examples and Code Snippets
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Trending Discussions on webrtcvad
QUESTION
I am trying to run the example code of webRTC VAD
found here.
But when I feed it a mono-16bit wave file of just me speaking with very long pauses, it just detects the entire file to be voiced, and the voiced output chunk-00.wav
is the entire audio file.
Any help is greatly appreciated. Below I have given the console output that I receive.
...ANSWER
Answered 2022-Jan-31 at 11:45I think I have found an alternative method of how to get VAD data. Instead of trying to get VAD from the pre-defined method shown in the link above, I create my own function.
The function basically measures the amplitude of the wave and any sharp spike observed above the base noise level (1.6x the base value
) is taken to mean a voiced activity. This function assumes that only 1 human is speaking and that the noise level remains relatively constant.
QUESTION
I am trying to make a speech to text system using raspberry pi. There are many problems with VAD. I am using DeepCpeech's VAD script. Adafruit I2S MEMS microphone accepts only 32-bit PCM audio. So I modified the script to record 32-bit audio and then convert it to 16 bit for DeepSpeech's processing. Frames generation and conversation parts are below:
...ANSWER
Answered 2022-Jan-26 at 13:36I searched for DeepCpeech's VAD script and found it. The problem is connected with the webrtcvad. The webrtcvad VAD only accepts 16-bit mono PCM audio, sampled at 8000, 16000, 32000 or 48000 Hz. So you need to convert the 32-bit frame to 16-bit (I am about PyAudio output frame) to process webrtcvad.is_speech(). I changed and it worked fine.
QUESTION
I am able to read the audio but I am getting an error message while passing it to VAD(Voice Activity Detector). I think the error message is because the frames is in bytes, when feeding it to vad.is_speech(frame, sample_rate), should this frame be in bytes? Here is the code below:
...ANSWER
Answered 2021-May-02 at 10:13I have solved it, you know vad.is_speech(buf=frame, sample_rate)
, it takes the buf and calculates it length, but an integer value does not posses the len()
attributes in python.
This throws an error for example:
QUESTION
I am working with audio file using webrtcvad and pydub. The split of any fragment is by silence of the sentence. Is there any way by which the split can be done at word level boundry condition? (after each spoken word)? If librosa/ffmpeg/pydub has any feature like this, can split is possible at each vocal? but after split, I need start and end time of the vocal exactly what that vocal part has positioned in the original file. One simple solution or way to split by ffmpeg is also defined by :
https://gist.github.com/vadimkantorov/00bf4fbe4323360722e3d2220cc2915e
but this is also splitting by silence, and with each padding number or the frame size, the split is different. I am trying split by vocal. As example, I have done this manually the original file, split words and its time position in json is in a folder provided here under the link:
www.mediafire.com/file/u4ojdjezmw4vocb/attached_problem.tar.gz
...ANSWER
Answered 2020-Oct-20 at 10:17Delimiting words is out of the audio domain and requires a kind of intelligence. Doing it manually is easy because we are intelligent and know exactly what we are looking for, but automatizing the process is hard because, as you already noticed, a silence is not (not only, not always) a word delimiter.
At audio level, we can only approach a solution and this require both analyzing the amplitude of the signal and adding some time mechanisms. As an example, Protools provides a nice tool named Strip Silence that cuts audio regions automatically based on the amplitude of the signal. It always keeps the material at its original position in the timeline and naturally each region knows its own duration. In addition to the threshold in dB, and to prevent creating too much regions, it provides several useful parameters in the time domain : a minimum length for the created regions, a delay before the cut (the delay is computed from the point the amplitude passes below the threshold), an inverted delay before reopening the gate (the delay is computed backward from the point the amplitude passes above the threshold).
This could be a good starting point for you. Implementing such a system probably won't be 100 % successful, but you could obtain a quite good ratio if the settings are well adjusted to the speaker. Even if it's not perfect, it will significantly reduce the need for manual work.
QUESTION
I am getting a problem while trying to get the binary result using webrctvad in a wave format audio file. I am using librosa in order to load the audio file in .wav format. Can anyone tell me how to use librosa along with webrtcvad in order to get the binary output of whether the audio contains speech or not?
Webrtcvad module works correctly with the wave module
The above link helped me a lot but still, I am confused as the link contains a good explanation but during implementation lot of errors are coming.
...ANSWER
Answered 2020-May-16 at 11:28py-webrtcvad, expects the audio data to be 16bit PCM little-endian - as is the most common storage format in WAV files.
librosa
and its underlying I/O library pysoundfile
however always returns floating point arrays in the range [-1.0, 1.0]
. To convertt this to bytes containing 16bit PCM you can use the following float_to_pcm16
function.
And I have tested to use the read_pcm16
function a direct replacement of read_wave
in the official py-webrtcvad example. But allowing to open any audio file supported by soundfile (WAV, FLAC, OGG) etc.
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