Fidelizer 2.0 - Introducing new UI with multi-core optimizations
Dec 17, 2012 at 8:38 PM Post #166 of 188
Apparently my laptop is a piece of ****. I downloaded the latency checker and it says my computer has problems with realtime audio. It listed some potential problems, I checked into these problems and no change. Fidelizer does help to almost eliminate any audio stuttering though, with both iTunes and the MOG app. I sure wish I knew how to solve these issues though. 
 
Dec 17, 2012 at 8:58 PM Post #167 of 188
The biggest culprit for latency is wifi, after that trying to reduce the amount of background tasks can help. I also set windows processor scheduling for background tasks, set HPET for the OS (32 or 64 bit) you are using and that's about all you can do with software. Newer OS like Windows 8 seem to be a little better at processor scheduling as well.

In my experience doing things that otherwise improve the performance of a computer will help audio playback and allow you to use a smaller buffer.

On that, if you don't really want to go to all the effort of the steps I mention above, you could just use a player with an adjustable output buffer like Foobar etc. and increase the buffer size until you don't get dropouts. Not sure if you can do this in iTunes.
 
Dec 17, 2012 at 9:34 PM Post #168 of 188
I have it set to prioritize background tasks, what is HPET?
 
I have done virus checks and have no virus's on my computer, I have run a scan to check and fix registry errors, I have deleted every single program on my computer that does not get used, including all the HP bloatware that came installed. I have checked for driver and BIOS updates. I run a small SD card for readyboost thinking that may also help. I have tried disabling drivers one by one, I have turned off all animations and all the visual crap that isn't needed. I have done everything I can think of. The biggest help so far to reduce audio stutters has been to run Fidelizer. 
 
My laptop is an hp G60-630US. It's a dual core pentium 2.2 GHZ with 3gig of ram, Win 7 64 bit. Not a great machine, but if my iPhone can play audio smoothly, why cant this laptop? 
 
EDIT: google searched HPET, and did the tweak. Tested and confirmed that I did switch it to HPET. WinTimerTester now shows 14.3MHZ.
Quote:
The biggest culprit for latency is wifi, after that trying to reduce the amount of background tasks can help. I also set windows processor scheduling for background tasks, set HPET for the OS (32 or 64 bit) you are using and that's about all you can do with software. Newer OS like Windows 8 seem to be a little better at processor scheduling as well.
In my experience doing things that otherwise improve the performance of a computer will help audio playback and allow you to use a smaller buffer.
On that, if you don't really want to go to all the effort of the steps I mention above, you could just use a player with an adjustable output buffer like Foobar etc. and increase the buffer size until you don't get dropouts. Not sure if you can do this in iTunes.

 
Dec 17, 2012 at 11:04 PM Post #169 of 188
HPET is one thing, but you'll need a third party app if you wanna force its tightest 0.5ms granularity(it's 10ms stock on W7).....and Fidelizer does exactly that 
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Dec 18, 2012 at 12:52 AM Post #170 of 188
what exactly do you do to HPET?
i have disabled it in BIOS actually as i was told by CMP guide
 
Dec 18, 2012 at 12:58 AM Post #171 of 188
HPET is by far the best timer available: http://msdn.microsoft.com/en-us/library/windows/hardware/gg463347.aspx
 
on XP you can always use the rock solid ACPI clock, but on Vista/W7/W8 if you disable HPET you'll end up using some utterly sloppy and obsolete timer.
 
Dec 18, 2012 at 6:51 AM Post #172 of 188
Any multimedia application that requires a high precision timer will call the appropriate function(s). You don't need to do anything. If you have to, there are problems elsewhere.
 
The biggest culprit is high DPC latency that can cause audio glitches, especially with external audio interfaces.
My PC has an average DPC latency of 60 us. Anything below 250 us is excellent, below 500 is OK. This is with no extra "tweaking" or "tuning" tools, no services disabled and lots of software installed. No glitches, no dropouts, bit-perfect audio.
With a RME interface I can go down to 64 samples buffer size with high CPU load and there are still no glitches.
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Again, fix the actual problems or switch to a better OS.
 
Dec 18, 2012 at 7:26 AM Post #173 of 188
Hmm.. I don't hear any glitches, or problems with audio playback. Was just hoping for some more control over the audio. But I guess that would take away from the original music...
 
Dec 18, 2012 at 8:12 AM Post #174 of 188
Quote:
Hmm.. I don't hear any glitches, or problems with audio playback.

 
The point of a tighter timer is not to "fix audio glitches", it has never been and Fidelizer never advertized itself as a magic fix for this either
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The effects can be measured via Reclock's jitter measurement of the audio clock reference for instance and this page also raises some interesting points: http://www.cicsmemoryplayer.com/index.php?n=CMP.03Jitter
 
But it's all explained in the link of my previous message really = Windows is not a realtime OS:
Quote:
HPET should have a programmable granularity [..] so that interrupts are not generated or delivered late, and minimal drift so that it is consistently reliable.

 
Dec 18, 2012 at 8:59 AM Post #175 of 188
From the tool's website:
Quote:
I use DPC Latency Checker but Fidelizer doesn’t improve system latency at all. Isn’t it scam?
-Graph in system latency shows latency of hardware I/O meaning you’re checking hardware latency. Fidelizer will decrease software latency in operating system so DPC Latency Checker isn’t capable of checking software latency. You need to test it with other methods. If you want to decrease hardware latency in DPC Latency Checker, try disabling drivers like networking for example.

 
That's wrong. The graph shows the DPC latency, not hardware I/O latency. DPCs, like the name suggests, are deferred procedure calls = software. DPC latency is software latency.
This tool can't fix it, but it's the most important aspect of real-time audio.
 
 
Quote:
 
The point of a tighter timer is not to "fix audio glitches", it has never been and Fidelizer never advertized itself as a magic fix for this either
wink_face.gif

Yeah all it advertises is "better sound quality". How this is supposed to work nobody knows. An audio player doesn't even need a timer. It just fills a buffer, the OS will fill a hardware buffer (if it can, DPC latency needs to be low enough to do this in time) and then it's in the hands of the hardware. No amount of software tweaking can improve the jitter generated by the hardware, not that jitter components below -120 dBFS would matter.
 
The cics page you linked to is funny. It doesn't mention timers or even software-induced jitter. It doesn't show comparative measurements, but concludes it is better. The listening test protocol is ridiculous. In other words: it sucks, like the player. (as I've shown a couple of years ago comparing CPU cycles with foobar2000, fb2k won
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)
 
Dec 18, 2012 at 10:14 AM Post #176 of 188
wolf...
 
the idea here is to setup your pc to play songs in a way that doesnt interfere with the original recording in any way....to allow what the actual recording was to come thru and allow you to hear what the artist and recording engineer was trying to provide you the listener.
 
Many of us, me included strive to setup our systems in a manner that will allow the source bits to flow to electronics to be reproduced in an analog format so we can indeed listen to them via our cans or speakers...
 
Adding a PC in the mix vs usng a cd player connected to a preamp and amp then speakers etc adds the additional complexity that comes along with a PC.
 
IE...ripping, noise, dacs, software, conversions etc...its mind boggling at times.
 
But we have come a long way and its possible to have a pc in the mix and set it up along with a application like a 'music player' in a manner that will allow the source to come thru unaltered.
 
Then its down to how the source really sounds with your cans or speakers or transducers.
 
So when you say you want to have control or more control...this can be coming from several areas that you may or may not understand based on your listening experience etc...
 
The first thing I would try to do is to get your pc and software setup to run in a "bit-perfect" manner...this setup will allow the bits from the source to flow through the pc to the DAC and to your AMP etc..and hear what the orignal cd or source intended.....your source files should be in a lossless format like FLAC etc...
 
Once this bitperfect mode is setup you can then decide if you really need to manipulate the audio in anyway....I tend not to, and i look for recordings that are world class and mastered very well.
 
But I agree that there are many recordings that are compressed and sound like crap and you may want to tinker with them via an equilizer etc....but i dont like to waste the time with crappy recordings....there is too much good stuff our there to listen to.....which is another whole topic.
 
I find things like JPLAY are absolutely snake oil....and your wasting your money here on this code....but hey its your money....
 
Hope some of this makes sense..
 
have a great day
 
Alex
 
Dec 18, 2012 at 11:41 AM Post #177 of 188
It should be noted that on supporting Windows OSes as well as supporting hardware (not all hardware is capable of), HPET is disabled by default and has to be manually enabled. It should also be noted that while HPET can help with latency, it introduces issues on video related tasks, so that might be something to consider. Audio wise, HPET should be (in most cases) a superior option to RTC. And as mentioned previously, the Windows 8 scheduler is indeed improved, as well as the core USB drivers being much more stable to the point where the USB 3.0 DAC issue was fixed entirely.
 
Also, Windows 7's default timer isn't 10ms, it's 15.6ms.
 
Dec 18, 2012 at 11:56 AM Post #178 of 188
it's 15.6ms when idle and 10ms when there's a multimedia app running:
 
Quote:
 
GetModuleHandle("ntdll.dll" ) returns 0x77760000
GetProcAddress(ntdll.dll,"NtSetTimerResolution" ) returns 0x77781C44
GetProcAddress(ntdll.dll,"NtQueryTimerResolution" ) returns 0x77781598
--- Get Performence Counter Frequency ---
QueryPerformanceFrequency returns 14318180 /s (14318 /ms)

Possible Timer: HPET (14.32 MHz)

--- Get Timer Information ---
NtQueryTimerResolution(&MinResolution,&MaxResolution,&CurrentResolution) returns 0
MinResolution = 156001 (15.6 ms)
MaxResolution = 5000 (0.5 ms)
CurrentResolution = 156001 (15.6 ms)
--- Get Current Resolution ---
NtSetTimerResolution(10000,0,&CurrentResolution)
CurrentResolution = 156001 (15.6 ms)

 
Quote:
 
GetModuleHandle("ntdll.dll" ) returns 0x77760000
GetProcAddress(ntdll.dll,"NtSetTimerResolution" ) returns 0x77781C44
GetProcAddress(ntdll.dll,"NtQueryTimerResolution" ) returns 0x77781598
--- Get Performence Counter Frequency ---
QueryPerformanceFrequency returns 14318180 /s (14318 /ms)

Possible Timer: HPET (14.32 MHz)

--- Get Timer Information ---
NtQueryTimerResolution(&MinResolution,&MaxResolution,&CurrentResolution) returns 0
MinResolution = 156001 (15.6 ms)
MaxResolution = 5000 (0.5 ms)
CurrentResolution = 100000 (10.0 ms)
--- Get Current Resolution ---
NtSetTimerResolution(10000,0,&CurrentResolution)
CurrentResolution = 100000 (10.0 ms)

 
and the goal is:
 
Quote:
CurrentResolution = 5000 (0.5 ms)

 
HPET and PM Timer are much more reliable timers than the 30yo RTC fosho, hah! anyway case closed on my side 
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Dec 18, 2012 at 12:17 PM Post #179 of 188
Hum, I was under the impression the timer dropped to 1ms when in use, and custom settings were required to lower it to 0.5ms.
 
Do you happen to know of any method to lower it below 0.5ms?
 
Dec 18, 2012 at 1:27 PM Post #180 of 188
Quote:
The first thing I would try to do is to get your pc and software setup to run in a "bit-perfect" manner...this setup will allow the bits from the source to flow through the pc to the DAC and to your AMP etc..and hear what the orignal cd or source intended.....your source files should be in a lossless format like FLAC etc...  
Once this bitperfect mode is setup you can then decide if you really need to manipulate the audio in anyway....I tend not to, and i look for recordings that are world class and mastered very well.
 
But I agree that there are many recordings that are compressed and sound like crap and you may want to tinker with them via an equilizer etc....but i dont like to waste the time with crappy recordings....there is too much good stuff our there to listen to.....which is another whole topic.
 
I find things like JPLAY are absolutely snake oil.... [...]

Yeah I couldn't agree more with you. But there's always some gullible people that will buy into pretty much anything.
 

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