Im having problems with the reconstruction through two-channel filter bank when I use my audio files.
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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
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