How can i make 16000Hz sample from 44100Hz sample in a real-time stream?

Question

I use portaudio in a Cpp work. My signal model treats the only 16000Hz audio input and

When the First released my work, I don't need to use 44100 sample rate. It was just about 48000Hz microphone. So I resampled my signal like 48000 -> 16000 -> 48000 with a simple decimation algorithm and linear interpolation.

But now I want to use a 44100 microphone. In real-time processing, My buffer is 256 points in 16000 Hz. So it is hard to find the input buffer size in 44100 Hz and downsample from 44100 to 16000.

When I used just decimation or average filter(https://github.com/mattdiamond/Recorderjs/issues/186), the output speech is higher then input and windowed sinc function interpolation makes a distortion.

is there any method to make 44100->16000 downsampling for realtime processing? please let me know...

thank you.

Answer 1

I had to implement a similar problem in the past, not for audio, but to simulate an asynchronism between a transmitte signal sampling frequency and a receiver sampling frequency.

This is how I will proceed:

Let us call T1 the sampling time duration of the incoming signal x: T1=1/44100 and let us call T2 the sampling time duration of the signal to be generated y.

To calculate the value of the signal y[n*T2], select the two input values x[k*T1]and x[(k+1)*T2] that surround the value to be calculated:

k*T1 <= n*T2 < (k+1)*T1

Then perform a linear interpolation from these two values. The interpolation factor must be recalculated for each sample.

If t = n*T2, a = k*T1 and b = (k+1)*T2, then

p = (x[b] - x[a])/T1
y[t] = p*(t-a) + x[a]

With a 44.1kHz frequency, x|a]and x[a+T1] should be rather well correlated, and the linear interpolation could be goood enough.

With the obtained quality is not good enough, you can interpolate the incoming signal with a fixed interpolation ratio, for example 2, with a classical well defined good interpolation filter.

Then you can apply the previous procedure, with the help of the new calculated signal, the sampling duration of which is T1/2.

If the incoming signal has some high frequencies, then, in order to avoid aliasing, you need to apply a low-pas filter to the incoming signal prior to the downsampling. Note that this is necessary even in your previous case 48kHz -> 16kHz