[drez]

Quote:

You all need to realize that playback is not the same as recording ..
Obviously, when recording, any 'latency' is unacceptable -
YES, the guitarist,bassist, drummer whatever - will notice a 30ms delay from doing something till hearing it ..
It's enough to make the drummer off-beat !
 
When playing back a digital file from a computer - There is no such thing as 'latency' !
You are NOT playing a LP-record, 'decoded' in 'real-time' .
Ultimately, you will want the entire track read from disk to RAM and played from RAM without further disk-access .
CD's are almost read like LP's - That's why Sony made a load of money on that 'anti-skip' thing on the Discmans .
They did what foobar and all other computer-playback software does : Load it to RAM and play from there .

 
I think you misunderstood me, I said I prefer to use a small buffer for reasons"other than input delay."
 
There most definitely is such a thing as latency and it is there in varying amounts at every stage of the computer setup.  You want to to input a command to a music player, there is latency, to access a hard disk, there is latency, you have different bus speeds, different speeds of data flow, and on top of this data does not usually arrive in a smooth manner so you need buffers at strategic points to ensure there is a constant flow of data and no dropouts.  If you have a problem where the latency is inconsistent (very typical in Windows machines) you need a larger buffer, or on the other hand you can use techniques like ram disk etc to ensure a smoother flow of data and therefore not need such a large buffer.  You can also not use a ram disk at all and just use a larger size buffer, but I find that this does not sound as good to my ears.  Most music software do not load the entire file into ram, they instead load a number of samples ahead of time thus use a buffer.  Audio data coming out of a USB port does not have consistent timing, which is why timing correction mechanisms are needed at the USB receiver, maybe even another buffered control mechanism.  IMO none of these systems are perfect at fixing timing issues, whether this is audible is another matter.
 
The simplistic view is that you can just use a large buffer at any stage and you will get perfect results, but in my opinion this is not the case, as in I believe the buffer size and nature affects the sound quality.  You may not hear any differences from these settings and this is fine.  Even if I could provide evidence for any of my opinions I do not think it would be in anyone's interest.  I could for argument's sake throw a million ferrite beads on a USB cable, prove I can hear a difference with DBT and then say a million ferrite beads sounds better, but still not be contributing anything useful.  What I mean to say is that audio is so inherently subjective that even if a difference can be heard, people will not agree on whether there is an improvement.  IMO If you are really interested in something you need to do your own research and not rely on extrapolation and prediction from other peoples research, and not mistake one (or more) negative result/s for a universal truth in circumstances outside the scope of the test setup.  If you don't worry about subtle differences in sound quality then you could just not do the testing and direct your efforts elsewhere.

 

[bigshot]

I've found that I do a lot better improving the sound of my system by not surrendering to pure subjectivity. I get great results by figuring out how things work and addressing specific problems. Sound isn't magic. It's fairly easy to understand and control.
 
If you're getting skipping, you need a buffer. It's pretty simple.

 

[dleblanc343]

Quote:

 
Ineteresting, can't say I have noticed anything similar on my system, but my setup is different as I am using USB transport with galvanic isolation connected to grounded computer and DAC on the same circuit.  So this is using USB from a laptop into a Bifrost DAC?  Without knowing what is inside the Tellurium Q cable it is hard to figure out what is going on there

  If we could debug the USB receiver then maybe we could say whether or not the cable is affecting the buffer management causing buffer underrun or whether the dropout is from some sort of electrical interference.

I am clueless to what can be causing this; it's a bit way too technical to me. But I can still confirm that what I've heard is still occuring
Quote:

The Bifrost reportedly clicks when the sampling rate changes.

Yes this is true

 

[AKG240mkII]

Quote:

 
I think you misunderstood me, I said I prefer to use a small buffer for reasons"other than input delay."
 
There most definitely is such a thing as latency and it is there in varying amounts at every stage of the computer setup.  You want to to input a command to a music player, there is latency, to access a hard disk, there is latency, you have different bus speeds, different speeds of data flow, and on top of this data does not usually arrive in a smooth manner so you need buffers at strategic points to ensure there is a constant flow of data and no dropouts.  If you have a problem where the latency is inconsistent (very typical in Windows machines) you need a larger buffer, or on the other hand you can use techniques like ram disk etc to ensure a smoother flow of data and therefore not need such a large buffer.  You can also not use a ram disk at all and just use a larger size buffer, but I find that this does not sound as good to my ears.  Most music software do not load the entire file into ram, they instead load a number of samples ahead of time thus use a buffer.  Audio data coming out of a USB port does not have consistent timing, which is why timing correction mechanisms are needed at the USB receiver, maybe even another buffered control mechanism.  IMO none of these systems are perfect at fixing timing issues, whether this is audible is another matter.
 
The simplistic view is that you can just use a large buffer at any stage and you will get perfect results, but in my opinion this is not the case, as in I believe the buffer size and nature affects the sound quality.  You may not hear any differences from these settings and this is fine.  Even if I could provide evidence for any of my opinions I do not think it would be in anyone's interest.  I could for argument's sake throw a million ferrite beads on a USB cable, prove I can hear a difference with DBT and then say a million ferrite beads sounds better, but still not be contributing anything useful.  What I mean to say is that audio is so inherently subjective that even if a difference can be heard, people will not agree on whether there is an improvement.  IMO If you are really interested in something you need to do your own research and not rely on extrapolation and prediction from other peoples research, and not mistake one (or more) negative result/s for a universal truth in circumstances outside the scope of the test setup.  If you don't worry about subtle differences in sound quality then you could just not do the testing and direct your efforts elsewhere.

 
 
Quote:

Even if I could provide evidence for any of my opinions I do not think it would be in anyone's interest.

Please do !
I would REALLY like to hear why there is such a thing as 'latency' during playback of a digital file !
 
Or move this to one of the voodoo-allowed threads
 
Latency - Relative TO WHAT during playback ?
Either your soundcard/DAC - call it what the hell you want - plays music.. Or it doesn't !
 
 
 
Quote:

IMO If you are really interested in something you need to do your own research and not rely on extrapolation and prediction from other peoples research, and not mistake one (or more) negative result/s for a universal truth in circumstances outside the scope of the test setup.  If you don't worry about subtle differences in sound quality then you could just not do the testing and direct your efforts elsewhere.

Yeah, the engineers don't know **** and science even less, right ?
All their megabucks analyzers are wrong, because they don't use Black Gates and you cant roll the op-amps ?
 
I worry about the subtle differences between a 1962 and a 20XX Fender-guitar -
I don't worry about playback-latency because there is no such thing !
 
SERIOUSLY !
This thread is bollocks, please move it to one of the many non-science threads !
USB-cable and sound-quality ?
LATENCY during playback ??
VOODOO-HIFI BOLLOCKS IS WHAT THIS IS ! 

 

[bigshot]

The reason that latency is an issue for recording is that you are laying down tracks. You listen to the playback from the last track you cut as you play the one you're recording. Latency can screw up musicians' performances because if their performance is being filtered in real time, a delay caused by an overloaded processor can totally mess up them monitoring themselves, and it can cause sync issues with the track they're laying down. You can't record with latency.
 
But for playback, you only have one track, and if there's a microsecond delay in playback it doesn't affect anything. It doesn't have to sync to anything else. Latency isn't an issue for playback. Buffer underruns are the problem there. That results in skips and clicks.

 

[Griploc]

I am an audio engineer and music producer and i back this quote by big shot. When dsp is being used from your computer big or small amounts of latency will occur. Audio plug ins are usually the culprit for this, but if you have a really low buffer setting that may fix the problem when recording takes. The larger the buffer the more latency occurs.

 

[joeyjojo]

Quote:

 
How could an increased buffer size possibly hurt sound quality?

 
I'd like to know this as well. Head-fi rumour or truth?

 

[liamstrain]

 
How could an increased buffer size possibly hurt sound quality?

On playback? It cannot. If anything, it helps since it has time to compensate for data transmission errors. 

 

[mikeaj]

Set the buffer to 20 years.
 
When you come back and listen, your hearing might be worse!

 

[Stommager]

What an interesting thread!
 
I find myself amazed by the number of different concepts being presented here.
Personally I agree with the ones that say that USB can't have any influence on sound quality. For those who thing different I have this little experiment:
 
- prepare a wave file of Your favorite song and an pan drive
- take the first cable (the expensive one, for example) and connect Your pen drive via it (it might require additional connector)
- make a copy of Your wave file into the pen drive
- now connect the pen drive via the second, standard USB cable and make a second copy of Your wave file
- now connect Your pen drive directly to Your PC and compare the two files bit-wise
- keep repeating until You find any difference
 
Now someone might ask: what does it have to do with DAC and audio playback?
Well, actually everything, because that's exactly what DAC's USB component does: makes a copy of Your audio data and stores it in a memory.
The only read difference is that it does not copy the entire track at once, but rather a small portion at a time. Once this portion in written into the DAC's memory it is read sequentially and fed into the actual DAC chip. Once we get near the end of the portion being played, another one gets copied over USB. This way we make just another copy of Your audio data, only the destination memory and the copying procedure is slightly different. How can anyone expect that a decent DAC cannot do what any no name pen drive can?
 
For me it is really that simple. If the portions of Your audio data arrives on time (in normal circumstances they always do) there is nothing within USB cable that can affect the sound. I think It is very important to understand that data transfer rate inside USB cable has nothing to do with the DAC's clock. Data is being sent in bursts and then for considerably long time nothing happens. DAC's memory (the buffer) is being read sequentially, sample by sample according to DAC's internal clock. My E-MU gives a great example. When I use USB as an input the diode always indicates that the device runs on it's internal clock. In a properly designed DAC no jitter nor any typical level of electric noise can affect analog audio signal, because they are being separated by the buffer and I'm sure that, as well as any pen drive, the buffer gets perfect copy of Your data. Try copying some documents onto Your pen drive via USB cable and see if You can detect any typing errors being introduced due to the jitter or electric fluctuations.

 

[drez]

Quote:

 
How could an increased buffer size possibly hurt sound quality?

 
I have no idea how it could, from what I understand I would not expect it to make a difference, and in my experience it does not make a significant difference.  As I do not have enough understanding of the inner workings of computer audio software, drivers etc. I can't speculate on why buffer size could make a difference.
 
On the other hand I find I prefer using a smaller buffer size, and with a software like JPlay I find using too large buffer can undo some of the benefit.  The changes I think I am hearing however are very small, even with large changes in software buffer size, for example in the record I am listening to now the backing violins are a little more vague with a buffer size of 1 second compared to 0.1 second.  Put it this way I do not like my chances of telling the difference in a blind test.
 
For all sensible intents and purposes you probably don't have to worry about it.  If you are chasing that last 1-2% of performance then maybe it might be worthwhile comparing.  It is free to test after all.
 
But the only way I can see anything feeding an asynchronous USB DAC making a difference beyond dropouts is by affecting the buffer management.  If it were a case of the stream either working or not working ie the buffer at the DAC either working or not working then I can't see how anything on the computer or USB cables etc could make a difference, but I do not think this is the case.  Plenty of variables from bitstream size (24 or 16 bit) to CPU scheduler settings seem to make a difference in my experience.  If you haven't noticed anything similar it is probably a lot to swallow.
 
Yes this is truly getting into magical audiophool terriroty here so I am probably trying to dig my way out of a hole here...  Probably getting away from discussing USB cables also...

 

[bigshot]

"There's no reason that it would make any difference, but I hear a tiny bit of vagueness in the backing violins... I'm sure it would disappear in blind testing."

You've got a textbook example of expectation bias there. It's a useful thing to think about and learn from when it's that clear cut. What other areas may expectation bias be creeping into your analysis of sound quality? Use what you're experiencing here to test other things you believe you hear.

Good sound isn't about perfecting the last 1% or 2%... That's the realm where expectation bias rules the roost. Good sound is about driving the bugs out of the 98% or 99%. It is a LOT harder than it seems to do that. It's so hard, a lot of people don't even bother to address it. They put their faith in high price tags and fancy sales literature and piddle away their energy worrying about things that just don't matter.

 

[Clarkmc2]

Good sound isn't about perfecting the last 1% or 2%... That's the realm where expectation bias rules the roost. Good sound is about driving the bugs out of the 98% or 99%. It is a LOT harder than it seems to do that. It's so hard, a lot of people don't even bother to address it. They put their faith in high price tags and fancy sales literature and piddle away their energy worrying about things that just don't matter.

Using this as a litmus test, I have been able to ignore nearly all the posts on audio forums for years. It has also directed my personal pursuit of improving my own systems.

 

[ultrabike]

Quote:

What an interesting thread!
 
I find myself amazed by the number of different concepts being presented here.
Personally I agree with the ones that say that USB can't have any influence on sound quality. For those who thing different I have this little experiment:
 
- prepare a wave file of Your favorite song and an pan drive
- take the first cable (the expensive one, for example) and connect Your pen drive via it (it might require additional connector)
- make a copy of Your wave file into the pen drive
- now connect the pen drive via the second, standard USB cable and make a second copy of Your wave file
- now connect Your pen drive directly to Your PC and compare the two files bit-wise
- keep repeating until You find any difference
 
Now someone might ask: what does it have to do with DAC and audio playback?
Well, actually everything, because that's exactly what DAC's USB component does: makes a copy of Your audio data and stores it in a memory.
The only read difference is that it does not copy the entire track at once, but rather a small portion at a time. Once this portion in written into the DAC's memory it is read sequentially and fed into the actual DAC chip. Once we get near the end of the portion being played, another one gets copied over USB. This way we make just another copy of Your audio data, only the destination memory and the copying procedure is slightly different. How can anyone expect that a decent DAC cannot do what any no name pen drive can?
 
For me it is really that simple. If the portions of Your audio data arrives on time (in normal circumstances they always do) there is nothing within USB cable that can affect the sound. I think It is very important to understand that data transfer rate inside USB cable has nothing to do with the DAC's clock. Data is being sent in bursts and then for considerably long time nothing happens. DAC's memory (the buffer) is being read sequentially, sample by sample according to DAC's internal clock. My E-MU gives a great example. When I use USB as an input the diode always indicates that the device runs on it's internal clock. In a properly designed DAC no jitter nor any typical level of electric noise can affect analog audio signal, because they are being separated by the buffer and I'm sure that, as well as any pen drive, the buffer gets perfect copy of Your data. Try copying some documents onto Your pen drive via USB cable and see if You can detect any typing errors being introduced due to the jitter or electric fluctuations.

 
That wouldn't be fair though. A file copy would more than likely be a "Bulk" transfer, while most time sensitive devices (DACs) use the higher priority "Isochronous" trasfer (usually Adaptive mode):
 
http://www.beyondlogic.org/usbnutshell/usb4.shtml
 
My undestanding is that "Bulk" uses ACKs to synch transfers, while Adaptive "Isochronous" requires clock recovery since the slave device (DAC) has no control over the rate at which data arrives to it.
 
All that said, I don't think the jitter coupled through the source clock is an issue as long as the receiver in the slave devices does a good job at mittigating it. I believe Stereophile publishes jitter rejection of the DACs they review. Some manufacturers may publish their specs also...
 
Note that this may only be an issue with poorly implemented external DACs. If using a Sansa Zip/Clip, iPOD/iPAD internal DAC, or such, where data is stored in memory (though USB "Bulk" transfer), then this source coupled jitter deal becomes a non-issue AFAIK.

 

[grokit]

Quote:

Quote:

What an interesting thread!
 
I find myself amazed by the number of different concepts being presented here.
Personally I agree with the ones that say that USB can't have any influence on sound quality. For those who thing different I have this little experiment:
 
- prepare a wave file of Your favorite song and an pan drive
- take the first cable (the expensive one, for example) and connect Your pen drive via it (it might require additional connector)
- make a copy of Your wave file into the pen drive
- now connect the pen drive via the second, standard USB cable and make a second copy of Your wave file
- now connect Your pen drive directly to Your PC and compare the two files bit-wise
- keep repeating until You find any difference
 
Now someone might ask: what does it have to do with DAC and audio playback?
Well, actually everything, because that's exactly what DAC's USB component does: makes a copy of Your audio data and stores it in a memory.
The only read difference is that it does not copy the entire track at once, but rather a small portion at a time. Once this portion in written into the DAC's memory it is read sequentially and fed into the actual DAC chip. Once we get near the end of the portion being played, another one gets copied over USB. This way we make just another copy of Your audio data, only the destination memory and the copying procedure is slightly different. How can anyone expect that a decent DAC cannot do what any no name pen drive can?
 
For me it is really that simple. If the portions of Your audio data arrives on time (in normal circumstances they always do) there is nothing within USB cable that can affect the sound. I think It is very important to understand that data transfer rate inside USB cable has nothing to do with the DAC's clock. Data is being sent in bursts and then for considerably long time nothing happens. DAC's memory (the buffer) is being read sequentially, sample by sample according to DAC's internal clock. My E-MU gives a great example. When I use USB as an input the diode always indicates that the device runs on it's internal clock. In a properly designed DAC no jitter nor any typical level of electric noise can affect analog audio signal, because they are being separated by the buffer and I'm sure that, as well as any pen drive, the buffer gets perfect copy of Your data. Try copying some documents onto Your pen drive via USB cable and see if You can detect any typing errors being introduced due to the jitter or electric fluctuations.

 
That wouldn't be fair though. A file copy would more than likely be a "Bulk" transfer, while most time sensitive devices (DACs) use the higher priority "Isochronous" trasfer (usually Adaptive mode):
 
http://www.beyondlogic.org/usbnutshell/usb4.shtml
 
My undestanding is that "Bulk" uses ACKs to synch transfers, while Adaptive "Isochronous" requires clock recovery since the slave device (DAC) has no control over the rate at which data arrives to it.
 
All that said, I don't think the jitter coupled through the source clock is an issue as long as the receiver in the slave devices does a good job at mittigating it. I believe Stereophile publishes jitter rejection of the DACs they review. Some manufacturers may publish their specs also...
 
Note that this may only be an issue with poorly implemented external DACs. If using a Sansa Zip/Clip, iPOD/iPAD internal DAC, or such, where data is stored in memory (though USB "Bulk" transfer), then this source coupled jitter deal becomes a non-issue AFAIK.

 
 
That's what I was thinking, that Stommager was describing how an asynchronous DAC works. In an adaptive DAC, we are relying on the computer's internal clock for better or worse. The bits may get there on time, but not necessarily in time. IMO there's a difference, and that's where jitter issues can come into play.