MOTUNATION (formerly UnicorNation) is an independent community for discussing Digital Performer and other MOTU audio software and hardware. It is not affiliated with MOTU.
https://www.motunation.com/forum/
Check out the link that I posted in my first reply.FMiguelez wrote:Wait! Now that I remember, didn't Shooshie post a website link where they compare different DAW's SRC algorithms?
If memory serves me well, DP scored fairly high in there...
I'll see if I can dig it up and find it. The thing is that it wouldn't include DP7, but only DP5 or DP6.
It all depends on the SRC algorithms... some are better written than others and cost more etc. because of the time and efforts involved.FMiguelez wrote:.
Hey, Newrigel.
I'd like to understand this a bit better...
If I do a SRC from 192 to 44.1, what happens to all that "extra" information of the original file? Does it get truncated?
Would it be a good thing to put a LPF around 22 KHz just before I do the SRC? Or that is exactly what SRC does anyway?
Example
CDs are sampled at 44.1 kHz, but a Digital Audio Tape, or DAT is usually sampled at 48 kHz. How can material be converted from one sample rate to the other?
First, note that 44.1 and 48 are in the ratio 147/160. If the original audio signal had been recorded at 7.056 MHz sampling rate, the process would be simple. Since 7.056 MHz is 160 x 44.1 kHz, and also 147 x 48 kHz, all we would need to do is take every 160th sample to get a 44.1 kHz sampling rate, and every 147th sample to get a 48 kHz sampling rate. Taking every Nth sample like this preserves the content provided the information (the audio signal) does not have any content above half the lowest sampling rate used (22.05 kHz) in this case.
So now the problem is how to generate the 7.056 MHz sampled signal, given that the original has only 1/160 of the samples needed. The somewhat surprising answer is to replace the missing samples with zeros. So if the original audio samples were ..,a,b,c,.., then the 7.056 MHz sequence is ..,a,0,0,0,...0,0,b,0,0...0,0,c,.., with 159 zeros between each original sample. This too will create extra high frequency content (in fact it is worse in this respect than linear interpolation) but at least the frequency response is flat. Then the digital filter removes the unwanted high frequency content. The work of this digital filter is also much easier if zeros are inserted, since the filter is basically an average and almost all of the samples are known to be zero.
So inserting the zeros, then running the digital filter (almost always an FIR filter since these can be designed to have no phase distortion) gives the needed signal - sampled at 7.056 MHz, but with no content above 24 kHz. Then just taking every 147th sample gives the desired output. Which sample to start with does not matter - any set will work as long as they are 147 samples apart.
