You cannot do it using standard user mode APIs. You need to either hook APIs or create virtual devices to accept application streams/sessions. Intercepting and postprocessing all audio streams on Windows Recording Audio Output from a Specific Program on Windows Is it possible to caputre the rendering audio session from another process? Capture audio of … Read more
$(‘thumbnail’).toggleClass(‘pausing playing’); ToggleClass: Add or remove one or more classes from each element in the set of matched elements, depending on either the class’s presence or the value of the switch argument.
Yes, it is possible to recover most of the signal and estimate the phase with e.g. Griffin-Lim Algorithm (GLA). Its “fast” implementation for Python can be found in librosa. Here’s how you can use it: import numpy as np import librosa y, sr = librosa.load(librosa.util.example_audio_file(), duration=10) S = np.abs(librosa.stft(y)) y_inv = librosa.griffinlim(S) And that’s how … Read more
An other way to do it , with AVAudioConverter in Swift 5 let engine = AVAudioEngine() func setup() { let input = engine.inputNode let bus = 0 let inputFormat = input.outputFormat(forBus: bus ) guard let outputFormat = AVAudioFormat(commonFormat: .pcmFormatFloat32, sampleRate: 8000, channels: 1, interleaved: true), let converter = AVAudioConverter(from: inputFormat, to: outputFormat) else{ return } … Read more
The current top answer is a bit outdated and only works for python 2. Here is a version updated for python 3. It wraps the functions into classes and packages everything into one simple easy-to-use version. Note that there is one key difference between the top answer and my script: The script at the top … Read more
You can use the atempo filter in ffmpeg 1.0 and later: ffmpeg -i input.wav -f:a atempo=1.25 output.wav atempo only accepts values between 0.5 and 2.0, but you can bypass the limitation by chaining multiple filters: ffmpeg -i input.wav -f:a atempo=2,atempo=1.5 output.wav
mSoundManager.addSound(1,R.raw.dit); mSoundManager.addSound(1,R.raw.dah); You need to change the second line to: mSoundManager.addSound(2,R.raw.dah); In order to play multiple sounds at once, first you need to let the SoundPool know that. In the declaration of SoundPool notice that I specified 20 streams. I have many guns and bad guys making noise in my game, and each has a … Read more
Principally the problem seems to be that you are reading the audio data incorrectly. Specifically I’m not really sure what this excerpt is supposed to mean: if (tempBuffer[j] > tempBuffer[j+1]) … tempBuffer[j] – tempBuffer[j+1]; else … tempBuffer[j + 1] – tempBuffer[j]; But anyhow since you are recording 16-bit data the bytes in the byte array … Read more
from pydub import AudioSegment from pydub import effects root = r’audio.wav’ velocidad_X = 1.5 # No puede estar por debajo de 1.0 sound = AudioSegment.from_file(root) so = sound.speedup(velocidad_X, 150, 25) so.export(root[:-4] + ‘_Out.mp3’, format=”mp3″)
OK, I am answering my own question if it helps someone else. After few days of tireless trying, at last i have got it working. Below is a complete Function written with Objective-C and C. It takes a file path as a parameter which contains RAW PCM data directly captured from microphone and returns a … Read more