RE: [audiodsp] conversion | Speech Coding

The answer from harishankar is not quite correct. Theory tells the following: Re-sampling from 44100 Hz to 48000 Hz: 1) Up-sample by expansion factor of 160 via zero insertion (sampling frequency is now 7.056 MHz!!) 2) Run up-sampled signal through a low-pass filter with cutoff 22050 Hz. This filter combines the interpolation filter required by the up-sampling process and the anti-aliasing filter for the following down-sampling process. Pass-band gain of this filter must be equal to the expansion factor of 160. 3) Down-sample filter output by compression factor of 147 (sampling frequency is now 48000 Hz). Re-sampling from 48000 Hz to 44100 Hz: 1) Up-sample by expansion factor of 147 via zero insertion (sampling frequency is now 7.056 MHz!!) 2) Run up-sampled signal through a low-pass filter with cutoff 22050 Hz. This filter combines the interpolation filter required by the up-sampling process and the anti-aliasing filter for the following down-sampling process. Pass-band gain of this filter must be equal to the expansion factor of 147. 3) Down-sample filter output by compression factor of 160 (sampling frequency is now 44100 Hz). The problem with this theoretical approach is the high intermediate sampling frequency of 7.056 MHz at which the low-pass filter needs to run. If we assume that this filter would need 100 cycles per sample, required processing power just for running the filter alone would reach 700 MIPS!! So I would think that there must be a more practical approach for this particular sampling frequency conversion. Does anyone know? Regards, Roland -----Original Message----- From: Simon Magus [mailto:] Sent: Wednesday, May 28, 2003 7:52 AM To: harima; ; ; ; matlab; Subject: Re: [audiodsp] conversion harima <> wrote:Hi there, How can we do CD(44.1kbps) <---> DAT(48kbps) sample rate conversion. Thanks in advance, ciao, harishankar _________________________________________________________ There is a subtopic on DSP called multirate signal Processing. From 44.1k to 48kHz you need to multiply by 480/441. That means you have to upsample by 480 and then downsample by 441. Upsampling is done by inserting 479 zeros in between the samples and the a low pass filtering with cutoff at 44100*480Hz. Downsampling is done by low pass filtering with cutoff at 48kHz and then removing 440 samples for every 441 samples Yahoo! Groups Sponsor _____________________________________

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Discussion Groups | Speech Coding | RE: [audiodsp] conversion

RE: [audiodsp] conversion - Author Unknown - May 28 15:07:00 2003