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Old 9th September 2010, 03:55 PM   #771
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Default off topic but it's been quiet...

Quote:
Originally Posted by barrows View Post
My experience with other ASRC solutions (like the TI 4192) has shown that although ASRC does reduce measured jitter, the DAC still sounds better every time one reduces the jitter level at the input to the ASRC. Some digital engineers have tried to explain why this is, saying that the ASRC reduces (actually they say "filters") the incoming jitter, but really just chages the jitter to other artifacts that demodulate to noise in the output of the DAC. I do not fully understand the process, but my listening experience suggests that it is always wise to feed a DAC (reclocking or not) the lowest jitter data stream possible, and that DAC claims like: "immune to jitter" are generally not true.
I think this quote is very wise.

Something happened yesterday that reminded me of this comment. I merely replaced the motherboard, CPU and memory in my PC sound server. Nothing more. The difference in sound was surprising (and better, thankfully)! The old motherboard was more than adequate for the task computationally. The new one [Gigabyte 880 series] has extra copper in the ground and power planes in order to act as a heat sink for passive chip cooling. Whatever the source for the sonic improvement, "jitter" would seem to be the most likely variable. Upsampling using Quicktime (w/ ASIO output to an external sound board from which I2S is hijacked), the improvements are lost when I push frequency up to 192kHz. 96kHz is preferred - slightly better than 88, which is better than 44, which is about equal to 192. Surprising that the computer source could make all of the Twisted Pear gear so much better! Obviously, our individual experiences with jitter will vary, but I don't assume that any component is immune to it's effects. I just wish that finding and eliminating it wasn't such voodoo...
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Old 9th September 2010, 06:05 PM   #772
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What we need is a transport protocol inmune to jitter. You have them all around you in almost any cheap device.

Sorry for being tiring but, I cannot stop wondering why all other areas like telecommunications had surpassed the jitter issue, and in the 21th century, we are still suffering corrupted digital transfers..
Really, I cannot get it inside my head. USB transfer to a pendrive, compact disk storage, ethernet connections, SATA links, even WI-FI! They're all inmune to jitter, have buffers and recovery mechanisms.
Let's say, it would be solved by implementing an easy point-to-point protocol based on packets. CRC error detection and such. Even the good old Ethernet will work, we don't need low latency or high bandwidth (Meaning bytes per second, not audio frecuency response). Even 1.000ms of latency will work, whatever it takes. Even 687.5 Kb/s (44,000 x 16 / 1024) data rate would be enough for CD quality. That's a DSL link of 5.5Mbits, so this theoretical protocol could deliver CD quality jitter-free to any part of the world by just having a "cheap" internet connection.
Just buffering the incoming packets in a memory, and then processing it as it needs the data. This would work prefect for applications not demanding low latency, like listening to music.

We need a whole new philosophical approach. Would it be acceptable for you if some of these characters you are reading now would be randomly changed for another ones? "it's because of jitter, you've to live with it"
NO! NEVER! totally unacceptable in any consumer enviroment, even less in a business one. So why accept and approve that as normal in the Audio enviroment?

Had the world gone mad or is just me thinking too much?

Is there any other tech guy out there to share impressions?
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Old 9th September 2010, 06:53 PM   #773
LeonvB is offline LeonvB  Netherlands
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In essence you're asking for a buffer inside the DAC chips, and you prefer an ethernet interface to the current I2S interface. That makes sense, but the DAC manufacturers aren't there yet. Give them a few years of further integration and we might get there.
And CD Quality is 44000 x 2 / 1024 in Kb/s, as it's 44Khz and 16 bits (i.e. 2 bytes).
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Old 9th September 2010, 10:24 PM   #774
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Quote:
Originally Posted by LeonvB View Post
In essence you're asking for a buffer inside the DAC chips...
Or perhaps a separate chip on the same board. The important initiative would be to introduce some kind of error-checking. One persistent issue relative to the chip market will be acceptable response latency in the face of very diverse applications for which DACs are required. (e.g. video synch, live performance, etc.)
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Old 9th September 2010, 10:59 PM   #775
LeonvB is offline LeonvB  Netherlands
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Error checking is useless without a buffer. If you detect an error but there's no time to perform a re-transmit of the data the error could just as well go on undetected. So you need a buffer to obtain data to decode right now, and that buffer has to be filled fast enough to not run out of data. But if that buffer is outside of your DAC chip, you have to get the data into the DAC eventually, and then you're running into the same problems designers had before.
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Old 10th September 2010, 12:40 PM   #776
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Please please please. There are a thousands threads in the Digital forum about jitter...
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Old 11th September 2010, 02:47 PM   #777
Bunpei is online now Bunpei  Japan
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Default Examples of playing DSD sources on Buffalo II

These days, I find three examples of playing DSD sources using Buffalo II DAC in Japanese blog sites. I'd like to introduce them on this post.

There are two major sources.
1. Tapping DSD signals from commercial SACD players
2. "USB Audio board" kit + "PlayAudio" software by ElectrArt

1. Buffalo II with SACD players
A. "Ken" reported that he connected his SONY SCD-XE6 to Buffalo II in his blog page.
SACD‚ŠO•t‚DAC ƒPƒ“‚ƒI[ƒfƒBƒIƒƒ‚/ƒEƒFƒuƒŠƒuƒƒO (In Japanese)
His impression was "staging is deep and more natural".
B. Sunacchi's case, his player is SONY SCD-X501
SDƒJ[ƒhƒvƒŒƒCƒ„[“†‹@i‚‚‚Rj: Ama Ama Audio Visual(In Japanese)
He got the similar impression with Ken's. "Sound is more stereoscopic and natural."

2. Buffalo II with ElectrArt's USB Audio board
The "USB Audio board" is designed and distributed to audiophiles by ElectrArt.
ElectrArt?Digital Audio??? ???????? USB AUDIO???13(In Japanese)
Using the board with his proprietary software "PlayAudio",
ElectrArt?Digital Audio??? ???????? PlayAudio ?????????2(In Japanese)
we can record and play DSD64 and DSD128 as well as PCM 44.1 - 192 kHz by way of USB.
"Hiyohiyo" reported his successful play of both DSD64 and DSD128 files available on 2L commercial site with the USB Audio board connected to Buffalo II.
?????: 2L?DSD64????DSD128???(In Japanese)

Are there any other DSD sources?

Last edited by Bunpei; 11th September 2010 at 02:54 PM.
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Old 11th September 2010, 03:11 PM   #778
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Quote:
Originally Posted by LeonvB View Post
And CD Quality is 44000 x 2 / 1024 in Kb/s
You forgot to multiply it times 16 bits.


You are right Brian, my fault.
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Old 11th September 2010, 03:46 PM   #779
LeonvB is offline LeonvB  Netherlands
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Quote:
You forgot to multiply it times 16 bits.
No, I didn't. To get a datarate in bytes, one has to divide the bitrate by 8. So a 16 bit wide signal takes 2 bytes. Multiply this by your sample frequency (44000 samples/sec) and you get bytes/s. Divide it by 1024 and you get Kb/s.
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Old 11th September 2010, 04:06 PM   #780
ichiban is offline ichiban  United States
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Kb=Kilobits KB=Kilobytes I think that's the standard nomenclature.
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