On Mon, 15 Dec 2008, Paulo Moura Guedes wrote: > On Friday 12 December 2008 20:34:21 Sergei Steshenko wrote: >> On Fri, 12 Dec 2008 20:24:25 +0000 >> >> Paulo Moura Guedes <moura@xxxxxxxxxxxxx> wrote: >>> I don't know about ALSA. >> >> So, ALSA, as well as many Linux applications, have a choice of qualities >> for resampling, and I bet you won't hear the difference between middle and >> high quality. >> >> There is no "OS resampling", resampling is a well researched field, and >> resampling quality does not depend on OS, it depends on the chosen >> algorithm. >> >> Again, no serious sound recording company records at CD sample rate, they >> record, I think, at 96kHz - the ADCs at 192kHz do not perform that well. >> >> So, every CD you listen to is resampled. > > That's correct, CD quality is 16bit 44.1kHz, which should be much lower as the > recording rate as you said. > Nowadays, it's not so dificult to find (pay) for files with 24bit 96kHz, but that > is the result of resampling as well. > There is nothing one can do about that though... only make it worse, which is > what I'm trying to avoid ;) > > BTW, Benchmark DAC1 resamples internally to 110kHz: > > "The process of upsampling does not inherently improve sound quality during > D/A conversion. However, Benchmark converters re-sample for a very specific > reason: jitter immunity. Benchmark converters use a proprietary clocking > system (we refer to it as UltraLock). It works like this... The incoming > digital signal is immediately re-sampled by an ASRC (asyncronous sample rate > converter). The ASRC, as the name implies, is not syncronized to the clock of > the incoming digital signal. Therefore, its performance is independant of the This makes no sense at all. If the incoming signal is a digital signal, it has no clock. It is a series of bytes which are by definition once every way 1/44100 sec. > quality of that clock. In other words, it doesn't matter if the signal came > from a cheap transport with cheap cables, or from a $10,000 signal chain. The And what the hell do cables have to do with anything? Cables do NOT affect the timing. (and cheap cables transmit signals at even 96KHz as well as expensive ones do.) This sounds like a bunch of incompetent gobbledygook whose only purpose is to impress the pigeons. > large amount of jitter caused by the cheap transport and cheap cable will be > moot with respect to the ASRC process. The output of the ASRC is then clocked > to an on-board clock with extremely low jitter and strategic sheilding and > board traces. The output of the ASRC can be configured to any sample rate that > we choose, including the original sample rate. However, we dictated the re- > sample rate as 110 kHz because it is the highest sample-rate at which the > digital interpolation filter of the D/A chip will operate optimally. The ill- > effects of the digital interpolation filter at higher-then-110 kHz include pass- > band ripple (non-linearities in frequency response) and inferior attenuation > of stop-band frequencies (which results in aliasing). Therefore, the D/A > performance is optimized by maintaing 110 kHz. Many converter designers have > since employed similar topologies, but use lower re-sampling frequencies, such > as 96 kHz. By resampling to 110 kHz, the low-pass filter of the ASRC and D/A > are moved as far up as possible as to not infringe on the analog bandwidth of > the audio" > > "On the question of: Why does the DAC1 re-sample to 110 kHz? Here is why: it > is the highest frequency to maintain the full oversampling of the D-A chip. > EVERY D-to-A chip on the market cuts the oversampling rate in half to > accommodate 192 kHz. This will also implement a different type of digital low- > pass filtering which is inferior to the filter used at and below 110kHz. This is > also why most recording engineers don't use 192 kHz. The higher bandwidth > seems appealing, but the stat-of-the-technology is such that 192 kHz > conversion is actually inferior to 96 kHz. Also, the DAC1's oversampling ASRC > and resulting 110 kHz sample rate reproduces 96 kHz signals much more > faithfully then a D-A converting the original 96 kHz signal. This is because > the Nyquist frequency is on the slope of the filter (attenuated, but not > completely). This is undesirable for two reasons. The first reason is the > Nyquist frequency is not faithfully converted to analog (ie, the analog > bandwidth of 96 kHz conversion is actually less then 48 kHz). With the DAC1, > the full bandwidth of a 96 kHz signal can be faithfully reproduced. The second > problem with 96 kHz conversion is the frequencies at and above Nyquist (48 kHz > and up) are not completely attenuated, so some aliasing and imaging will OOOh. and this matter because there is so much sonic energy between 49KHz and 55KHz. Sheesh. > occur. With the 110 kHz upsampling and conversion in the DAC1, the frequencies > below 55 kHz are not in danger of being aliased." Since there are no frequencies above 20KHz this really matters a lot. > > Thanks, > Paulo I assume you spent $1000 on some oxygen free copper cables as well. ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
