IPB

Welcome Guest ( Log In | Register )

 
Reply to this topicStart new topic
Microphone Delay
mzso
post Apr 18 2012, 12:43
Post #1





Group: Members
Posts: 177
Joined: 2-May 07
Member No.: 43131



Hi!
I bought a microphone, plugged it in and I have an irritating few tenth of a second delay. Really annying becuse Its basically impossible to talk when you're hearing your ownvoice, withthis delay. I guess it would be similarly irritating if I would chat with someone else. Where should I look for the problem? (I have a xonar D1 with UNi Xonar drivers, if its any relevance)
Go to the top of the page
+Quote Post
Dynamic
post Apr 18 2012, 13:34
Post #2





Group: Members
Posts: 803
Joined: 17-September 06
Member No.: 35307



Do you need to hear your own voice played back?

Try your driver settings.
If you have the Magic Voice or any of the DSP features in your input or output chain, they will have a degree of latency. Switching such effects to zero or flat or plain, rather than bypass or disable, will still involve the buffering delay. It might be enough to cause the problems you hear.
Go to the top of the page
+Quote Post
mzso
post Apr 18 2012, 14:23
Post #3





Group: Members
Posts: 177
Joined: 2-May 07
Member No.: 43131



I only have the virtual speaker dsp. But en/disabling it doesn't change anything.

This post has been edited by mzso: Apr 18 2012, 14:24
Go to the top of the page
+Quote Post
Dynamic
post Apr 18 2012, 15:03
Post #4





Group: Members
Posts: 803
Joined: 17-September 06
Member No.: 35307



It's not a problem I get, but I don't have that soundcard. Firstly, because I don't need to hear back my recorded line, so I don't get any audio feeedback.

My soundcard has a direct monitoring output in analogue, and I don't use any DSPs on soundcard or Windows mixer, except occasionally in the application level (such as fb2k).

Virtual speakers (presumably for headphone use) would have a modest delay, but I'd be surprised it needs to be so big as that. Try removing it to troubleshoot [edit: I know you have removed it and still have a problem, but if there are two similar sources of delay, you might miss the fix by leaving one turned on while hunting]. Also beware that you might inadvertently be stacking both an internal hardware DSP delay in the Xonar and one in the Windows mixer system (which might be doing sample-rate conversion and mixing). If you use (intentionally or otherwise) very high quality SRC in either, that's likely to introduce a delay.

Some filter designs in DSP want to look beyond the 'current' sample so look ahead for so many samples, more samples ahead if you set them to higher quality mode. Some are 'causal' (just like electronic passive filters) and can't look 'ahead' into the 'future' stream of samples, so can only rely on present and past sample values to determine the output. Many sound DSPs assume you normally feed only recorded, pre-buffered audio to them so will introduce a delay to have enough 'future' samples to process because a non-causal design produces better performance in other ways.

I'm guessing you might be using Windows 7? The OS could have an influence.

This post has been edited by Dynamic: Apr 18 2012, 15:06
Go to the top of the page
+Quote Post
Brand
post Apr 18 2012, 15:58
Post #5





Group: Members
Posts: 317
Joined: 27-November 09
Member No.: 75355



Some programs let you choose ASIO or WASAPI, which would give you lower latency.

And do you need to hear your voice when speaking?
Go to the top of the page
+Quote Post
mzso
post Apr 19 2012, 10:01
Post #6





Group: Members
Posts: 177
Joined: 2-May 07
Member No.: 43131



QUOTE (Brand @ Apr 18 2012, 15:58) *
Some programs let you choose ASIO or WASAPI, which would give you lower latency.

And do you need to hear your voice when speaking?


Latency is bad even if only the other hears it. Plus its good to know what the other person actually hears if I chat.

QUOTE (Dynamic @ Apr 18 2012, 15:03) *
It's not a problem I get, but I don't have that soundcard. Firstly, because I don't need to hear back my recorded line, so I don't get any audio feeedback.

My soundcard has a direct monitoring output in analogue, and I don't use any DSPs on soundcard or Windows mixer, except occasionally in the application level (such as fb2k).

Virtual speakers (presumably for headphone use) would have a modest delay, but I'd be surprised it needs to be so big as that. Try removing it to troubleshoot [edit: I know you have removed it and still have a problem, but if there are two similar sources of delay, you might miss the fix by leaving one turned on while hunting]. Also beware that you might inadvertently be stacking both an internal hardware DSP delay in the Xonar and one in the Windows mixer system (which might be doing sample-rate conversion and mixing). If you use (intentionally or otherwise) very high quality SRC in either, that's likely to introduce a delay.

Some filter designs in DSP want to look beyond the 'current' sample so look ahead for so many samples, more samples ahead if you set them to higher quality mode. Some are 'causal' (just like electronic passive filters) and can't look 'ahead' into the 'future' stream of samples, so can only rely on present and past sample values to determine the output. Many sound DSPs assume you normally feed only recorded, pre-buffered audio to them so will introduce a delay to have enough 'future' samples to process because a non-causal design produces better performance in other ways.

I'm guessing you might be using Windows 7? The OS could have an influence.

I actually tried with the built in soundcard. Same thing. And yes I'm using windows 7. I tried matching the input sample rate to the output one, but nothing changed. Out of ideas. It was working without noticable delay on my brothers computer without setting anything...
Go to the top of the page
+Quote Post
Arnold B. Kruege...
post Apr 19 2012, 12:12
Post #7





Group: Members
Posts: 3689
Joined: 29-October 08
From: USA, 48236
Member No.: 61311



QUOTE (mzso @ Apr 18 2012, 07:43) *
Hi!
I bought a microphone, plugged it in and I have an irritating few tenth of a second delay. Really annying becuse Its basically impossible to talk when you're hearing your ownvoice, withthis delay. I guess it would be similarly irritating if I would chat with someone else. Where should I look for the problem? (I have a xonar D1 with UNi Xonar drivers, if its any relevance)


As you are no doubt aware, your problem is latency.

There is software that analyzes the latency in audio interfaces. I don't know any names, but its out there and free. Google is your friend, but I bet someone reading this thread in the next day or two has a recommendation based on personal experience.

I suspect that you could measure the latency of the overall system by setting up a feedback loop and priming it with an acoustic click, and then analyzing the recording.

Other than the thrill of hearing yourself sound cool, there's really no reason to listen to your voice while you are recording. Bone conduction, room reverb, and all that. You are able to talk and sign without a monitor speaker going on in the background, right?

The reason vocalists use monitor speakers for live sound and recording is to hear the instrumental mix, and because the sound field from other sources may be so loud they have a problem hearing themselves. Neither should apply to you.
Go to the top of the page
+Quote Post
mzso
post Apr 19 2012, 13:32
Post #8





Group: Members
Posts: 177
Joined: 2-May 07
Member No.: 43131



QUOTE (Arnold B. Krueger @ Apr 19 2012, 12:12) *
QUOTE (mzso @ Apr 18 2012, 07:43) *
Hi!
I bought a microphone, plugged it in and I have an irritating few tenth of a second delay. Really annying becuse Its basically impossible to talk when you're hearing your ownvoice, withthis delay. I guess it would be similarly irritating if I would chat with someone else. Where should I look for the problem? (I have a xonar D1 with UNi Xonar drivers, if its any relevance)


As you are no doubt aware, your problem is latency.

There is software that analyzes the latency in audio interfaces. I don't know any names, but its out there and free. Google is your friend, but I bet someone reading this thread in the next day or two has a recommendation based on personal experience.

I suspect that you could measure the latency of the overall system by setting up a feedback loop and priming it with an acoustic click, and then analyzing the recording.

Other than the thrill of hearing yourself sound cool, there's really no reason to listen to your voice while you are recording. Bone conduction, room reverb, and all that. You are able to talk and sign without a monitor speaker going on in the background, right?

The reason vocalists use monitor speakers for live sound and recording is to hear the instrumental mix, and because the sound field from other sources may be so loud they have a problem hearing themselves. Neither should apply to you.

Its usefull to hear your own voice if you voice-chat. Also delays like these are unwanted even if only the chat partner hears it.

Latency for sure. I'm just out of ideas why....
Go to the top of the page
+Quote Post
Arnold B. Kruege...
post Apr 21 2012, 15:30
Post #9





Group: Members
Posts: 3689
Joined: 29-October 08
From: USA, 48236
Member No.: 61311



QUOTE (mzso @ Apr 19 2012, 08:32) *
QUOTE (Arnold B. Krueger @ Apr 19 2012, 12:12) *
QUOTE (mzso @ Apr 18 2012, 07:43) *
Hi!
I bought a microphone, plugged it in and I have an irritating few tenth of a second delay. Really annying becuse Its basically impossible to talk when you're hearing your ownvoice, withthis delay. I guess it would be similarly irritating if I would chat with someone else. Where should I look for the problem? (I have a xonar D1 with UNi Xonar drivers, if its any relevance)


As you are no doubt aware, your problem is latency.

There is software that analyzes the latency in audio interfaces. I don't know any names, but its out there and free. Google is your friend, but I bet someone reading this thread in the next day or two has a recommendation based on personal experience.

I suspect that you could measure the latency of the overall system by setting up a feedback loop and priming it with an acoustic click, and then analyzing the recording.

Other than the thrill of hearing yourself sound cool, there's really no reason to listen to your voice while you are recording. Bone conduction, room reverb, and all that. You are able to talk and sign without a monitor speaker going on in the background, right?

The reason vocalists use monitor speakers for live sound and recording is to hear the instrumental mix, and because the sound field from other sources may be so loud they have a problem hearing themselves. Neither should apply to you.

Its usefull to hear your own voice if you voice-chat. Also delays like these are unwanted even if only the chat partner hears it.

Latency for sure. I'm just out of ideas why....


If there are device driver buffer parameters that you can decrease and maintain reliable operation, then adjust that.

Other than that, the latency came with the hardware you bought - no extra charge! ;-)
Go to the top of the page
+Quote Post
Roseval
post Apr 21 2012, 18:28
Post #10





Group: Members
Posts: 476
Joined: 26-March 08
Member No.: 52303



Try
http://www.thesycon.com/deu/latency_check.shtml
http://www.resplendence.com/latencymon


--------------------
TheWellTemperedComputer.com
Go to the top of the page
+Quote Post
andy o
post Apr 22 2012, 20:09
Post #11





Group: Members
Posts: 1310
Joined: 14-April 09
Member No.: 68950



I think this latency is not related to audio hardware, he's chatting, presumably over the internet (?).
Go to the top of the page
+Quote Post
YellowOnion
post May 4 2012, 06:01
Post #12





Group: Members
Posts: 28
Joined: 9-April 12
Member No.: 98589



Several things can cause latency,

Buffers, most systems have quite large buffers of 100ms, unless you know what you're doing you will need a large buffer, when your computer switches tasks you'll likely get buffer underruns, to work with a small buffer size you need Real-time processing and a special low latency driver interface like ASIO or jackd, and also turning power saving features off on your CPU as there's usually a delay in switching frequencies (Linux has a 10ms delay, I can get a stable 1.5ms buffer on Linux with jackd if I turn off power features)

most voice codecs work with lower frequency sample rates to save bandwidth
Resampling algorithms require a buffer as well

voice codecs usually don't add to much delay specifically for this reason, but they're all going to add at least 10ms, a codec like Vorbis adds up to 200ms delay

if you want to drop as much latency as possible, use ASIO if possible, set the 'default' sample rate for the recording device in windows to match the VoIP software if you can't use ASIO.

if you can't handle it still and don't care about the VoIP software

use a high performance voice codec like CELT or Opus, I believe the Team Speak and Mumble both use CELT, and Mumble has ASIO driver support.

I setup a Mumble server at my house, and a friend was in his room, the only difference in delay was a slight phase shift across the hall, this was even without ASIO, and the delay was still impressively low when we joined a mumble server 10ms away.


more technical info here: http://people.xiph.org/~xiphmont/demo/celt/demo.html
Go to the top of the page
+Quote Post
mzso
post May 4 2012, 07:13
Post #13





Group: Members
Posts: 177
Joined: 2-May 07
Member No.: 43131



QUOTE (YellowOnion @ May 4 2012, 06:01) *
[...]

Thanks for the tips.
Go to the top of the page
+Quote Post

Reply to this topicStart new topic
1 User(s) are reading this topic (1 Guests and 0 Anonymous Users)
0 Members:

 



RSS Lo-Fi Version Time is now: 22nd August 2014 - 00:27