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#121 | |
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Also what you say about inaudiable stuff is very important to consider. It is always funny how companies use data about things that we can not see or hear to try to prove something is better. Just have a look at all the bullshit high-end cable claims. There is not a single blind test that prove people can hear a difference in a cable. The cables can still measure different. That said, it is easy to tell from a good hi-fi setup that it can be cleaner than Paula. That does not equal better SQ for most. Last edited by nikosidis; 17 January 2021 at 15:37. |
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#122 |
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Ok, this begs the same question I asked before (but in reverse): how come Bruce's measurement results are so much better than the other ones?
There's a huge difference between 60dB and 75dB. Reading all these different numbers is kind of confusing, I'd expect everyone who starts measuring to get roughly the same results (at least say within 3-6 dB, which is already quite a large difference) and that doesn't appear to be the case. |
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#123 | ||
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SNR is (as the name says) the ratio between the signal, and all deviations (distortions) of the signal compared to the original. Quote:
I measured at maximal signal amplitude. |
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#124 |
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Because they measure something different. See my answer below. If you just turn down the volume until the signal fades away completely, you ignore all other noise that may arise on the channel, such as noise due to imperfections in the quantizer linearity, but also signals from other sources such as the video signal noise at 56Hz I found in the spectrum.
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#125 | |
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Fine, but how do you feed that into your speakers from a computer? The PC soundcard cannot provide a PWM modulated signal with pulse modulation frequency at Mhz rate. They typically play back at 48Khz or 96kHz, or some adjustable frequency below, depending on the technology. |
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#126 |
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That's called "subjective quality", which is certainly the gold standard. However, that wasn't quite the starting point. Maybe the Paula "14 bit audio" is satisfactory for some listeners - and I do not object to that - but the question was whether it would be equivalent to the output of a 14-bit DAC. This doesn't ask for subjective quality. It asks for an analysis of the quantizer noise and other noise sources in the loop. This is a question which can be perfectly answered by "objective quality", with all of its deficiencies and its probably poor correlation to subjective quality understood. |
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#127 | |
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That is why this format has to be transferred into something the PC or the Amiga can handle - on each machine a different approach would lead to the best outcome. After this the results can be compared to each other and to the "original". (DSD can eg be played back using a RasPi and a Teac 301 ...) Last edited by Gorf; 17 January 2021 at 17:24. |
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#128 | |
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#129 |
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I would not underestimate the report by Bruce and also confirmed by others in the forum (that A1200 dynamic range is close or better than 75 dB).
If we find a way to make quantization uniform, then 12 bits are usable/reachable. The problem is figuring out if and how to do it (14 bit calibrated mode is good, but could something better be done?).. EDIT: In any case, I don't know how much this 'quest for uniform quantization' makes sense. The today digital signals are processed and compressed in so many ways and the analog outputs are so artifact that having a system that sounds like a 12 bits dynamic range one, even if somehow distorted and colored, is fine by me. Last edited by ross; 17 January 2021 at 18:22. |
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#130 | |
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#131 |
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I used the small condensator mics that are build into Zoom R16 and recorded A1200, EaglePlayer 14-bit amp, through the speakers playing.
I do not know if it is possible to tell the dynamic range of that sound wave but if someone is interested here it is: https://easyupload.io/cbznv0 Last edited by nikosidis; 17 January 2021 at 21:23. |
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#132 | |
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I did all my tests (further up in the thread, see the linked PDF there for details on materials and methods) with a 24 Bit ADC @ 48 kHz and saved the files as 32 Bit Floating Point AIFF prior to the analysis. The algorithm I've applied works as follows: - trim away silence at start/end of the file - avoidance of leakage by zero cross detection and matching trim on both ends - PSD calculation plus maximum detection of fundamental frequency - SNR calculation from PSD after removal of the fundamental and it's harmonics (for reference: Matlab's snr() removes the first 6 harmonics) |
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#133 | |||
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I also encoded the same song without dither in 16 bit stereo at 44kHz, and played it back with HippoPlayer using the AHI 14 bit engine. It sounded vastly better than either of the 8 bit samples, and the noise floor was inaudible at normal listening level. I know that in '14 bit' mode there is additional distortion and noise compared to a perfect DAC, but the difference is too small to worry about. What is important is that I was able to play a 44kHz 16 bit stereo file on my A1200 with a quality that is good enough for my ears, while 8 bits is not. |
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#134 |
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Attempt for a new approach, it might even be completely useless or bad
![]() All Amiga 14-bit players, calibrated or not, try to make the most by spreading 6-bit (the least significants) from the discrete original PCM signal to the 'PWM' stage (this give the fine steps to the output signal). Unfortunately, as we have seen, making it uniform is complex and imprecise (the full volume PCM channel is usually instead quite linear). Furthermore, seems that the dynamic range output from the [A1200] Amiga does not allow in any case more that ~76db. So why not deliberately decrease the resolution but generate a less distorted signal? I'll try to explain it. What is now being done is this: Vbit = lg2(PCM(h8)) + lg2(PWM(PCM(l6))/x), which to fully exploit the 14 bits uses: x = 1 -> Vbit = lg2 (256) + lg2 (64/1) = 8+6 =14.0 bit Suppose we change instead the x value to: x = 2 -> Vbit = lg2 (256) + lg2 (64/2) = 8 + 5 = 13.0 bit x = 3 -> Vbit = lg2 (256) + lg2 (64/3) = 8 + 4.415 = 12.415 bit x = 4 -> Vbit = lg2 (256) + lg2 (64/4) = 8 + 4 = 12.0 bit x = 5 -> Vbit = lg2 (256) + lg2 (64/5) = 8 + 3.678 = 11.678 bit x is simply the Volume for the 'precision' channel (the one using the PWM property), so we would no longer use volume 1 but 2, 3, 4, 5 ... This would mean changing the calibration and managing the distribution of values completely differently to make it as uniform as possible (how values overlap? maybe a table approach is needed for high and low bits in both channels...). Last edited by ross; 18 January 2021 at 19:52. |
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#135 | |
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Your proposal boosts the LO channel by a given factor. While that won't help with the fundamental problem of HI-channel quantization noise we'd like to suppress as far as possible, there is a potential gain around the zero level (in the HI channel). The biggest problem I see with the former is (at the moment) Paula's comb filter that emits 1/3.5 MHz pulses in bypass mode, which are not sufficiently suppressed by the simple RC-lowpass used in the integrator OpAmp such that they produce harmonics levels we would normally not expect from an 8 bit DAC. But I tend to think you are onto something with the latter. There are currently nasty distortions in the low amplitude area, typically somewhere at and below -30 dBFS signals, resulting in +1/0/-1 on the HI channel. I fixed most of these when I've proposed the corrected quantizer (for EP and AHI-Paula) but some may still be audible. With a higher amplitude on the LO channel, that channel can be used without resorting to changes in the HI channel amplitudes at a wider range around 0. Also, another look at the VOL63+VOL1 approach to 14 Bit approximation may be worthwhile. I've played a little around with 8 Bit / 44.3 kHz at volume 63 and found the intermodulation effects way less pronounced than I remembered them. This way, HI and LO of channel stacking would be synchronous in time, though still not to the same temporal location (but IIRC closer than the current 55.4kHz cycle average). There's a chance that some more of the HI channel quantization noise could be cancelled out by the LO channel in that setup (speculation, for now). Im pursuing a further approach for a while now, by not looking "down" into the dynamic range but rather "upwards" this time. I do get about 50 dB SNR in 9 Bit mode out of my A1200 (close to uncalibrated 14 Bit btw.) Moreover, the 9 Bit approach provides twice the amplitude (+6dB) compared to 14 Bit. In the next step I've combined the 9 Bit mode with a classic 14 Bit mode, adaptively switching between the former and the latter based on the current amplitudes of the signal to be played. This happens in my prototype code at a granularity of about 11 ms. This still won't rival a true uniform 15 Bit quantizer. On the other hand, adaptive quantization relying on masking effects has been successfully used for ages: in G.711 PCM, ADPCM, MPEGA audio etc. |
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#136 | |
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![]() --- Just out of curiosity, how did you solve the problem of the different phase between the left and right channel? From the DMA allocator it seems that for the right channels (HI+LO) there are 45°, for the left(s) 135°... Or did you use the same calibration for both and it works fine? Last edited by ross; Yesterday at 01:37. Reason: opss, inverted the stereo.. |
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#137 | |
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#138 | |
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