Im having problems with the reconstruction through two-channel filter bank when I use my audio files.
Show older comments
Following these steps: https://es.mathworks.com/help/dsp/examples/reconstruction-through-two-channel-filter-bank.html
With the default audio, all works fine. But when I use my own audio files (audioInput = dsp.AudioFileReader('random.mp3')), the hi and lo coefficients are 0. Any idea?
This is my code, for the analysis part:
audioInput = dsp.AudioFileReader('filename.mp3'); % read the audio file
N = 99; % filters order
[LPAnalysis, HPAnalysis, LPSynthsis, HPSynthesis] = firpr2chfb(N, 0.45);
analysisFilter = dsp.SubbandAnalysisFilter(LPAnalysis, HPAnalysis);
while ~isDone(audioInput)
input = audioInput(); % Load a frame of audio
[hi, lo] = analysisFilter(input);
end
end
Accepted Answer
More Answers (0)
Categories
Find more on Audio Processing Algorithm Design in Help Center and File Exchange
Community Treasure Hunt
Find the treasures in MATLAB Central and discover how the community can help you!
Start Hunting!