Hi Erik, As you can see, we made it back after chasing Ivan around. Luckily we saw nothing but a few drops of rain. I'm sure others here or close to those here must have had bad luck instead of good, and my sympathies go out to them. Thanks for your response. I applaud your enthusiasm, but it appears to me that you are laboring under some misconceptions and misunderstanding of what I have posted. Among the misconceptions is a serious one that should cause concern amongst those who use libsamplerate. ------------------------- Minor problems: A good portion of your recent post is aimed at something I never claimed. It appears to me that you are failing to distinguish between two cases: 1) Series in general; 2) Series which have been previously prepared, specifically frequency spectra which are band- limited and lowpass filtered. >From my earlier post on the current subject: "The reason is that I assume that the input is band-limited, and this is usually true for my own work. Not only is it band-limited, but usually also tapered in the frequency domain, i.e. already effectively lowpass filtered." >From your recent post: "Now I will admit that if the signal is already bandlimited filter many not be necessary, but that is a different matter all together." No, it is certainly not a "different matter altogether." Your quote was nearly all I was saying as you can easily see, except that in addition signals I use (as well as most that others use) are also normally effectively lowpass filtered. This is more restrictive than your assumption, so filtering is even less necessary than you admit it to be. Your elementary examples from undergrad engineering courses and attempt to rephrase what you claim I said don't apply to the situation I've described; they have nothing to do with what I've said and are true but irrelevant statements which cause me to wonder what it is that you are trying to prove; you may want to give some thought to that: What is it that you are actually trying to prove or demonstrate to everyone? As a very minor note, you are also attempting to answer an ontological question (that something which has no observable effect on the universe can nonetheless be said to exist) which has plagued philosophers for millenia by merely declaring your opinion as fact. Here is what I actually wrote: "If nothing was removed or even altered, then no filtering has actually occurred." Note that I am not so reckless as to claim anything regarding existence or nonexistence here. ------------------------- ** Serious misconception **: Two paragraphs from your recent post: "If you work out the mathematical expression for your frequency domain converter, you will find that there is a time domain expression that is mathematically identical and that the time domain expression is in part a FIR lowpass filter very much like my converter." (You mean your implementation of Smith's converter, don't you?) "The difference between the two would be that my version uses linear interpolation into a very large table to obtain the filter coefficients while yours are more exact. However mine provides a *measured* SNR of at least 97dB." This second paragraph is considerably flawed and reveals a serious misconception on your part. Although the *starting point for the derivation* of both methods is the same, the method you are using as it is actually implemented is a very localized method in which only few samples are used to construct the new values. This is due to the *severe* truncation of the series. The horrible effects of this severe truncation are mitigated through the application of a Kaiser window which effectively localizes the estimation even further as it improves the result substantially. The method I'm using remains a very much more global method in which each new value is constructed from hundreds of thousands to millions of samples. Given that audio I process taper off to zero at the beginning and end --- in addition to being band-limited and effectively lowpass filtered --- I could have written a global method wherein each sample is the result of calculations involving ALL other samples rather than one which contains windows with hundreds of thousands to millions of samples. This is a far cry from what you are doing. This is all BEFORE the linear interpolation step which you claim is the only difference between our methods, a step which is totally unnecessary by the way for most fixed-sample-rate conversions (for example: unnecessary for 96,000 to 44,100, for 48,000 to 44,100, and for 44,100 to either of the others) and further degrades the sample rate conversion unnecessarily for the most common use. (This was done by Smith to permit variable-sample-rate conversions, a useful goal to say the least.) Now as I've already said, the sinc-based sample-rate converter you've constructed from Smith's work seems to work well in the sample I've listened to. But knowing what I know about sinc-based methods based on my own implementations, Smith's published work, and the lack of necessity for such a method for fixed-rate conversions, not to mention the fact that linear interpolation is being used where it really isn't necessary, leads me to reject libsamplerate and anything related to it unless it's forced on me by someone else via a dependency. I rarely do variable-rate conversions, so I have very little need for sinc- based sample-rate converters. ------------------------- For "ordinary" users: For those of you who have read this far --- hopefully not many: PLEASE don't be alarmed. As I have repeatedly said, Erik's implementation of Smith's work seems to work well. The inaccuracies in that method for fixed rate conversions, including the additional inaccuracy of the unnecessary linear interpolation, seem to be virtually inaudible. The main advantage is that a single method can be used for both fixed- and variable-rate conversions, and this is indeed a *considerable* advantage for a lib-based implementation. We are all indebted to Erik for creating libsamplerate, and this includes me. ------------------------- For developers: Developers, however, should at least be aware of what is going on there. I would advise anyone who has a *critical* dependence on sample-rate conversions to carefully read Smith's notes and to make sure that they understand what is going on. They should also experiment with Smith's technique, including implentation of different windows to see how bad it can be if things go wrong. Erik's "measurements" that I've seen so far aren't very useful for these worst-case situations. Smith may have had something more useful such as a bound of some sort, but frankly I don't recall at this moment and I'm not motivated to check it out at this time (sorry!). I have too much to do following this trip, and this post has taken all my spare time. Sorry for the length, but I wanted to clear up some misconceptions about what I've posted as well as raise a flag (without causing some sort of fiasco) for those who may not have had time to study sinc-based methods. Regards to all, Dave.
