USB audio transmission isn’t bit true

[Ralf11]

Just now, kumakuma said:

 

GUTB agrees with you.

 

 

[sandyk]

1 minute ago, Ralf11 said:

 

got data??

 

 All you need to do is tune your AM Clock Radio into a station then move it close to the typical Class D amplifier's power cord and LISTEN with your cakehole closed. 

[sandyk]

11 minutes ago, Ralf11 said:

 

 

 

 How long did it take you to recover from the blast that hit you ?

It would appear that you still haven't  recovered because there is no Z in Phaser

 It also seems that you and Kumakuma are still part of the C.A. Borg collective after being assimilated

[Ralf11]

maybe it's the magic unicorn noise on copies of CDs that makes the Class-D amps sound bad???

[beerandmusic]

I wonder why some designers are moving to bulk-pet over isosynchronous for DSD to shorten the audio latency.

 

https://www.itf.co.jp/prod/audio_solution/bulk-pet/bulk-pet-en

 

It is defined to using Isochronous Transfer to transmit the audio data in USB Audio Class.
Audio data is transmitted in a constant period in Isochronous Transfer.
On the other hand, huge proccessing loading also appears in a constant period on host CPU and device CPU.
This huge processing loading causes the sound quality slight changes.

Bulk Pet is the technology that transmitting the audio data in Bulk Transfer.
In Bulk Transfer, it’s able to control the transmission data volume and transmission frequency.

 

Also...Still no one has injected noise in quad dsd and see if error rate goes up?

[beerandmusic]

What is the theory that 100 different usb cables can give 100 different SQ?

[beerandmusic]

Forget about noise for a moment and pretend all there is, is music being transferred. 

Tone, harmonics, bass, treble, everything that is music is represented by the digital data.

If all of these musical attributes are represented by this digital data, how can losing a bit here or there just cause a drop out and not a difference in tone?  If tone is represented by data, and data is lost how can tone not be incorrect?

Sure, you can say if it recieves too many errors it drops the entire packet, which would result in a dropout....but what is the case in interpolation, where it interpolates rather than drops the entire packet? If the dac never interpolates, why even have the interpolation circuitry.  I mean is there one DAC engineer in the bunch here that can explain this?  Just curious is there a DAC engineer that reads these threads?  Not just a hobbyist, but a professional of a known company?

[diecaster]

17 minutes ago, beerandmusic said:

What is the theory that 100 different usb cables can give 100 different SQ?

 

It’s like using 100 different filters. Each is going to sound a bit different. 

[beerandmusic]

2 minutes ago, diecaster said:

 

It’s like using 100 different filters. Each is going to sound a bit different. 

So how does a filter give more detail, more sound stage?  The music will be the same (bit perfect) regardless of the filter, correct?

[esldude]

10 minutes ago, beerandmusic said:

Forget about noise for a moment and pretend all there is, is music being transferred. 

Tone, harmonics, bass, treble, everything that is music is represented by the digital data.

If all of these musical attributes are represented by this digital data, how can losing a bit here or there just cause a drop out and not a difference in tone?  If tone is represented by data, and data is lost how can tone not be incorrect?

Sure, you can say if it recieves too many errors it drops the entire packet, which would result in a dropout....but what is the case in interpolation, where it interpolates rather than drops the entire packet? If the dac never interpolates, why even have the interpolation circuitry.  I mean is there one DAC engineer in the bunch here that can explain this?  Just curious is there a DAC engineer that reads these threads?  Not just a hobbyist, but a professional of a known company?

Well the real case is what you're talking about doesn't happen.  Nothing gets dropped.

[beerandmusic]

Just now, esldude said:

Well the real car is what you're talking about doesn't happen.  Nothing gets dropped.

Mansr stated when there are errors it is either dropped or interpolated.

[esldude]

1 minute ago, beerandmusic said:

Mansr stated when there are errors it is either dropped or interpolated.

And that it happened about one time every 30 hrs of listening maybe.

[diecaster]

3 minutes ago, beerandmusic said:

So how does a filter give more detail, more sound stage?  The music will be the same (bit perfect) regardless of the filter, correct?

 

Look, this has been covered ad nauseam. 

 

The USB cables change the noise and other crap that come along with the digital data. This means the USB receiver sees different signals with different cables so the sound is changed in different ways depending on the USB cable used. 

[beerandmusic]

12 minutes ago, diecaster said:

 

Look, this has been covered ad nauseam. 

 

The USB cables change the noise and other crap that come along with the digital data. This means the USB receiver sees different signals with different cables so the sound is changed in different ways depending on the USB cable used. 

 

Besides noise, what is "the other crap"?  It is my understanding is all that exists is music and noise (albeit different kinds of noise...so lets just call it noise).

If the music is the same regardless of the noise how do you get more music (e.g. more detail) by using a "noise limiting" cable?  You can't create more music than what exists...and does that mean if you eliminate the noise FIRST, that a noise limiting cable does nothing?  And again, all the more reason to use fiber and forget usb to isolate noise.

[beerandmusic]

16 minutes ago, esldude said:

And that it happened about one time every 30 hrs of listening maybe.

There are lots of different environments, and lots of different hardware, and different dac engineering, etc..etc.......my guess is that if someone is not using a dedicated pc, if it is noisy, it is certainly feasible some environments will have more errors, and interpolation can happen, and give subtle differences, that the listener doesn't even realize there is an issue because of interpolation.

[yamamoto2002]

Musiland USB devices also uses bulk transfer.

 

USB shares one bus to all USB devices connected to one host controller. In order to transfer 96kHz 24bit 2ch PCM (4.6Mbps) on USB 1.x FullSpeed bus (12Mbps) (in this case the PCM stream size is about 40% of USB bus bandwidth capacity), It is necessary to use a mechanism to allocate some length of time slot sorely for the audio data transfer before audio transfer starts and return it after use, isochronous transfer mechanism do the job.

 

If bulk transfer is used, the amount of the data transferred varies on the bus traffic, it depends on other devices' transfer activity, maybe USB memory data transfer occupy 80% of the USB bus capacity. in this case, your bulk transfer USB Audio DAC cannot send 40% of the capacity audio data properly and playback stops or other glitch happens.

 

But now host controller becomes USB 3.0 (5,000 Mbps), it has plenty room for spare, so bulk transfer can be used for USB Audio data transfer scenario unless USB bus is intentionally saturated

 

 

 

 

[beerandmusic]

5 minutes ago, yamamoto2002 said:

Musiland USB devices also uses bulk transfer.

 

USB shares one bus to all USB devices connected to one host controller. In order to transfer 96kHz 24bit 2ch PCM (4.6Mbps) on USB 1.x FullSpeed bus (12Mbps) (it is about 40% of USB bus bandwidth capacity), It is necessary to use a mechanism to allocate some length of time slot sorely for the audio data transfer before audio transfer starts and return it after use, isochronous transfer mechanism do the job.

 

If bulk transfer is used, the amount of the data transferred varies on the bus traffic, it depends on other devices' transfer activity, maybe USB memory data transfer occupy 80% of the USB bus capacity. in this case, your bulk transfer USB Audio DAC cannot send 40% of the capacity audio data properly and playback stops or other glitch happens.

 

But now host controller becomes USB 3.0 (5,000 Mbps), it has plenty room for spare, so bulk transfer can be used for USB Audio data transfer scenario unless USB bus is intentionally saturated

+1

Thanks for sharing...you clearly know a lot more than most here. 

To clarify in layman's terms, USB 3.0 because of higher speed, along with bulk transfer, is ideal?

Can processing quad DSD on USB 2.0 be an issue if not properly managed?...e.g. is there ever a bigger concern with processing quad DSD as opposed to 44.1 PCM?

 

PS are you a dac engineer?  Do you see any advantage in doing fiber (e.g. the new LUMIN X1 with fiber input)?

[yamamoto2002]

53 minutes ago, beerandmusic said:

+1

Thanks for sharing...you clearly know a lot more than most here. 

To clarify in layman's terms, USB 3.0 because of higher speed, along with bulk transfer, is ideal?

Can processing quad DSD on USB 2.0 be an issue if not properly managed?...e.g. is there ever a bigger concern with processing quad DSD as opposed to 44.1 PCM?

 

PS are you a dac engineer?  Do you see any advantage in doing fiber (e.g. the new LUMIN X1 with fiber input)?

 

USB speed increased and bulk transfer becomes a new option for audio transfer. I'm not sure if it is ideal or not but there is a drawback, bulk transfer audio dac needs its proprietary device driver to be installed while operating system supplies stock device driver for isochronous USB audio devices.

 

Quad DSD(23Mbps to 45Mbps) on USB 2.0(480Mbps) should not be an issue. I've tested it and it runs fine.

 

I'm just another layman and a fan of optical link from 1990s S/PDIF because optical gear looks cool and futuristic, and long waiting the fibre version of thunderbolt. Sound quality may not changed very much but it has galvanic isolation as its advantage.

 

[beerandmusic]

18 minutes ago, yamamoto2002 said:

 

 

Quad DSD(23Mbps to 45Mbps) on USB 2.0(480Mbps) should not be an issue. I've tested it and it runs fine.

 

 

 

Ok, thanks...I am hoping to find opinion from DAC engineer perspective regarding quad dsd, noise, error rate, and interpolation.  Bulk seems like it would make sense where real time isn't concern...but still think fiber and enet ideal....e.g. get it right first and don't worry about fixing or simplify dac receiving engineering.

[beerandmusic]

https://www.audiocircle.com/index.php?topic=148763.0

[esldude]

2 hours ago, beerandmusic said:

What is the theory that 100 different usb cables can give 100 different SQ?

There are decades of solid science behind that issue.  But it isn't very popular.  

 

There are other theories that don't really hold up.  But they are very, very popular among a certain subset of people.  

[esldude]

1 hour ago, yamamoto2002 said:

 

USB speed increased and bulk transfer becomes a new option for audio transfer. I'm not sure if it is ideal or not but there is a drawback, bulk transfer audio dac needs its proprietary device driver to be installed while operating system supplies stock device driver for isochronous USB audio devices.

 

Quad DSD(23Mbps to 45Mbps) on USB 2.0(480Mbps) should not be an issue. I've tested it and it runs fine.

 

I'm just another layman and a fan of optical link from 1990s S/PDIF because optical gear looks cool and futuristic, and long waiting the fibre version of thunderbolt. Sound quality may not changed very much but it has galvanic isolation as its advantage.

 

https://www.corning.com/optical-cables-by-corning/worldwide/en/products/thunderbolt-optical-cables.html

 

https://www.amazon.com/Thunderbolt-Optical-Self-Powered-Peripherals-AOC-MMS4CTP060M20/dp/B00JMKG7TM

 

A bit pricy, but there it is. 

 

These are a bit cheaper for 10 meters, but reviews indicate limited reliability.

https://www.amazon.com/Optical-Cables-CorningTM-AOC-MMS4CVP010M20-ThunderboltTM/dp/B00HSTC496

 

[sandyk]

40 minutes ago, esldude said:

There are other theories that don't really hold up.  But they are very, very popular among a certain subset of people.  

 

They aren't just theories. Audible differences between selected USB cables are readily demonstrated under non sighted listening conditions by many members using half decent Audio equipment, even though a certain subset of members are unlikely to hear ANY differences if also present. Their Expectation Bias (Selective Deafness ?) will prevent them from hearing the differences that others present have heard and were statistically verified.

 I also base this around personal experience at several listening sessions where only one member of the around 6 present was unable to hear any differences, despite him realising that all of the others present were able to.  (All were DIY Audio members)

 The same happened again at a subsequent listening session at his own house using his own VAF Dc-x speakers.

 

[beerandmusic]

Here also suggests that noise can corrupt the data, not just dac circuitry processing...

From mojo audio:

 

=====----

 

All computer communication works on a system of checks and error correction (check sum). If a packet of data doesn’t pass the check, a new packet of data is sent to replace the original. The lower the power supply noise, the lower the amount of data corruption, the lower the amount of corrupted data to correct, and the greater the system resources.

 

... When a low-noise power supply is used with a computer-based music server or streamer the result is more liquid and articulate sound, combined with greater depth, detail, and dynamics.

 

====----

I still would like test run on quad dsd signal and purposely inject noise and see if error rate is increased....

 

 

^^^^disregard...i am not sure i buy some of the stuff mojo audio states in that article (lol).

 

 

 

[beerandmusic]

I made a comment earlier...not sure if in this thread or another, but that since we know we can never achieve 100% accuracy, why even worry.  Even if we had 100% accuracy (in regards to flat file on disk), that we wouldn't even know it, like OMG that sounds perfect, since we are very close to a plateau of sort already...

It is very possible, even if we had 100% accuracy, we may find that we like an algorithm provided by one dac over another as more "enjoyable", and you can have the most expensive components to try to achieve accuracy, yet generating internal noise...or you can have a simpler circuit with a good algorithm that makes the music pleasing...again, not even knowing what noise or error rate exists before the DAC did it's magic.....

 

Here, Barrows says something very similar here:

==========

 I find with DACs, the trick is getting all the details, while at the same time having a natural sound.  It does appear,so far, that this approach of reducing the processing inside the DAC itself, is promising in this regard.  Apparently the forthcoming model from PS Audio is addressing the same concerns....

====