You can. Use software to increase the dynamic range. But it's time-consuming, difficult, and lazy. It's easier to just not listen.
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Let's develop an ASR inter-sample test procedure for DACs!
- Thread starter cyril_meysson
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We are talking about intersample overs in this thread, not loudness compression...
Intersample overs are part of the problem with loudness compression, and eliminating them alone won't help.
No, loudness compression is not related to loudness normalization. Two completely separate topics.
You cannot remove a loudness compression during playback. No way. Unfortunately.
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Be careful, some of them have a digitally controlled volume after the DAC, so that would make no difference. And some use an ASRC as volume control, and these are generally very sensitive to inter-sample overs.
Blew
Active Member
Yikes! That's interesting to know, thanks. Do you know of any examples?
totti1965
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I am a huge fan of @NTTY's extremely in-depth CD player/Blu-ray player reviews.
Since many CDs/Files nowadays are recorded far too loudly, a table like this one (taken from a test @NTTY did) makes a lot of sense:
Wouldn't it be a good idea to also test devices such as Streamer-DACs with the same CD uploaded to a hard drive in the LAN network?
It would be extremely exciting, for example, as a WiiM Pro Plus user, to receive a specific recommendation as to which setting would be ideal for avoiding overloads in order to gain an additional 2 dB or even 3 dB of headroom.
My gut feeling tells me that it would be beneficial to the sound quality not to set the maximum output of my Wiim Pro Plus to 100% in the settings, but rather to values between 90% and 96%, for example, so that the resulting sound quality, even with loudly recorded files, corresponds to that of the very best CD-players such as the SMSL PL200, which are least susceptible to intersample-overs.
A specific measurement-based suggestion for older Linn streamers in the German forum “aktives-hoeren.de” was once 82%, for example.
Since it's likely to be different for each device, it would be great if experts like @NTTY or @amirm could do some basic research here.
Or am I completely wrong in assuming that such an analysis would be useful for Streamer-DACs?
Since many CDs/Files nowadays are recorded far too loudly, a table like this one (taken from a test @NTTY did) makes a lot of sense:
| Intersample-overs tests Bandwidth of the THD+N measurements is 20Hz - 96kHz | 5512.5 Hz sine, Peak = +0.69dBFS | 7350 Hz sine, Peak = +1.25dBFS | 11025 Hz sine, Peak = +3.0dBFS |
| Teac VRDS-20 | -30.7dB | -26.6dB | -17.6dB |
| Yamaha CD-1 | -84.6dB | -84.9dB | -78.1dB |
| Denon DCD-900NE | -34.2dB | -27.1dB | -19.1dB |
| Denon DCD-SA1 | -33.6dB | -27.6dB | -18.3dB |
| Onkyo C-733 | -88.3dB | -40.4dB | -21.2dB |
| Denon DCD-3560 | -30.2dB | -24.7dB | -17.4dB |
| Myryad Z210 | -70.6dB (noise dominated) | -71.1dB (noise dominated) | -29.4dB (H3 dominated) |
| Sony CDP-X333ES | -30.5dB | -24.8dB | -16.3dB |
| BARCO-EMT 982 | -32.7dB | -24.5dB | -16.3dB |
| TASCAM CD-200 | -73.5dB | -36.3dB | -19.7dB |
| Sony CDP-597 | -30.4dB | -24.7dB | -16.5dB |
| SMSL PL100 | -53.1dB | -31dB | -19.1dB |
| OPPO BDP-95 | -39dB | -28.8dB | -19.2dB |
| OPPO BDP-95 (vol -2dB) | -95dB | -97.5dB | -32.7dB |
| SMSL PL200 | -94.8dB | -97dB | -39.5dB |
| SMSL PL200 (vol -1dB) | -94.8dB | -97dB | -58.7dB |
Wouldn't it be a good idea to also test devices such as Streamer-DACs with the same CD uploaded to a hard drive in the LAN network?
It would be extremely exciting, for example, as a WiiM Pro Plus user, to receive a specific recommendation as to which setting would be ideal for avoiding overloads in order to gain an additional 2 dB or even 3 dB of headroom.
My gut feeling tells me that it would be beneficial to the sound quality not to set the maximum output of my Wiim Pro Plus to 100% in the settings, but rather to values between 90% and 96%, for example, so that the resulting sound quality, even with loudly recorded files, corresponds to that of the very best CD-players such as the SMSL PL200, which are least susceptible to intersample-overs.
A specific measurement-based suggestion for older Linn streamers in the German forum “aktives-hoeren.de” was once 82%, for example.
Since it's likely to be different for each device, it would be great if experts like @NTTY or @amirm could do some basic research here.
Or am I completely wrong in assuming that such an analysis would be useful for Streamer-DACs?
staticV3
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As long as ISOs aren't terribly broken with integer overflows and the like, I still struggle to understand the fixation with how different DACs handle edge cases that may happen for a few milliseconds per track at most.
There are so much more important things in audio and I genuinely doubt that one can reliably tell the best and worst performers on NTTY's list apart in a blind test, even with tracks that have lots of ISOs.
There are so much more important things in audio and I genuinely doubt that one can reliably tell the best and worst performers on NTTY's list apart in a blind test, even with tracks that have lots of ISOs.
Geert
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45 pages and counting: Let's develop an ASR inter-sample test procedure for DACs!
You're not wrong. I agree with you.
Reviewers like Rja4000 and NTTY (and even myself...) have been testing for ISOs for a while now.
Many people on the forum (founder among them) don't agree that ISOs are a significant enough issue, and others believe that mastering engineers are the cause of the problem, not DACs. These are the main reasons why you don't see headroom tests in most DAC reviews here.
Audibility is likely not a pressing issue, but still... I think digital clipping should not be allowed (I've made a lot of arguments about it, I'm tired lol).
Luckily it's not too difficult to test for ISOs for yourself. All you need is an ADC (not even a particularly great one, just as long as it's not clipping).
Why Fixate on SINAD if an DAC with SIAND over 96dB can be bought for less money then a CD.
This "edge cases" happen in the real world with Real CDs and in real Popular music That people actually listening to... (Unlike 1kHz sine waves from the SINAD test)
So i would not call this a edge cases.
staticV3
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If you record typical pop radio music for 24 hours straight, for how many minutes do you think the digital waveform is above 0dBFS in that time frame?
I'd wager <1 minute, which would equal 0.04s or 10-50% of the time it takes to blink, per minute of playback .
In contrast, a noisy Amp is noisy 100% of the time, an overcooked bass response is boomy 100% of the time etc.
If you record Audiophile classic music for 24 hours straight, for how many minutes do you think the digital waveform is pure 1khz seine wave in that time frame?
Why Would you divert from DAC to AMP? And there are plenty Cheap DACs that don't have a Noises problem.
So it can be considered a solved problem for all realistic purposes. even in some of the cheap models.
staticV3
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Who in this thread said anything about 1kHz SINAD? I didn't that's for sure. Strawman much?
I'm not trying to divert, I'm trying to refocus on things that have actual, tangible, consistent effect on listening experience.
So how would you measure and rank DACs? The standard ranking here seams to be 1kHz SINAD
And if you would stay on the topic of DACs and not divert to to AMPs this argument is invalid since,
There are panty DACs that don't add precivabel amounts of noise (at least as long they are not driven to over 0dBFS) and therefore its not an "actual, tangible, consistent effect on listening experience."
staticV3
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I wouldn't.
Distilling a complex product down into a single number is inherently flawed. See e.g. Olive/Welti Preference score.
The rest of your post seems in bad faith at this point so I'll stop here.
I would, unless you can demonstrate otherwise. And to do that -- for predicting audibility -- you must take the density of ISOs into account.
It is in no way as simple as generating a table like in the OP.
If I recorded an acoustic guitar, and I set my gain so that the captured signal is very close to 0dBFS (very high SNR), to the point that it often just barely clips the ADC for fractions of a second (almost inaudible), would you say that's a good recording? Would it be acceptable if almost every recording were like that?
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