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Asked: 2026-05-31T17:06:54+00:00

When you set soundcard rate to, for example, 44100, you cannot guarantee actual rate

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When you set soundcard rate to, for example, 44100, you cannot guarantee actual rate be equal 44100. In my case traffic measurements between application and ALSA (in samples/sec) gave me value of 44066…44084.

This should not be related to resampling issues: even only-48000 hardware must “eat” data at 44100 rate in “44100” mode.

The problem occurs when i try to draw a cursor over waveform while this waveform is playing. I calculate cursor position using “ideal” sampling rate read from WAV-file (22050, …, 44100, …, 48000) and the milliseconds spent after playing start, using following C++ function:

long long getCurrentTimeMs(void)
{
    boost::posix_time::ptime now = boost::posix_time::microsec_clock::local_time();
    boost::posix_time::ptime    epoch_start(boost::gregorian::date(1970,1,1));
    boost::posix_time::time_duration dur = now - epoch_start;
    return dur.total_milliseconds();
}

QTimer is used to generate frames for cursor animation, but i do not depend on QTimer precision, because i ask time by getCurrentTimeMs() (assiming it is precise enough) every frame, so i can work with varying framerate.

After 2-3 minutes of playing i see a little difference between what i hear and what i see – the cursor position is greater than playing position for something like 1/20 of second or so.

When i measure traffic that go through ALSA’s callback i get mean value of 44083.7 samples/sec. Then i use this value in the screen drawing function as an actual rate. Now the problem disappears. The program is cross-platform, so i will test this measurements on windows and another soundcard later.

But is there a better way to sync sound and screen? Is there some not very CPU-consuming way of asking soundcard about actual playing sample number, for example?

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1 Answer

  1. Editorial Team

    This is a known effect, which is for example in Windows addressed by Rate Matching, described here Live Sources.

    On playback, the effect is typically addressed by using audio hardware as “clock” and synchronizing to audio playback instead of “real” clock. That is, for example, with audio sampling rate 44100, next video frame of 25 fps video is presented in sync with 44100/25 sample playback rather than using 1/25 system time increment. This compensates for the imprecise effective playback rate.

    On capture, the hardware itself acts as if it is delivering data at exactly requested rate. I think the best you can do is to measure effective rate and resample audio from effecive to correct sampling rate.

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