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WASAPI plug-in version 3.0 beta [closed], Discussion & feedback
jaro1
post Jun 25 2012, 11:27
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"if possible; please add an option to "run with high(est) process priority" --- like option of ASIO (v2.1.2) output plugin" - with beta5 this has been done already, it is sure now that possible clicks aren't caused by insufficient priority. My experience with onboard low latency-ready (event mode) audio under W7 (wavert supported - some Via or Realtek HD) is flawless playback also during heavy CPU load. I don't know the case with usb devices,no possibility for me to test it. I am more than sure, the same behavior will be also with pci/pci-e sound cards, that properly support this mode through their drivers.
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Donunus
post Jun 27 2012, 02:31
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I got a blue screen of death on windows 7 when using wasapi beta 3, 4, and 5. I realized that at all times it happened I was ripping a cd on dbpoweramp while playing music on foobar. Just wanted to share in case you could use that info for anything.
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elik_r
post Jun 27 2012, 19:33
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QUOTE (elik_r @ Jun 19 2012, 09:50) *
QUOTE (elik_r @ Jun 15 2012, 20:52) *
I'm using AudioEngine's D1 DAC.

My best experience is using beta 4. Smooth as ever, no crackling whatsoever.
I play mostly flac files, some hi resolution ones as well.



After further testing, the only problems I'm experiencing is when I play "non conventional" combinations of bitrates and samplerates.
For instance, playing 24/96 flac is usually smooth (although more prone to errors due to cpu load), but playing 24/88 files is glitchy from the start.




Beta 5 properly solved all my troubles.
I'm sticking with this one until someone forces me otherwise.
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jologsmaster
post Jun 29 2012, 12:37
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Update:

Beta 5 event mode still works perfectly with my HRT Musicstreamer II USB DAC with firmware rev 2.1.

Strictly 16-bit 44.1kHz FLAC/MP3/etc. file playback.

Foobar output settings at 1000ms buffer length and 24-bit output data format. DSP and Replaygain deactivated.
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Sandrine
post Jun 29 2012, 23:29
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QUOTE (Sandrine @ Jun 14 2012, 10:35) *
If I may add my 2 cents: The only valid version so far for me personally has been beta2.


I have to amend this. At least beta5 also does pass-through. However, through some trial-and-error I found that the DSP component "Advanced Limiter" seems to alter the stream to such an extent that passthrough may be impossible.

So, thumbs up for beta5!

This post has been edited by Sandrine: Jun 29 2012, 23:29
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Canar
post Jun 29 2012, 23:58
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QUOTE (Sandrine @ Jun 29 2012, 15:29) *
I have to amend this. At least beta5 also does pass-through. However, through some trial-and-error I found that the DSP component "Advanced Limiter" seems to alter the stream to such an extent that passthrough may be impossible.
Adding DSPs will break passthrough in almost all cases.


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Bradapalooza
post Jul 1 2012, 19:10
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Currently using beta 5 with no noticed issues so far. Operating system is Windows 7 Enterprise SP1, 64 bit. Buffer is set to 1000ms, onboard audio is a Realtek ALC269 with w/e my computer manufactures latest drivers are for it. All music played has all been FLAC, that was ripped from my personal CD collection using accuraterip, paranoid settings, and the recorded offset of my drive found via accuraterip's data base. So it's all 2 channel, 16 bit, 44100 Hz. Currently letting the realtek do its thing and going straight from the headphone jack via a 3.5 that y-splits to analog RCA into the wireless headphone transmitter for my gaming headphones (Turtlebeach x41 model). Just updated Foobar to 1.1.13 today. My testing yesterday was with 1.1.12 (also no issues).

Only question I have is the presence of both a WASAPIHost32.exe and a WASAPIHost64.exe in the WASAPI component folder. In task manager I see WASAPIHost64.exe running while playing music, is it safe to assume I can remove WASAPIHost32.exe with no issues?

I also have access to a USB soundcard if you'd like me to do any testing with that (its the Turtle Beach Amigo II, usually I only use it as an adapter to let me use my headphone mic w/ voiceplayback, but I could use it for audio playback testing.
I also have S/PDIF out (coaxial) but I'm waiting on a few components to arrive (should be sometime this week) before I start using it. Once I receive those components I can go SPDIF/coaxial out -> FiiO D3 DAC (via RCA) -> Schitt Valhalla -> Beyerdynamic DT-990 (600ohms) for additional testing (and my primary listening source).

Let me know if there's anything you'd like me to try specifically - currently I'm only using regular mode - not event. But I'm happy to help with testing on such a great component, thanks for your work on it!

(Sidenote: I typically set foobar2k to above normal priority within task manager as well, and I use no DSPs, foobar set to output at 16bits (since all the files I play are 16 bit I assume this is the proper choice))

This post has been edited by Bradapalooza: Jul 1 2012, 19:12
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Bradapalooza
post Jul 3 2012, 02:31
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Sorry for the double post - would edit in this report but I suppose to much time has passed to allow me to edit it.

Had issues with playback today, it began skipping, then came to a full stop for a few seconds after which it went back to skipping before I manually paused it. CPU was under a heavy load (60-70%), as well as RAM, and the program didn't actually crash; I suppose I was just asking the computer to do to much and it decided playback didn't need all that it was requesting. I would consider the skipping under that level of a CPU load to be a WASAPI fixable issue - as it merely needed to be recognized as higher priority, but since I wasn't monitoring the usage of my ram at the moment when this occurred I can't confirm that. I suspect that I simply demanded too much of my RAM while simultaneously having WASAPI output settings at a 1000ms buffer.

And despite foobar2k and the WASAPI.exe being run at above normal priority, its perfectly viable it wasn't a WASAPI issue - as the heavy load I told my computer to do was asking the iPod manager component of Foobar2k to wipe what was on my iPod (perhaps 6 gigs worth of music) and then convert 10.7 gigs worth of FLAC into ALAC and place it all on the iPod without actually creating permanent ALAC files on my computer. Sooooooo, maybe that was a bit much to handle since it was a request within foobar (giving it the same priority of the WASAPI.exe; not to mention the fact that the WASAPI has to ask for stuff from F2k).

Still, I figured it was worth mentioning, since it was the absolute first issue I've had with playback that could possibly be attributed to WASAPI v3 beta 5.

Also still curious if I can get rid of the WASAPIHost32.exe since I only ever see 64 operating within taskmanager. And still open to testing playback with my previously mentioned (not yet tested) available resources (USB sound card, or S/PDIF coax out to a DAC). I can attempt to replicate any issues anyone's having with v5 beta, as I have access to all the methods of playback I've seen mentioned up to this point.

Happy to help in anyway I can with this
-Bradapalooza
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jaro1
post Jul 3 2012, 07:30
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I'm convinced the basic goal of creating the component that works flawlessly with the right drivers (WaveRT) was done already. MMCSS usage was important step for utilizing all the possibilities WV/7 audio stack can give, so thanks for this.
The problem is, Peter tries to make it working on wider range of audio hardware and here could be difficult to make it right for all of them without the possibility to adjust the settings manually. Maybe it increases the complexity of the component, but with additional options of the component everyone can find the right values for his own piece of hardware.
My idea is to have the possibility to adjust manually these basic settings:
- data feed mode (pull-event or push-regular mode), this has been done already
- render thread task type (Audio, Pro Audio, Playback) because of the right priority
- latency in ms
With some non low latency-ready hardware and drivers Pro Audio setting together with very low latency value can cause too high CPU cycles and glitching, so it isn't universal setting and my own best experience with onboard audio is to use thread task type Audio and latency in 10-50 ms range, but these values can be unsuitable for another hardware, of course. All this can be solved with manual adjustment in component options window.

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frustrated
post Jul 4 2012, 23:00
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I wonder if somebody could help me. I have been using an HRT musicstreamer II for a while now, with foobar2000 and kernel streaming with a buffer of 50ms, as recommended by the manufacturer due to the asynchronous nature of the streamer. It was generally fine, but suffered occasional glitches.

I then found the event styale WASAPI plugin (the standard one was terrible in my setup), and was very pleased, not a single glitch with 16/44.1 flac after days of listening. I then played 24/96 flac files and all was fine for about 30 mins, then it started to glitch.

My question is whether my continued use of 50ms buffer is the correct choice. Given the pull nature of event style, do I need to set the buffer much higher instead of as low as possible, as was required for push mode.

Any insights or recomendations would be very gratefully received. I know I could experiment with the buffer size, but just thought someone on here could maybe help with this process given their knowledge of the difference between push/pull.

BTW, I am using beta 5.



Many thanks in advance.
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A_Man_Eating_Duc...
post Jul 5 2012, 00:41
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Are you running the latest firmware on your HRT Musicstreamer II?


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frustrated
post Jul 5 2012, 07:28
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QUOTE (A_Man_Eating_Duck @ Jul 5 2012, 00:41) *
Are you running the latest firmware on your HRT Musicstreamer II?



Thanks for replying.

I am using firmware 2.1, which I think is the latest.
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frustrated
post Jul 8 2012, 00:54
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Anybody?

I wrote to HRT as well, but have had no response either.

Can I assume that there is no simple answer to whether asynchronous DACs need low or high buffers for event mode WASAPI?

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jologsmaster
post Jul 8 2012, 04:38
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HRT user here using WASAPI 3.0 Beta 5 event mode.

I usually stick with 16-bit 44.1khz audio files for the most part which is probably the main reason why I don't get those glitches anymore. I rarely switch to and from different formats on the fly.

I also have the hardware buffer set at 1000ms without any problems whatsoever. It seems like with beta 5, the hardware buffer settings don't matter much anymore.

Maybe you can try using a self powered USB hub with the DAC as well as a higher quality USB cable. HRT DACs are known to require quite a bit of power from the USB port to function properly.

Another thing is to make sure that the foobar2000 WASAPI output bit depth is set at the same bit depth with the sound format settings in the windows control panel and vice versa. That solved the muting issues that I was having recently.

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frustrated
post Jul 8 2012, 09:08
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Thanks mate.

Like you, I don't get any glitches at all with 16/44.1 flac. However, I do get occasional glitches with 24/96. Running from ramdisk does help, I'll just have to try and see if there is a sweet

spot for the buffer length when running High Def.

If I find one, I'll let you know.

Thanks again.
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WarwickBuz
post Jul 9 2012, 04:16
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Been struggling all week with glitches on a USB port to a Fiio E17.

My home machine has been rock solid, however on my work machine it just glitched. Tried every beta at every setting I could think of. They all worked for a bit then glitched away.

I changed USB ports and the front one was better than one in the back, but still I got the glitches.

I checked the USB driver in the machine (HP Z400) and Widows reported that it was the best USB driver.

After a week of this I went to the HP site and found there was a later USB driver for the machine.

I installed it and the performace has been rock solid ever since. I haven't been able to actually get it to glitch on any setting so far, although I haven't tried anything to extreme.


TL;DR - If you are having issues with glitches over USB make sure you have the latest USB drivers installed on your machine.
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redwald
post Jul 10 2012, 05:47
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HRT Music Streamer II with F/W 2.1, FB2000 1.1.13, WASAPI 3 beta5, Win7 64-bit.

Tried all possible combinations of Kernel Streaming, ASIO, WASAPI v2. Couldn't get glitch free playback unless I had buffer set really low (<50ms) Then I flashed F/W from 1.7 to 2.1 and installed WASAPI 3beta 5. Now I can get the buffer up to 1000ms without glitches. Also I don't know the difference between WASPI (Event) and regular WASPI but the (Event) works much better. Most of my files are 16-bit mp3 and FLAC/APE. I haven't tried any 24-bit files yet.
Sound from HRT into vintage gear is awesome.

Note that HRT MSII needs 250milliamp of clean current to do its job. Download the USBview app from their website to make sure its on it own bus and not sharing a USB port with some other device.

Thanks for all the work improving WASAPI for FB2000 Peter!

This post has been edited by redwald: Jul 10 2012, 05:59
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frustrated
post Jul 11 2012, 17:55
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Thanks for all your replies.

I think I've got all possible areas of weakness covered, such as USB ports, cables etc.

I used to use asio with win xp and got it working perfectly. However, since upgrading (?!) to win 7 (same computer)

I've been having problems. The most reliable I found to be kernel streaming at 50ms buffer, which glitches rarely.

I was just hoping that event WASAPI would remove the glitches permenantly. I've tried playing with the buffer

and think that it is slightly better with a bigger buffer, which leads me to think that ''pull mode'' has different requirements

to ''push mode''.
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Sandrine
post Jul 11 2012, 19:13
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QUOTE (frustrated @ Jul 11 2012, 18:55) *
since upgrading (?!) to win 7 (same computer) I've been having problems. The most reliable I found to be kernel streaming at 50ms buffer, which glitches rarely.


What are you talking about? There is no kernel streaming on Win 7.
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frustrated
post Jul 11 2012, 19:39
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http://wiki.hydrogenaudio.org/index.php?ti...rt_(foo_out_ks)

excerpt:


''This component was originally written for Windows 2000 and Windows XP. It is not guaranteed to cooperate with newer versions of Windows.

Kernel Streaming is known to work on certain Windows Vista and Windows 7 configurations, but not with devices having WaveRT drivers such as High Definition Audio Devices integrated with newer motherboards - such devices simply won't be shown on foobar2000's output device list as available KS devices.''


I was told to try this from HRT technical support, I told them I had win 7. I suppose I got lucky.




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Sandrine
post Jul 21 2012, 20:23
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QUOTE (Sandrine @ Jun 30 2012, 00:29) *
QUOTE (Sandrine @ Jun 14 2012, 10:35) *
If I may add my 2 cents: The only valid version so far for me personally has been beta2.
I have to amend this. At least beta5 also does pass-through. However, through some trial-and-error I found that the DSP component "Advanced Limiter" seems to alter the stream to such an extent that passthrough may be impossible. So, thumbs up for beta5!


Argh, I have to amend this again. It only worked once, after updating from beta2 to beta5. I guess the port or whatchmacallit was still programmed the "right" way from beta2 after the update. After restarting Windows, no more pass-through.

Someone also reported that channels were only set correctly in beta2... maybe this could have something to do with it?
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ailef
post Jul 22 2012, 08:43
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QUOTE (Mr.Grey @ May 18 2012, 08:24) *
QUOTE (Moofasa~ @ May 18 2012, 03:48) *
I assume this plug-in will only support exclusive mode?

working together a foobar2000 (exclusive mode) & other programs (non-exclusive)
with the previous version of plugin this was not possible

LynxTWO card




ah, thanks for the help, it now works for me using those settings.
got a Lynx L22
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Hoppla
post Aug 2 2012, 15:21
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So i have this testing question.

using Realtek ALC889A with S/PDIF (optical cabel)
When looking in the Realtek configuration options i can choose


16bit, 44100hz (cd)
to
24bit, 192000hz (studio)
and then also
Dolby Digital Live 5.1 (surround)

So i figured that i use the highest my card can do (24bit, 192000hz) and 24bit in wasapi foobar output, then 50-1000ms
on 3.0 beta 5 (event mode) it works the same but what about this DDL 5.1? the volume gets louder when turning it on but i don't know how it works together
with this WASAPI plugin. What should i use? 2-channel only?

// Hoppla
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SEMteXXL
post Aug 2 2012, 16:41
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Hello,

Recently I found your new beta update but it didn't help with my long time problem. I have been using wasapi plugin 2.1 with some strange bass redirection problem (Flex bass) on 24/96 hirez audio files. Immediately after playing something above 44.1 KHz bass redirection stops working. Same problem with KS output. Everything ok on redbook files (16/44.1). I thought this is asus driver related issue but right after switching on primary or DS driver bass is here. I'm using Xonar d2x (7.12.8.1794 driver) card on win 7 ultimate with 24 bit output. It's seems that this is some kind of never ending story for me. Anybody else having the same issue? Thanks for help.

This post has been edited by SEMteXXL: Aug 2 2012, 16:48
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elik_r
post Aug 5 2012, 10:01
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QUOTE (elik_r @ Jun 27 2012, 21:33) *
QUOTE (elik_r @ Jun 19 2012, 09:50) *
QUOTE (elik_r @ Jun 15 2012, 20:52) *
I'm using AudioEngine's D1 DAC.

My best experience is using beta 4. Smooth as ever, no crackling whatsoever.
I play mostly flac files, some hi resolution ones as well.



After further testing, the only problems I'm experiencing is when I play "non conventional" combinations of bitrates and samplerates.
For instance, playing 24/96 flac is usually smooth (although more prone to errors due to cpu load), but playing 24/88 files is glitchy from the start.




Beta 5 properly solved all my troubles.
I'm sticking with this one until someone forces me otherwise.



Been using beta 5 for more than a month now.
Very good results.

The only problem I've encountered is glitches caused during the closing of google chrome tabs.
I'm using win7x64.
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