wav

问题 I've only used MATLAB as a calculator, so I'm not as well versed in the program. I hope a kind person may be able to guide me on the way since Google currently is not my friend. I have a wav file in the link below, where there is a human voice and some noise in the background. I want the noise removed. Is there anyone who can tell me how to do it in MATLAB? https://www.dropbox.com/s/3vtd5ehjt2zfuj7/Hold.wav 回答1: This is a pretty imperfect solution, especially since some of the noise is

问题 I need to analyze sound written in a .wav file. For that I need to transform this file into set of numbers (arrays, for example). I think I need to use the wave package. However, I do not know how exactly it works. For example I did the following: import wave w = wave.open('/usr/share/sounds/ekiga/voicemail.wav', 'r') for i in range(w.getnframes()): frame = w.readframes(i) print frame As a result of this code I expected to see sound pressure as function of time. In contrast I see a lot of

问题 I want in the win Application (written in C#), the sound (Wav Format) play as Background Sound, and mouse over Control play small wav sound File, and when Click on Button, Stop Background Sound and ... . Thanks For your guidance. 回答1: you can use Windows Media Player: reference C:\Windows\System32\wmp.dll in your project to launch a wav file: WMPLib.WindowsMediaPlayer wmp = new WMPLib.WindowsMediaPlayerClass(); wmp.URL = "[wav file path]"; //then control the player with : wmp.controls.play(),

问题 I am trying to play sound by using QAudioOutput and wav in "raw format". After timer's timeout (every 50ms) I do following: QByteArray TempSBuffer; short int *hi; // Check if wav has reached their end and reset its position to the beginning if yes if((m_timerStepNum+1)*m_audioOutput->periodSize()>=m_soundBuffer.size()) { m_timerStepNum=0; } // 2. Write the buffer data for the next timecycle into a temporary QByteArray TempSBuffer TempSBuffer=m_soundBuffer.mid(m_timerStepNum*m_audioOutput-

问题 I'm trying to analyse the audio quality of a file by detecting the highest frequency present (compressed audio will generally be filtered to something less than 20KHz). I'm reading WAV file data using a class from the soundstretch library which returns PCM samples as floats, then performing FFT on those samples with the fftw3 library. Then for each frequency (rounded to the nearest KHz), I am totalling up the amplitude for that frequency. So for a low quality file that doesn't contain

问题 I want to convert a given flac file to a wav file. Most of the questions on StackOverflow are the opposite of my question. Have anyone an idea how to solve this problem ? I heard something about JavaFlacEncoder link but I could not find a method to convert flac to wave. 回答1: Use flac -d to (from the man page) Decode (the default behavior is to encode) Another option would be the JustFLAC library, again you need to decode from FLAC. WAV is uncompressed audio. 来源： https://stackoverflow.com

问题 could someone provide me a link to source code (preferably C/C++) of a simple WAV 16-bit to 8-bit (and back if possible) converter? I have basic knowledge of C++ and I need to have a resource to understand wav writing and converting values for my new project. At this moment I have a lecture about RIFF chunks structure. Also any formulas for converting values between different bit depths (also these un standard) appreciated... I'll prefer it to be simple and command line so I can edit it with

问题 When an application calls waveOutWrite(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh) on windows, I save a copy of the audio data (pointed to by pwh->lpData) sent to the Audio Card in memory. Now I want to save the audio data as a WAV format file on the disk, how should I do this? I have the pointer to the audio data block and the size of the audo data. 回答1: The wav format is pretty trivial, and easy to write your own implementation for. Alternatively, you could use a library like libsndfile. 回答2:

问题 I've been asked to sample some data in a .wac file type. I'm not familiar with this standard and there is very little on the internet with regards to this format. I got given the .wav file but I don't think it was converted correctly, in that there was a none existent of the RIFF header so no .wav reader was able to read it. Could anyone therefore shed some light into how I could possibly convert the .wac file into a .wav file? Doing some research, I cannot seem to find a converter tool on

问题 How can I remove the first .5 of a second from Wav file? 回答1: Here is the format. You need to open up the file, find the size of the header, then remove the required number of samples. You can find the number of bits per sample in the header. A 16-bit WAV file at a sampling rate of 44100 Hz would require you to remove (16/8) * 44100 * 0.5 = 44100 bytes 回答2: A quick Google search turned up a library that may be able to do this. 回答3: If you need to do it programatically, you need a wav file