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5th March 2013, 19:57 | #1 | Link |
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ReClock and Audio Discussion
For people using Reclock: What is the optimum sample rate to set in Reclock's configuration settings? The Realtek chip on my motherboard supports 24-bit, 192Khz so I have 24-bit padded to 32-bit and 192Khz set in Reclock but I'm not sure if that's how it should be.
Also, has anyone experienced problems with the audio "cutting" when using PAL slowdown with time stretch and 192Khz? I have to set the output sample rate in Reclock to 96Khz in order to get a consistent audio stream... |
5th March 2013, 21:22 | #2 | Link | |
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From experience I recommend disabling time stretching as it often causes audible artifacts. |
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5th March 2013, 21:44 | #3 | Link |
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How to disable reclock?
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6th March 2013, 13:35 | #4 | Link | |
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Last edited by iSunrise; 6th March 2013 at 14:07. |
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6th March 2013, 14:55 | #5 | Link | |||
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The sweet spot is prolly 88.2kHz if your audio device supports it. This said, those measurements were made with the old Reclock resampler and that will depend on the oversampling rate of your DAC chip as well(only the AKM chips such as AK4396 can go 128X all the way up to 192kHz, most other chips will fall back to 64X at 96kHz and 32X at 192kHz). It's all discussed there really: http://www.audioholics.com/education...-digital-audio Ideally you wanna keep the oversampling rate as high possible and yet upsample one "notch" higher than the input sample rate so 88.2kHz is prolly the sweet spot. 96kHz will do if your gear doesn't support it. Dan Lavry confirms it too: The Optimal Sample Rate for Quality Audio Quote:
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Last edited by leeperry; 7th March 2013 at 04:29. |
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6th March 2013, 15:50 | #6 | Link | |
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88.2kHz/96kHz in that case. I´m not that familiar with reclock, since I never really felt the need for it. I´m fine with my native 23,976p, 24p, 47,8p, 48p, 50p, 59,9p and 60p modes on my Eizo CG243W. I simply cannot see the extremely small deviations, even if I know that they are there. Last edited by iSunrise; 6th March 2013 at 16:02. |
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6th March 2013, 22:01 | #8 | Link | ||||
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So, thanks for proving my point. Quote:
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You really need to stop spreading such misinformation and start putting your measurements and the documents you're quoting in context before coming up with issues that don't exist. Last edited by e-t172; 6th March 2013 at 22:03. |
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6th March 2013, 22:18 | #9 | Link |
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It does if you enable its frame rate adaptation(and it will resample 48kHz to 47952Hz if you play 23.976fps content in a 24.000Hz multiple), but this is completely irrelevant to this thread I'm afraid. There are people to whom everything sounds/looks the same(jitter is inaudible, yada yada)...be it then, lucky them. I wish it all sounded the same to me too.....I really do.
Last edited by leeperry; 7th March 2013 at 04:30. |
6th March 2013, 22:30 | #10 | Link | ||||
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7th March 2013, 01:28 | #11 | Link | |
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I see how it can get confusing. Here's the explanation: when I say that you don't need more than 48kHz, I mean that sample rates higher than 48kHz do not convey any additional audible information.
The reason why a DAC is oversampling is because you need a low-pass filter in the analog stage to remove the high-frequency aliasing artifacts from the Sigma-Delta demodulation. It's much easier to do that with a high sample rate because the aliasing artifacts are higher in frequency, so you don't need a steep filter to filter them out, so it's easier (and cheaper) to build. If the DAC was working at 48kHz, it would need a filter with near-zero response at 20kHz (top of the audible range) but high attenuation above 24kHz (Nyquist frequency) which is an extremely steep analog filter and so difficult to achieve. On the other hand, if the DAC is working at a higher sample rate then it's not an issue. That's why it's always oversampling internally, no matter what sample rate you use for the input signal. But again, it's an implementation detail. It's about preventing aliasing artifacts in the DAC process, not preserving information that is not audible anyway. As an example, a quick Google search returns an Analog Devices paper which describes how this works (see figure 6.34, page 12). In this example there is a 8kHz input (pretty Low-fi, heh) which is oversampled to 1 MHz using a digital filter, then modulated in delta-sigma, then converted to analog, then low-passed by a simple analog filter. Here the analog filter is simple to implement because the conversion is done at 1 Mhz, which leaves plenty of headroom above the 4kHz Nyquist frequency of the original signal. Quote:
It's oversampling by 8x, meaning that a 48 kHz signal will get oversampled to 384 kHz by interpolating, then low-passing in the digital domain. For audio one would almost certainly use the sharp roll-off mode for the lowpass filter, so according to page 4 you get a flat response up to about 21.8 kHz, no issue here. Then it gets modulated (probably some kind of Delta-Sigma modulator), converted to analog (DAC), and then it goes through an analog filter which doesn't seem to be specified (probably because no-one cares, it's not relevant to overall performance) but I would guess it's a low-pass filter with a pass-band extending up to fS/2 (Nyquist frequency of the original input signal) and a stop-band beginning at fS*4 (Nyquist of the modulated signal). If you're feeding it with 96 kHz, then you can basically multiply each of these numbers by 2, although I don't see the point. Last edited by e-t172; 7th March 2013 at 02:17. |
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7th March 2013, 07:07 | #12 | Link |
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Okay, some additional questions:
What is the optimal setting for "Format" in ReClock? Currently, I have it set to 24-bit padded to 32-bit because setting it to 24-bit integer results in ReClock giving me an error saying that the audio device doesn't support it even though the Realtek chip is stated to support 24-bit. Maybe that implies padding to 32-bit? Suppose, I'm NOT using WASAPI: What should I set for the "Default Format" under the speaker's properties page in Windows? This is confusing for me because doesn't this dictate the frequency output of computer generated sounds as well? Like certain software generated tones (not pre-recordings). Should the sample rate still be 48000 Hz= here? |
7th March 2013, 09:44 | #13 | Link | |||||
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As we looking for a perfection, we can't avoid the question here: what is the difference? As we know, ReClock provides the different quality options for its own resampling. And what do we know about dac's oversampling algorithm(s)? Hmmm... Is there a chance that ReClock provides a better resampling algorithm? So we can prepare a sound in best possible quality in ReClock and then just simple: Quote:
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The main ideas: - ReClock and windows sound mixer operate in float, while audio driver and sound devices accepts integer. - convertion from float to integer must be done at the very final stage to keep more accurasy. - the higher bitdepths keep more accurasy. |
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7th March 2013, 14:10 | #14 | Link | ||
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Regarding ReClock, I don't know what the quality options relate to in terms of lowpass FIR filter length, so it's difficult to answer. We would need to know the windowing function as well. I wouldn't be surprised, however, if the difference between the different quality options was extremely difficult (if not impossible) to hear, as it mainly affects the very top end of the frequency response. If you're using the best quality option then it is extremely likely that using a 96kHz output sample rate in ReClock is just a waste of CPU time. It would be possible to verify this by playing a test signal through ReClock and then intercepting it before it gets to the audio output (e.g. using Virtual Audio Cable), but I don't have time to do that. Last edited by e-t172; 7th March 2013 at 14:12. |
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7th March 2013, 15:45 | #15 | Link |
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How i always saw upsampling with ReClock was to give it some more data to work with, because it always needs to resample anyway.
So i figured, if it needs to resample from 48000 to 48050 because of speed adjustments, why not tell it to resample to 96100 instead, and hopefully have simply more samples where it can put its data, possibly hiding some distortions (not that i'm likely to hear them)
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7th March 2013, 16:19 | #16 | Link | |
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There is no question that it's better to upsample by 99% than to downsample by 0.1%. It's measurable too Last edited by leeperry; 7th March 2013 at 16:21. |
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7th March 2013, 19:10 | #19 | Link | |
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I use a $3K DAC, highly transparent orthodynamic headphones and I've got fairly trained ears as I've worked as an audio engineer for quite a while...so please be so kind as to not make gross generalizations of your personal experience. I've got nothing to sell and I can't be dragged into endless and pointless internet debates, so moar power to ya if all sounds the same to your ears....total harmonic distortion measurements that prove me right don't mean anything, that makes perfect sense |
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7th March 2013, 20:04 | #20 | Link | |
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0.00024% THD? Do you have alien ears from the future?
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- Measurements for your $3K DAC (seriously?) that shows that its distortion is less than the one produced by Reclock running at 48 kHz (0.00024% according to your own measurements). Good luck with that. Even professional lab equipment like the Audio Precision SYS-2722 ($25,000) is incapable of generating an analog sine wave with distortion that low. - Documented double-blind studies that show humans are capable of detecting 0.00024% of distortion. Let me spare you the trouble: they don't exist. When I say that it's not audible, I'm not speaking from personal experience, I'm speaking from what is widely recognized in the scientific community regarding the audibility thresholds of the human ear. Last edited by e-t172; 7th March 2013 at 20:41. |
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