Why Cross Correlation (xcorr) of Two simultaneously Recorded Audio Signals Always return randomly different lags?

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ishitaisha6840 - 2021-02-18T12:06:30+00:00
Question: Why Cross Correlation (xcorr) of Two simultaneously Recorded Audio Signals Always return randomly different lags?

Hi everyone! I'm working on a sound localization project in which I record two audio  signals simultaneously and then take their 'cross correlation' to find out the "lags" existing between the two signals! But what happens is that every time a random angle is calculated because of the abrupt values of the lags each time! I don't know where I'm going wrong! Please guide me if there is a better approach to achieve a better sound localization ! The code is given as follows:     if true fs = 48000 ; %sampling frequency in Hz recObj1 = audiorecorder(fs, 16, 1, 1); recObj2 = audiorecorder(fs, 16, 1, 2); record(recObj1); record(recObj2); pause(5); % record for 5 seconds simultaneously stop(recObj1); stop(recObj2); out1 = getaudiodata(recObj1, 'int16'); out2 = getaudiodata(recObj2, 'int16'); L = out1 ; R = out2 ; t1 = (0:length(L)-1)/fs; t2 = (0:length(R)-1)/fs; figure; plot(t1,L); figure; plot(t2,R); threshold = 100; k=1; win =200 ; [k max(L) max(R)] if max(L)>th && max(R)>th %set power threshold [c, lags] = xcorr(L, R); [a1,b1] = max(L); [a2, b2] = max(R); [a3, b3] = max(c); s = lags(b3); time_delay = s/fs ; disp(time_delay); s = abs(s); % taking absolute of s disp('Estimated angle'); c = 342; % avg speed of sound at room temperature dis = 1 ; % mean distance between the two microphones cal = ((time_delay*c)/dis) ; if cal<-1 cal=-1; elseif cal>1 cal=1; end ang =((acosd(cal)) disp(ang); %displays the angle of sound source due to these microphones end

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    • Expert Answer

    Profile picture of Prashant Kumar Prashant Kumar answered . 2025-11-20

    I work at Matlabsolutions in the Audio System Toolbox team. I can't exactly replicate your experiement as I don't have your hardware, but I could notice a few possible sources of issues.
     
    In your code you seem to be acquiring  simultaneously from two different devices using the default audio drivers (typically DirectSound or WASAPI on Windows). That gives you no guarantee of synchronous acquisition for L and R. The two devices themselves may be triggered asynchronously by the operating system, giving you an arbitrary new delay between the two signals every single time you run your script.
     
    The simplest guarantee of a synchronous acquisition comes from acquiring different channels of the same device, ideally using its ASIO driver instead of the default one. ASIO drivers guarantee synchronous multi-channel acquisition.
     
    At a minimum, you should try something like the following, which gets L and R from the first two channels of the same device:
    recObj = audiorecorder(fs, 16, 2, deviceID);
    Even better, you may want to try audioDeviceReader object in Audio System Toolbox, which supports ASIO drivers and is routinely used out there for latency-sensitive acoustic measurements. See for example the first example from the reference doc page (Read from Microphone and Write to Audio File):
     
     
    deviceReader = audioDeviceReader('Driver','ASIO','NumChannels',2);
setup(deviceReader);
fileWriter = dsp.AudioFileWriter('myTwoChannelRecording.wav');
disp('Recording...')
tic;
while toc < 5
    acquiredAudio = deviceReader();
    fileWriter(acquiredAudio);
end
disp('Recording complete.')
release(deviceReader);
release(fileWriter);

    audioDeviceReader also gives you the ability to acquire whatever arbitrary pair of channels from your device through what we call channel mapping.


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