"Hybrid" <l_tofastrud@yahoo.com> writes:> When replying you could at least read my posting first... > >>On Wed, 07 May 2008 15:48:42 -0500, Hybrid wrote: >>> Even if the question is why DVD-A uses 192kHz I'd like to add one > thing >>> that is important in audio recording: Latency >> >>Not especially usefully. One sample period at 48kHz is about 7mm through > >>the air, so one at 192kHz is a little under 2mm. > It's not like the audio passes through only one A/D converter during a > recording session. > >>Now where are you >>sitting in your listening room, and how many samples are there between >>your ears and the CD player? >> >>What's the latency induced by the delivery truck that got the bits on the > >>DVD-A from the pressing plant to the retail shop? >> >>We're not really dealing with a system with *any* significant latency, >>when talking about a DVD-A. > > *yawn* -ah yeah- read the posting you're replying to first... > >>> Latency is very important both >>> in live venue settings and in recording studios. >> >>Now this is a different situation, to be sure. But remember the 7mm, and > >>consider the distances involved in actually producing and listening to >>music in a live environment. Certainly latency can't be excessively >>large, but the difference caused by different sample rates is not in and > >>of itself significant. > > Just a few years ago it was no uncommon to see 5-10ms latency through > almost any typical digital effects box or loudspeaker management system. > This was an issue that has been reduced a great deal thanks to higher > sampling rates _and_ the technology that came along with that. > Now imagine that a live situuation had to process the audio through 5-10 > such digital devices and the latency will make it difficult to play on > stage.I don't think you got Andrew's point. None of this is relevent for the storage, transmission, and reproduction of consumer audio, which is clearly the topic here. Latency on the order of even hundreds of milliseconds is not really an issue. Professional audio is another realm, and there could be applications that do benefit from oversampling for the reasons you state. However, the storage, transmission, and reproduction of consumer audio do not.> Another benefit with higher sampling rates is reduced aliasing distortion > - of course most of this is caused by engineers and mastering engineers > that have no idea how bad it is to use digital clipping to increase > "loudness".Again, non sequitur. Any such oversampling can be performed in the recording process and the signal post-converted back to simple 44.1 kHz for standard CD distribution while still retaining the advantage you refer to. -- % Randy Yates % "Watching all the days go by... %% Fuquay-Varina, NC % Who are you and who am I?" %%% 919-577-9882 % 'Mission (A World Record)', %%%% <yates@ieee.org> % *A New World Record*, ELO http://www.digitalsignallabs.com
What's the use of a 192 kHz sample rate?
Started by ●May 3, 2008
Reply by ●May 8, 20082008-05-08
Reply by ●May 8, 20082008-05-08
On May 8, 2:59 am, John Phillips <news0...@DontUseThis.mainly.me.uk> wrote:> On 2008-05-07, Steven Sullivan <ssu...@panix.com> wrote: > > > ... It seems extremely likely, for example, that no one > > can actually *hear* frequencies above the mid-20 kHz. > > They can be perceived via bone conduction, if the signal > > is generated right at the skin surface. > > This seems to be confirmed in Oohashi's 2006 "Hypersonic effect" paper > [1] which seems to show that the "inaudible high frequeny content" > of his test material is not perceived directly through the ear but is > perceived through the body surface.As is often the caxse, it's interesting to note the background to such reports. Oohashi's peper was presented to the AES, but met with a great deal of skepticism centered around the statistical methods, sound pressure levels and, especially, the apparent rejection of data that did not support the premise. (Now, one might argue that it was rejected because of it's "radical" findings that were "contrary to the establishment" and all that but given the review voard on the AES AND the level of criticism that was levelled at the papers basic techniques, this criticism seems a bit wide of the mark). It's also interesting to note that a major portion of the funding and sponsorship of the project came from one or more manufacturers who had a direct financial stake in the success of high sample-rate technology at the consumer level The point being that Oohashi "confirmed" nothing of the sort, especially considering the results, thus far, have not been reproduced by any independent researcher. The "bone conduction" hypothesis is a favorite piece of fairy dust often invoked in the high-end audio realm as a means of trying to justify high sample rates. All of these "hypotheses," even Oohashi's, if it is ever reproduced, all suffer from any number of serious PHYSICAL problems: the attenuation of high frequencies in air, the severe attentuation of high frequencies in the soft tissue that sits between and around any potential source and the proposed detection aparatus, the almost TOTAL lack of these signals on ANY recordings whatsoever, the almost TOTAL lack of these signals issuing forth from mechanical transducers. and on and on and on.
Reply by ●May 8, 20082008-05-08
John Phillips wrote:> On 2008-05-07, Steven Sullivan <ssully@panix.com> wrote: >> ... It seems extremely likely, for example, that no one >> can actually *hear* frequencies above the mid-20 kHz. They can be >> perceived via bone conduction, if the signal is generated right at >> the skin surface. > > This seems to be confirmed in Oohashi's 2006 "Hypersonic effect" paper > [1] which seems to show that the "inaudible high frequeny content" > of his test material is not perceived directly through the ear but is > perceived through the body surface. > > In this development from his 2000 paper [2] he seems to show that his > subjects' highest comfortable listening level is increased by the > presence of inaudible content when not applied just to the ear. > > [1] http://linkinghub.elsevier.com/retrieve/pii/S0006899305019499 > > [2] http://jn.physiology.org/cgi/content/full/83/6/3548Does he call the effect "bone conduction"? Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●May 8, 20082008-05-08
Hybrid wrote: ...> Just a few years ago it was no uncommon to see 5-10ms latency through > almost any typical digital effects box or loudspeaker management system. > This was an issue that has been reduced a great deal thanks to higher > sampling rates _and_ the technology that came along with that. > Now imagine that a live situuation had to process the audio through 5-10 > such digital devices and the latency will make it difficult to play on > stage.Would you mind explaining how and in what circumstances raising the sample rate reduces latency? For a low-pass FIR filter at a given frequency -- not a given fraction of the sample rate -- the latency remains the same because the number of taps is proportional to sample rate. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●May 8, 20082008-05-08
Hybrid wrote: ...> Just a few years ago it was no uncommon to see 5-10ms latency through > almost any typical digital effects box or loudspeaker management system. > This was an issue that has been reduced a great deal thanks to higher > sampling rates _and_ the technology that came along with that. > Now imagine that a live situuation had to process the audio through 5-10 > such digital devices and the latency will make it difficult to play on > stage.Would you mind explaining how and in what circumstances raising the sample rate reduces latency? For a low-pass FIR filter at a given frequency -- not a given fraction of the sample rate -- the latency remains the same because the number of taps is proportional to sample rate. ... Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●May 8, 20082008-05-08
On Thu, 08 May 2008 12:12:52 -0400, Jerry Avins <jya@ieee.org> wrote:>Hybrid wrote: > > ... > >> Just a few years ago it was no uncommon to see 5-10ms latency through >> almost any typical digital effects box or loudspeaker management system. >> This was an issue that has been reduced a great deal thanks to higher >> sampling rates _and_ the technology that came along with that. >> Now imagine that a live situuation had to process the audio through 5-10 >> such digital devices and the latency will make it difficult to play on >> stage. > >Would you mind explaining how and in what circumstances raising the >sample rate reduces latency? For a low-pass FIR filter at a given >frequency -- not a given fraction of the sample rate -- the latency >remains the same because the number of taps is proportional to sample rate. > >JerryIf there are block-oriented processes, like FFTs, in the chain then increasing the sample rate can help even if the processing rate of the block isn't improved. This is because the block doesn't have to wait as long to get filled, and can therefore generate an output more quickly. There's a similar situation that happens in carrying comm traffic with latency-sensitive applications (like voice). If the data rate of the signal is low the latency suffers, and one way to fix it is to multiplex the sensitive channel into a stream with a much higher bit rate. It doesn't eliminate delays, but it can reduce them substantially. Eric Jacobsen Minister of Algorithms Abineau Communications http://www.ericjacobsen.org
Reply by ●May 8, 20082008-05-08
>Would you mind explaining how and in what circumstances raising the >sample rate reduces latency? For a low-pass FIR filter at a given >frequency -- not a given fraction of the sample rate -- the latency >remains the same because the number of taps is proportional to sample >rate.Lowering latency in the converters will make it easier to use latency introducing FIR or look ahead limiters. It will not make FIR filters "faster". The reason for my posting was a series of comments at the beginning of the thread that stated that 192kHz was just a marketing thing to make people pay more money. In reality there is no reason why pro-audio and consumers should have different sampling rates. When storage becomes cheaper and 24/192 should be the standard. Why not? L
Reply by ●May 8, 20082008-05-08
Steven Sullivan wrote:> *Maybe* your friend was imagining things.Exactly! I was close to say that *maybe* the Earth is flat. With this kind of reasoning, everything is possible. I tell you another story. Once I met another guy, responsible of the quality in one audio CD factory. He told me that, looking for quality, they asked also audiophiles about what they think. It turned out that some people buy 2~3 copies of the same audio CD, they "test" it at home, and then keep the one that "sounds better". So, he was asking what could it be, are there any difference between CDs of different batches, maybe. They told him something about jitter, it seems different CDs can have different jitter. He, then, took some high end CD player and measured all he could measure, including jitter, of different copies of the same CD(s). He could not find anything strange, the high end CD player has FIFOs, which keep the jitter constant and minimum (low end is different story, but audiophiles use high end). In the end he could not figure it out what was the reason why different copies of the same CD should sound differently. Next reply I'll tell you another of this stories. bye, -- piergiorgio
Reply by ●May 8, 20082008-05-08
Steven Sullivan wrote:> *Maybe* your friend was imagining things.Exactly! I was close to say that *maybe* the Earth is flat. With this kind of reasoning, everything is possible. I tell you another story. Once I met another guy, responsible of the quality in one audio CD factory. He told me that, looking for quality, they asked also audiophiles about what they think. It turned out that some people buy 2~3 copies of the same audio CD, they "test" it at home, and then keep the one that "sounds better". So, he was asking what could it be, are there any difference between CDs of different batches, maybe. They told him something about jitter, it seems different CDs can have different jitter. He, then, took some high end CD player and measured all he could measure, including jitter, of different copies of the same CD(s). He could not find anything strange, the high end CD player has FIFOs, which keep the jitter constant and minimum (low end is different story, but audiophiles use high end). In the end he could not figure it out what was the reason why different copies of the same CD should sound differently. Next reply I'll tell you another of this stories. bye, -- piergiorgio
Reply by ●May 8, 20082008-05-08
Eric Jacobsen wrote:> On Thu, 08 May 2008 12:12:52 -0400, Jerry Avins <jya@ieee.org> wrote: > >> Hybrid wrote: >> >> ... >> >>> Just a few years ago it was no uncommon to see 5-10ms latency through >>> almost any typical digital effects box or loudspeaker management system. >>> This was an issue that has been reduced a great deal thanks to higher >>> sampling rates _and_ the technology that came along with that. >>> Now imagine that a live situuation had to process the audio through 5-10 >>> such digital devices and the latency will make it difficult to play on >>> stage. >> Would you mind explaining how and in what circumstances raising the >> sample rate reduces latency? For a low-pass FIR filter at a given >> frequency -- not a given fraction of the sample rate -- the latency >> remains the same because the number of taps is proportional to sample rate. >> >> Jerry > > If there are block-oriented processes, like FFTs, in the chain then > increasing the sample rate can help even if the processing rate of the > block isn't improved. This is because the block doesn't have to wait > as long to get filled, and can therefore generate an output more > quickly.An FFT of the same bin separation needs to be twice as long, so where's the improvement?> There's a similar situation that happens in carrying comm traffic with > latency-sensitive applications (like voice). If the data rate of the > signal is low the latency suffers, and one way to fix it is to > multiplex the sensitive channel into a stream with a much higher bit > rate. It doesn't eliminate delays, but it can reduce them > substantially.Doesn't that amount to doing more tasks at the same time? Why does that speed any one task? Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
