I'm pretty new to the world of frequency domain as proven by this question: Is there any correlation between the quality of resampler required to make aliasing "imperceptible" and the magnitude of sampling rate change? I'm currently playing with various resampling algorithms, running some 44.1kHz content through them (with lots of high and low frequencies) and I noticed increasing the sampling rate change made it easier to hear the differences in quality between the various methods. Does this imply that as the sampling rate change decreases one can get away with a lower quality algorithm, or that I simply got lucky with my given content and ratios I happened to pick?
Perceived aliasing in small ratio SRC
Started by ●June 21, 2006
Reply by ●June 21, 20062006-06-21
Hi, quality depends on the following three
- Sampling resolution (no.of bits/sample)
- Sampling rate
- No. of channels sampled
sampling rate is directly proportional to quality.
(hopefully not misleading you)
- San
Reply by ●June 21, 20062006-06-21
somenoob wrote:> I'm pretty new to the world of frequency domain as proven by this > question: > > Is there any correlation between the quality of resampler required to make > aliasing "imperceptible" and the magnitude of sampling rate change? > > I'm currently playing with various resampling algorithms, running some > 44.1kHz content through them (with lots of high and low frequencies) and I > noticed increasing the sampling rate change made it easier to hear the > differences in quality between the various methods. Does this imply that > as the sampling rate change decreases one can get away with a lower > quality algorithm, or that I simply got lucky with my given content and > ratios I happened to pick?Unless the original signal is oversampled, resampling to a lower rate costs high-frequency response. If inadequately filtered, if will also create aliasing. A word about the methods and resampling ratios you use would likely get you a more complete answer. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●June 21, 20062006-06-21
sanindland wrote:> (hopefully not misleading you)Mostly, you are. Sampling rate must exceed twice the bandwidth. A reasonable (1.5 to 10, depending on the application) margin makes processing easier. Beyond that, no advantage. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●June 21, 20062006-06-21
> >Unless the original signal is oversampled, resampling to a lower rate >costs high-frequency response. If inadequately filtered, if will also >create aliasing. A word about the methods and resampling ratios you use >would likely get you a more complete answer. > >Jerry >--Thanks! I�m trying to figure out if I can use a less computationally expensive algorithm if the sample rate change is small. Say the original 44100Hz (containing some tones near 22k) stream is not oversampled and I resample to 44000Hz. Do I really need to use a 90+ tap windowed-sinc or is something like a cheap 3rd order Lagrange or even linear adequate to mitigate perceptible aliasing for such a small change?
Reply by ●June 21, 20062006-06-21
somenoob wrote:>> Unless the original signal is oversampled, resampling to a lower rate >> costs high-frequency response. If inadequately filtered, if will also >> create aliasing. A word about the methods and resampling ratios you use >> would likely get you a more complete answer. >> >> Jerry >> -- > Thanks! I�m trying to figure out if I can use a less computationally > expensive algorithm if the sample rate change is small. Say the original > 44100Hz (containing some tones near 22k) stream is not oversampled and I > resample to 44000Hz. Do I really need to use a 90+ tap windowed-sinc or > is something like a cheap 3rd order Lagrange or even linear adequate to > mitigate perceptible aliasing for such a small change?What platform will this run on? I assume some sort of embedded system, but if you can, just use Secret Rabbit Code (SRC also stands for Sample Rate Converter) from http://www.mega-nerd.com/SRC/. 44100 Hz sample rate is skimpy for 20 KHz, let alone 22. To keep 20 after downsampling to 44000, you will probably need humumgous filters. Going to 22050 is easy by comparison. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●June 21, 20062006-06-21
somenoob wrote:>>Unless the original signal is oversampled, resampling to a lower rate >>costs high-frequency response. If inadequately filtered, if will also >>create aliasing. A word about the methods and resampling ratios you use >>would likely get you a more complete answer. >> >>Jerry >>-- > > Thanks! I�m trying to figure out if I can use a less computationally > expensive algorithm if the sample rate change is small. Say the original > 44100Hz (containing some tones near 22k) stream is not oversampled and I > resample to 44000Hz. Do I really need to use a 90+ tap windowed-sinc or > is something like a cheap 3rd order Lagrange or even linear adequate to > mitigate perceptible aliasing for such a small change?I think, because of how you phrased your question, that you may be asking a slightly different question than Jerry answered. [I'm a fellow 'newbie' inspite of assertions by some ;] The keyword of interest is "Nyquist frequency". Sites of interest may include: http://www.answers.com/topic/nyquist-frequency-in-math http://www.dsptutor.freeuk.com/aliasing/AliasingDemo.html [I haven't used] http://en.wikipedia.org/wiki/Aliasing I thought the comp.dsp FAQ had reference, but I didn't find it. [Then again just got off long graveyard shift -- snore SNORE *SNORE*
Reply by ●June 21, 20062006-06-21
somenoob wrote:> Thanks! I�m trying to figure out if I can use a less > computationally expensive algorithm if the sample rate change is > small. Say the original 44100Hz (containing some tones near > 22k) stream is not oversampled and I resample to 44000Hz. Do I > really need to use a 90+ tap windowed-sinc or is something like > a cheap 3rd order Lagrange or even linear adequate to mitigate > perceptible aliasing for such a small change?Decimation quality presents a tradeoff between alias rejection and in-band treble conservation. You can always achieve a given stopband rejection by placing the filter's stopband edge at the new Nyquist frequency, instead of its passband edge as is commonly done. (Here, stopband edge means the least frequency where the filter attains the desired rejection, and passband edge is often taken to be the -3 dB or -6 dB frequency.) But doing so will increase damping of frequencies that are only just in band. To improve this measure (or both measures in the case of passband edge placement) you need a narrower transition band which requires higher filter order. There is no simple connection to the decimation ratio, though a given transition width may be harder to achieve around some frequencies than others. Martin -- There are some ideas so wrong that only a very intelligent person could believe in them. --George Orwell
Reply by ●June 21, 20062006-06-21
Richard Owlett wrote: ...> The keyword of interest is "Nyquist frequency". > Sites of interest may include: > http://www.answers.com/topic/nyquist-frequency-in-math > http://www.dsptutor.freeuk.com/aliasing/AliasingDemo.html > [I haven't used] > http://en.wikipedia.org/wiki/AliasingNyquist isn't the whole story. It takes just as long to resolve Fs - .01 a s it does to resolve .01 Hz. Few notes are held for 10 seconds. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●June 21, 20062006-06-21
>Decimation quality presents a tradeoff between alias rejection and >in-band treble conservation. You can always achieve a given stopband >rejection by placing the filter's stopband edge at the new Nyquist >frequency, instead of its passband edge as is commonly done. (Here, >stopband edge means the least frequency where the filter attains the >desired rejection, and passband edge is often taken to be the -3 dB >or -6 dB frequency.) > >But doing so will increase damping of frequencies that are only just >in band. To improve this measure (or both measures in the case of >passband edge placement) you need a narrower transition band which >requires higher filter order. There is no simple connection to the >decimation ratio, though a given transition width may be harder to >achieve around some frequencies than others. > >Martin > >--Thanks to all for your responses. While I�m trying to learn more about the theory -- I do believe my original question can be answered independently of my test implementation as those details appear to be misleading people on what I�m really after. All I want to know is if there is any correlation between the order of filter required to make aliasing imperceptible when resampling and the magnitude of sampling rate change. It sounds to me from Martin�s comment: �There is no simple connection to the decimation ratio� that the answer is �no�.
