Correctly decoding/encoding raw PCM data

Question

I'm writing my WAVE decoder/encoder in C++. I've managed to correctly convert between different sample sizes (8, 16 and 32), but I need some help with the channels and the frequency.

Channels:

If I want to convert from stereo to mono:

1. do I just take the data from one channel (which one? 1 or 2?)?
2. or do I take the average from channel 1 and 2 for the mono channel.

If I want to convert from mono to stereo: (I know this is not very scientific)

1. can I simply add the samples from the single channels into both the stereo channels?
2. is there a more scientific method to do this (eg: interpolation)?

Sample rate:

How do I change the sample rate (resample), eg: from 44100 Hz to 22050 Hz:

1. do I simply take the average of 2 sequential samples for the new (lower frequency) value?
2. Any more scientific algorithms for this?

Answer 1

• Stereo to mono - take the mean of the left and right samples, i.e. M = (L + R) / 2 - this works for the vast majority of stereo content, but note that there are some rare cases where you can get left/right cancellation.

• Mono to stereo - put the mono sample in both left and right channels, i.e. L = R = M - this gives a sound image which is centered when played as stereo

• Resampling - for a simple integer ratio downsampling as in your example above, the process is:

  • low pass filter to accommodate new Nyquist frequency, e.g. 10 kHz LPF for 22.05 kHz sample rate
  • decimate by required ratio (i.e. drop alternate samples for your 2x downsampling example)

Note that there are third party libraries such as libsamplerate which can handle resampling for you in the general case, so if you have more than one ratio you need to support, or you have some tricky non-integer ratio, then this might be a better approach