audio-processing

问题 I'm trying to record my microphone input and process it at the same time. I tried with a loop with this inside: recordblocking(recorder, 1); y = getaudiodata(recorder); % any processing on y But while I'm doing something with y , I'm losing information since not recording continuously. Is there something I could do to continuously record sound coming in my microphone, store it in some kind of buffer, and process chunks of it at the same time? A delay isn't a problem, but I really need the

问题 I'm trying to record my microphone input and process it at the same time. I tried with a loop with this inside: recordblocking(recorder, 1); y = getaudiodata(recorder); % any processing on y But while I'm doing something with y , I'm losing information since not recording continuously. Is there something I could do to continuously record sound coming in my microphone, store it in some kind of buffer, and process chunks of it at the same time? A delay isn't a problem, but I really need the

问题 I'm trying to record my microphone input and process it at the same time. I tried with a loop with this inside: recordblocking(recorder, 1); y = getaudiodata(recorder); % any processing on y But while I'm doing something with y , I'm losing information since not recording continuously. Is there something I could do to continuously record sound coming in my microphone, store it in some kind of buffer, and process chunks of it at the same time? A delay isn't a problem, but I really need the

问题 I have a wave file in 16bit PCM form. I've got the raw data in a byte[] and a method for extracting samples, and I need them in float format, i.e. a float[] to do a Fourier Transform. Here's my code, does this look right? I'm working on Android so javax.sound.sampled etc. is not available. private static short getSample(byte[] buffer, int position) { return (short) (((buffer[position + 1] & 0xff) << 8) | (buffer[position] & 0xff)); } ... float[] samples = new float[samplesLength]; for (int i

问题 I would like to get some sort of distance measure between two pieces of audio. For example, I want to compare the sound of an animal to the sound of a human mimicking that animal, and then return a score of how similar the sounds were. It seems like a difficult problem. What would be the best way to approach it? I was thinking to extract a couple of features from the audio signals and then do a Euclidian distance or cosine similarity (or something like that) on those features. What kind of

问题 I get two different audio samples from two sources. For microphone sound: audioRecord = new AudioRecord(MediaRecorder.AudioSource.DEFAULT, 44100, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT, (AudioRecord.getMinBufferSize(44100, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT)*5)); For Internal sound: audioRecord = new AudioRecord.Builder() .setAudioPlaybackCaptureConfig(config) .setAudioFormat(new AudioFormat.Builder() .setEncoding(AudioFormat.ENCODING_PCM

来源： https://stackoverflow.com/questions/50797550/python-compare-two-audio-files-which-may-have-noise

问题 I have a load of 3 hour MP3 files, and every ~15 minutes a distinct 1 second sound effect is played, which signals the beginning of a new chapter. Is it possible to identify each time this sound effect is played, so I can note the time offsets? The sound effect is similar every time, but because it's been encoded in a lossy file format, there will be a small amount of variation. The time offsets will be stored in the ID3 Chapter Frame MetaData. Example Source, where the sound effect plays

问题 We have two lowpass filters with a different cutoff value: b, a = signal.butter(2, 0.125) b2, a2 = signal.butter(2, 0.140) When applying the first filter to x[0:10000] and the second to x[10000:20000] with lfilter , we have to use initial conditions for the output to be "continuous", as seen here in the answer of Continuity issue when applying an IIR filter on successive time-frames: zi = lfilter_zi(b, a) x[0:10000], zi = lfilter(b, a, x[0:10000], zi=zi) x[10000:20000], zi = lfilter(b2, a2, x

问题 First of all I asked this question in Stack Exchange and I am getting only concept related answers and not implementation oriented. So, my problem is I am trying to create high pass filter and I implemented using Python. from numpy import cos, sin, pi, absolute, arange from scipy.signal import kaiserord, lfilter, firwin, freqz, firwin2 from pylab import figure, clf, plot, xlabel, ylabel, xlim, ylim, title, grid, axes, show # Nyquist rate. nyq_rate = 48000 / 2 # Width of the roll-off region.