wav

I get clicks at the start and end of playing a sound (a wav from the sdcard). It must be something to do with the track buffering but I dont know the solution. Also, I create a new one of these every time the sound plays, is this ok or is there a better way? There are many sounds playing over and over. Here is the code: public void PlayAudioTrack(final String filePath, final Float f) throws IOException { new Thread(new Runnable() { public void run() { //play sound here int minSize = AudioTrack.getMinBufferSize( 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT );

When reading a audio file using something such as rate, wavData = scipy.io.wavfile.read('test.wav') what exactly does the data inside of wavData represent? wavData is an array of numbers each representing a single sample of the audio signal. The samples are snapshots of the audio amplitude spaced evenly in time. So if your rate is returned as 48000 then the first 48000 elements of wavData would be 1 second worth of the audio signal. For more information read about PCM on wikipedia: https://en.wikipedia.org/wiki/Pulse-code_modulation . 来源： https://stackoverflow.com/questions/33640274/what

Is it possible to compress a wav audio file without reducing the sampling rate? I have an audio file with 256 bit rate and sampling rate - 8000Hz. I would just like to reduce the bit rate to 128/64 kbs I tried converting to mp3 and back to wav, ffmpeg -i input.wav 1.mp3 ffmpeg -i "1.mp3" -acodec pcm_s16le -ar 4000 out.wav but this reduced sampling rate as well. ffmpeg -i "1.mp3" -acodec pcm_s16le -ab 128 out.wav has default 256 bit rate PCM ("WAV") is uncompressed, so -b:a / -ab is ignored. Calculating PCM bitrate Assuming a stereo input, 8000 samples per second, 16 bits per sample: sample

I know that questions about .wav files in Python have been just about beaten to death, but I am extremely frustrated as no one's answer seems to be working for me. What I'm trying to do seems relatively simple to me: I want to know exactly what frequencies there are in a .wav file at given times. I want to know, for example, "from the time n milliseconds to n + 10 milliseconds, the average frequency of the sound was x hertz". I have seen people talking about Fourier transforms and Goertzel algorithms, as well as various modules, that I can't seem to figure out how to get to do what I've

I have 2 wav files that I want to concatenate into one file with the two sound tracks. Is there any api for that task or some built in commands in .NET that can I can use in some genius way to make this task possible? Many thanks for the help. :) If I'm not mistaken, you can just append the bytes from the second file to the end of the first. If there is any header data, you'll want to exclude that (see here ) Be advised that you may hear a click or pop between the two clips, as there will be no blending. You could probably get around this by fading (gradually reducing the values for the last

I'm trying to read a wav file from the TIMIT database in python but I get an error: When I'm using wave: wave.Error: file does not start with RIFF id When I'm using scipy: ValueError: File format b'NIST'... not understood. and when I'm using librosa, the program got stuck. I tried to convert it to wav using sox: cmd = "sox " + wav_file + " -t wav " + new_wav subprocess.call(cmd, shell=True) and it didn't help. I saw an old answer referencing to the package scikits.audiolab but it looks like it is no longer supported. How can I read these file to get a ndarray of the data? Thanks Your file is