I deduced my own answer. When audiowrite is given int32 data to write as 24-bit samples, it considers the data to have 32 bits and drops the least significant 8 bits. Thus, 24-bit data should be multiplied by 2^8 before sending to audiowrite. Inserting a scaling statement into the code
for bitsPerSample = [32 24]
% Generate random integers that fit within the sample width
B = 2^(bitsPerSample - 1);
xlims = [-B, B-1];
x = randi(xlims, [100 1], 'int32');
if bitsPerSample == 24, x = x * 2^8; end % Scale 24-bit data to 32-bit limits
% Write data to and read data from file
audiowrite('tmp.wav', x, 8000, 'BitsPerSample', bitsPerSample);
info = audioinfo('tmp.wav');
y = audioread('tmp.wav', 'native');
% Compare written and read data
fprintf('Bits per sample is %d\n', bitsPerSample)
fprintf('CompressionMethod is ''%s''\n', info.CompressionMethod)
fprintf('Std. deviation between written and read data is %.1f\n\n', ...
std(double(x) - double(y)))
end
eliminates the precision loss:
Bits per sample is 32
CompressionMethod is 'Uncompressed'
Std. deviation between written and read data is 0.0
Bits per sample is 24
CompressionMethod is 'Uncompressed'
Std. deviation between written and read data is 0.0
