0.9y-7

[PeterSt]

Gerard - 0.9y-8 doesn't change a bit or byte.

And, I just don't understand that table you presented.

FYI : One sample is always indicated as an "x channel sample" where x in our case is 2. So, one 16/44100 sample comprises of :
2 bytes (= 16 bits) per channel = 32 bits = 4 bytes for 2 channels = 44100 * 4 = 176400 bytes per second @ 16/44100.

1 sample thus (regardless of bit depth) takes 1/44100 = 0.0002267573 seconds. E.g. 88 samples take 88 times of that = 0.00199546424 seconds. This is 2ms.
At 176400 samples per second this is 1/176400 = 0.00000566893 for 1 sample, and 88 samples take 0.00049886584 seconds, or 499 microseconds (half a millisecond).

Maybe I am wrong somewhere, but if not, I don't understand that table.

Of course this has nothing to do with the (not so good) sound you perceive from it all.

[GerardA]

      3.6 ms  5.6 ms  9.6 ms 13.6 ms 17.6 ms  25.6 ms  37.6 ms
44      88       176     352      529       705       1058      1587
48      96       192     384      576       768       1152      1728
88     176       352     705    1058      1411       2116     3175
96     192       384     768    1152      1536       2304     3456
192    384      768    1536    2304      3072        313      2617

The first column is the samplerate, the other 7 columns are the 7 settings  I can do in the software.
Like at 44 kHz and a setting of 3.6 ms the display says this is 88 samples.
If this is true I dont know.

BTW. Apart from the one complaint it sounds super!

[PeterSt]

Yeah, but the "sibilance" in women voices seems a clear case of emphasizing the wrong. It's one of the most difficult things anyway (ever tried Diana Krall ?).

Let's say we're not done with this one, okay ?

[GerardA]

No not done yet, are you ever done?

But it is not the sibilance, it's more the aaaaa, eeeee, and iiiiiiii!!
Like a nylon string starts to sound like a metal one.
I have no musical education, so difficult to explain!

[xp9433]

GerardA

You said "Like a resonance somewhere between 2 and 8 kHz which especially vibrates in high female voices.
I think now it's not really in the transients but more in the vowels (klinkers) and long notes of violins and brass.
Maybe its a cone breakup of my metal cones..."

As a matter of interest have you tried the same track with reversed absolute phase to see if this makes any difference to the above sound?

Frank

[ivo]

Hello Peter,

Just installed yesterday the latest 0.9y-8, but this is also with 0.9y-7 using engine#4, no up-sampling, just going straight 16/44:

So, it seems that *on some albums (FLAC)* there are little holes during playback - kind of very short drop outs in sound, however there are no buffer errors in log file and there are no CPU spikes as well. Average CPU usage = 5-8 %. nothing else being done on machine. DAC is connected via USB, see my sig.

Setups problem seen with: Adaptive and Normal modes, Q1=1-4, Device buffer 128-256, Schemes 2 and 3. There are albums when I can easily go with Q1=1 and Buffer=128, but there are albums where Q1=4, buffer=256 and still occasionally small dropouts. BTW, tried the same tracks via engine#3 - all goes smooth.

Question1: Does *Adaptive* also means there is something dynamically being adjusted with buffers during playback?

Question2: Should I go and increase Q1 even more up? I guess higher Q1 is not that good for SQ in KS mode?

Question3: Earlier you mentioned that Q1's unit is undefined, but to my mind XX' s code contains either function or object, where Q1 serves as an input value. If that is so, then what is that function's/object's output then? What is the unit? Is it time or sample or offset or coefficient? I am not trying to understand your code, just want to understand what is the slider I move so frequently!

Otherwise when no drop-outs occur, playback is just amazing, faboulus, fantastic and whatever else great!

Thanks,
Ivo

[Marcin_gps]

Gerard - 0.9y-8 doesn't change a bit or byte.

And, I just don't understand that table you presented.

FYI : One sample is always indicated as an "x channel sample" where x in our case is 2. So, one 16/44100 sample comprises of :
2 bytes (= 16 bits) per channel = 32 bits = 4 bytes for 2 channels = 44100 * 4 = 176400 bytes per second @ 16/44100.

1 sample thus (regardless of bit depth) takes 1/44100 = 0.0002267573 seconds. E.g. 88 samples take 88 times of that = 0.00199546424 seconds. This is 2ms.
At 176400 samples per second this is 1/176400 = 0.00000566893 for 1 sample, and 88 samples take 0.00049886584 seconds, or 499 microseconds (half a millisecond).

Maybe I am wrong somewhere, but if not, I don't understand that table.

Of course this has nothing to do with the (not so good) sound you perceive from it all.

yyyy, does it mean that the higher sample rate, the lower latency is possible? or am I missing sth oO
because basically I can get lower latencies with my m-audio at 96kHz than at 44kHz.

[PeterSt]

In theory : yes. But then this counts when you reached 1 sample. About the more normal situation, I really don't know ! To me it seems it doesn't matter (or shouldn't) because it is about the response of the system to the driver needing samples, and how fast they are consumed at that other side (from the driver towards the DAC) is unrelated (sort of, but notice that for that too system resources are needed). My feeling though says that with the higher sample rate a lower latency can be achieved, yes.

[PeterSt]

Hello Peter,

Just installed yesterday the latest 0.9y-8, but this is also with 0.9y-7 using engine#4, no up-sampling, just going straight 16/44:

So, it seems that *on some albums (FLAC)* there are little holes during playback - kind of very short drop outs in sound, however there are no buffer errors in log file and there are no CPU spikes as well. Average CPU usage = 5-8 %. nothing else being done on machine. DAC is connected via USB, see my sig.

Setups problem seen with: Adaptive and Normal modes, Q1=1-4, Device buffer 128-256, Schemes 2 and 3. There are albums when I can easily go with Q1=1 and Buffer=128, but there are albums where Q1=4, buffer=256 and still occasionally small dropouts. BTW, tried the same tracks via engine#3 - all goes smooth.

Question1: Does *Adaptive* also means there is something dynamically being adjusted with buffers during playback?

Question2: Should I go and increase Q1 even more up? I guess higher Q1 is not that good for SQ in KS mode?

Question3: Earlier you mentioned that Q1's unit is undefined, but to my mind XX' s code contains either function or object, where Q1 serves as an input value. If that is so, then what is that function's/object's output then? What is the unit? Is it time or sample or offset or coefficient? I am not trying to understand your code, just want to understand what is the slider I move so frequently!

Otherwise when no drop-outs occur, playback is just amazing, faboulus, fantastic and whatever else great!

Thanks,
Ivo

Pwew, I don't know what to think of dropouts despending on the album, except from "can't be so". But I know, you won't be helped by that.
Are you sure this isn't the case for all albums, but that only a certain type of music shows it to you ?
In any case I would try another PC. I mean, drop outs without a spur of extra CPU (and thus player IO because that always goes along with CPU right after it), while towards the program also nothing is shown from buffer errors, must be a PC thing. Don't forget the driver though.


Q1 : No.

Q2 : No. But this makes me think ... It looks like you may be doing something wrong. I mean, how can it be that you work in different buffer sizes, while your USB device doesn't allow you to change the buffer size. Hmm ... I guess you didn't follow the procedure (for Adaptive Mode) to get to the buffer size and use that. So, that is 128 or 256 or whatever, but not 128 *and* 256, as you said (anyway, that is how I read it). So, when you found your buffer at 128, I think you should stay there, and increase Q1 when needed. You may even accept buffer errors (to test) but avoid the dropouts with it (??).
But also keep in mind what I said earlier in this post.

Q3: No, is not so. Specially for you : Let's say that sliding up Q1 (this is Adaptive Mode only !!) gives more headroom for other tasks to do their work, oriented towards the driver task. In the end sliding up means higher latency, so if it is that you wanted to know, now you do. But to keep in mind (I told it before, somewhere), this is net latency. Thus, this "slider you use so frequently" this time changes net latency (which is what musicians would observe).
So, you get that ?
Haha, right. Normal Mode does that too, but the KS means used is very different, and therefore the latency is way more high to begin with. This is still no normal Kernel Streaming because it allows a latency of minutes for certain devices !! (the first two KS versions used that). Currently this is not supported anymore, because not all devices support it (like USB).

Be warned : the more times you ask about this, the more you will get confused.

Peter

[ivo]

Thanks Peter for explanations.
I am actually waiting on HiFace arrival, let's see how it goes with it. I hope it will perform better as it has custom drivers and it works in async mode ove USB.
At least I know now what Q1 is doing

Ivo

[Telstar]

IMHHHO you have one of the best DACs around ... I probably would have chosen that, weren't it that, well, today they oversample (a little only).

Peter, note that the Gold link is NOT the DAC III Platinum, which I finally heard a few months ago and it sounds REALLY good indeed.