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post #21 of 30
Quote:
Originally Posted by SCollins View Post
12ms, audio latency ? not gonna happen. Look up DAC rate of coversion. Its on the order of 4-6msec. Now if your talking about outbound only latency, I can get down to 7-9msec on average.

I was talking about round trip latency, input to output. If I send a signal into the mic input I can round trip it back out on software to the output in 16msec range.

Windows ASIO nor Linux can do this. I know, I have tried them both, best I can on widnows with a VERY optimized system on software monitoring, is around 70msec sometimes in the high 60's.
If you were unable to achieve low latencies in Windows, then you either have a cheap sound card, bad drivers, or both.

Quote:
Originally Posted by SCollins View Post
Hook a scope up and a signal generator, start measuring your latency's.
You asked for it.

RME Fireface 800 in Windows 7 x64 with 48 sample buffer size at 44.1 kHz sampling rate:



Direct monitoring is disabled in Cubase. This means that the signal is going through Cubase before being routed back to the interface:



Here's a 50 Hz square wave (top trace) making a round trip through the RME and Cubase with only 5.43 ms latency (bottom trace):



I generally run it with a 64 sample buffer, but that still maintains a round-trip latency of only 6.15 ms.

This kind of performance doesn't come cheap though. This RME generally sells for about $1600 new.
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post #22 of 30
Thread Starter 
@mushroomboy

Quote:
There is no ALSA -> PulseAudio plugin, that's impossible. PulseAudio acts to catch all the ALSA audio streams before they hit dmix/ALSA. It then does it's internal software mixing and passes the sound to ALSA. What your describing would create a loop, there is no way ALSA can output to PulseAudio and then back to itself. Them be paradox waters my friend, it always goes to ALSA last.
Not quite. PulseAudio does indeed pass the audio to ALSA; I assume that like JACK it knows the difference between a hardware device and a software device, so PulseAudio itself will use the "hw:0" ALSA device. However, my .asoundrc looks like this:

Code:
pcm.!default {
     type pulse
}

ctl.!default {
     type pulse
}
I can only assume this interposes an ALSA interface between the calling application and the PulseAudio server, for those applications that can only talk to ALSA, such as the latest versions of WINE.

I know this works because I can tell an application (such as VLC) to use the default ALSA device (rather than connecting directly to the audio server) and see the PulseAudio VU Meter become active as the audio is played.

Additionally, this is taken from the libasound2-plugins package for Ubuntu 10.04 LTS:

Quote:
This package contains plug-ins for the ALSA library that are not included in the main libasound2 package.

The following plugins are included, among other:
- jack: play or capture via JACK
- oss: run native ALSA apps on OSS drivers
- pulse: play or capture via Pulse Audio
- samplerate and speexrate: rate converters
- upmix and vdownmix: convert from/to 2 and 4/6 channel streams

ALSA is the Advanced Linux Sound Architecture.
According to the description, there are ALSA plug-ins for PulseAudio, OSS and JACK. Or am I missing something?
Edited by parityboy - 10/8/11 at 12:09pm
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post #23 of 30
Quote:
Originally Posted by parityboy View Post
@mushroomboy



Not quite. PulseAudio does indeed pass the audio to ALSA; I assume that like JACK it knows the difference between a hardware device and a software device, so PulseAudio itself will use the "hw:0" ALSA device. However, my .asoundrc looks like this:

Code:
pcm.!default {
     type pulse
}

ctl.!default {
     type pulse
}
I can only assume this interposes an ALSA interface between the calling application and the PulseAudio server, for those applications that can only talk to ALSA, such as the latest versions of WINE.

I know this works because I can tell an application (such as VLC) to use the default ALSA device (rather than connecting directly to the audio server) and see the PulseAudio VU Meter become active as the audio is played.

Additionally, this is taken from the libasound2-plugins package for Ubuntu 10.04 LTS:



According to the description, there are ALSA plug-ins for PulseAudio, OSS and JACK. Or am I missing something?
Those aren't plugins per say but extend the lib function of ALSA. I would explain but I'm not exactly sure how the libs are used to interact with the kernel code, somebody better could probably explain that.

Also, your .asoundrc is just setting your system to default to pulse. =( So when your master sounds go out they go to pulseaudio and then pulse decides where they go. Your basically forcing dmix out of the picture, there is no real ALSA->Pulse->ALSA. What really happens is Program (ALSA) -> Pulse -> ALSA instead of Program (pulse) -> Pulse -> ALSA. Programs never really talk to ALSA, they just output sound according to ALSA standards.
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post #24 of 30
Thread Starter 
Actually I finally got dmixer to work correctly, the issue was the microphone (needed for TS). Found the right settings, and it appears to be stable.

So here's the next question: if every distro configured the audio properly and didn't leave it up to the user to figure out how to do it (i.e. configure dmix properly), what value would sound servers have?
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post #25 of 30
Quote:
Originally Posted by parityboy View Post

So here's the next question: if every distro configured the audio properly and didn't leave it up to the user to figure out how to do it (i.e. configure dmix properly), what value would sound servers have?
not sure if any distro would ever do this, ubuntu tries and you can see the problems there .

sound has always been a linux problem, its improved considerable in the last few years. i myself prefer pulse, when it first came out it was riddled with problems, but the last year or so, i've had no problem with it.

just out of curiosity, in one of your post, you mentioned you needed wine and teamspeak to work with alsa/pulse correctly, are you running teamspeak thru wine? there is a native ts3 client for linux, that works quite well and imho, is as feature complete as its windows counterpart, unlike say skype.
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post #26 of 30
Quote:
Originally Posted by valvehead View Post
If you were unable to achieve low latencies in Windows, then you either have a cheap sound card, bad drivers, or both.



You asked for it.

RME Fireface 800 in Windows 7 x64 with 48 sample buffer size at 44.1 kHz sampling rate:



Direct monitoring is disabled in Cubase. This means that the signal is going through Cubase before being routed back to the interface:



Here's a 50 Hz square wave (top trace) making a round trip through the RME and Cubase with only 5.43 ms latency (bottom trace):



I generally run it with a 64 sample buffer, but that still maintains a round-trip latency of only 6.15 ms.

This kind of performance doesn't come cheap though. This RME generally sells for about $1600 new.

cubase does not disable hardware monitoring on RME hardware, In fact the new Windows driver model for window7 actually natively supports Hardware loopback/monitoring. Your just outside the conversion rate for the DAC, thats your first clue. , I have Avid HD cards. Have had them for several years. If I stay on the cards, I get great latencys, its when things go outside the hardware on the cards that things change, rather dramatically.

BTW your transport loop on firewire is 7msec. So that must be hardware monitored.
Edited by SCollins - 10/8/11 at 4:16pm
post #27 of 30
Quote:
Originally Posted by SCollins View Post
cubase does not disable hardware monitoring on RME hardware, if it does, upgrade to cubase 5, I have Avid HD cards. Have had them for several years.

BTW your transport loop on firewire is 7msec. So that must be hardware monitored.
It is not hardware monitoring. If I apply effects in Cubase (eq, distortion, etc.) they affect the output without changing the latency.

Here's the same signal with an amp simulator, a graphic eq, and the channel eq activated:



The latency has not changed:



The RME does not have general purpose VST processing. The signal has to go through the CPU in order for effects to be applied. This cannot happen unless hardware monitoring is disabled.

Granted there is a limit to how many and what kind of effects can be applied with a 48 sample buffer. That's why I generally stick with 64 samples while tracking.


@parityboy:
I apologize for derailing this thread. My posts were merely to illustrate that with the right hardware and drivers, very low latency can be achieved.

On topic: I really hope that there will be better sound hardware support in Linux soon. There are a lot of people (me included) who are interested in giving Linux audio production a shot but can't/don't want to abandon their hardware. I really want to give Ardour a try, but it would have to use my onboard sound.
Edited by valvehead - 10/8/11 at 5:18pm
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post #28 of 30
Greetz
Trying to keep this reasonably concise...
1) If anyone thinks there is an indiscernible difference with latencies under 20msec, or even under 10msec, then they have never worked with lower latencies.

2) If you believe commercial equipment advertised latencies, or more accurately, don't realize they are talking about an idealized condition within the dedicated, embedded device itself, imho you are kidding yourself. Anyone with any time in Pro Audio is familiar with how fast and loose the Sales Department plays with specs "Our auto sound system amplifiers produce a devastating 500 watts from a single car battery and they are flat from Gravity to Light" - or - "Achieve 4msec latency with our DJ box!"

It is possible to achieve latencies on the order of 2msec within a single, simple dedicated device, but when you insert it is a system, while it may add less latency than a cheap alternative, the round trip system latency is additive I don't care how much you spend you cannot escape that basic physical fact. In a more or less dedicated PC the ability to recompile the kernel is of extreme importance if it is to be inserted into the loop.

To try to get somewhat On Topic, choose hardware very carefully with ANY OpSys because all drivers are not equal and this is even more true in alternative systems like Linux where some are actually better than for OSX or Windows, but many (maybe even, most) are not and even the best drivers are "brought down" from what could be possible by generally dismal sound daemons. Someday, someone will make it big with a truly customizable, dedicated Pro Audio computer and I would bet money it will NOT be Apple or Microsoft.
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post #29 of 30
Thread Starter 
Quote:
Originally Posted by transhour View Post
not sure if any distro would ever do this, ubuntu tries and you can see the problems there .

sound has always been a linux problem, its improved considerable in the last few years. i myself prefer pulse, when it first came out it was riddled with problems, but the last year or so, i've had no problem with it.

just out of curiosity, in one of your post, you mentioned you needed wine and teamspeak to work with alsa/pulse correctly, are you running teamspeak thru wine? there is a native ts3 client for linux, that works quite well and imho, is as feature complete as its windows counterpart, unlike say skype.
If Ubuntu used dmix instead of PulseAudio, and configured .asoundrc the way I have, the chances are there would be much fewer issues with sound.

To address your second point, I am indeed using the Linux version of TeamSpeak. The issue I had with it was that while the playback would work, the audio capture wouldn't; on the mic test, the VU meter would go into the red then freeze.

I also notice that TeamSpeak performance is much better with ALSA & dmix than with TeamSpeak->PulseAudio or with TeamSpeak->PulseAudio->JACK. Obviously, removing layers removes latency.
Edited by parityboy - 10/9/11 at 8:36am
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post #30 of 30
Quote:
Originally Posted by valvehead View Post
It is not hardware monitoring. If I apply effects in Cubase (eq, distortion, etc.) they affect the output without changing the latency.

Here's the same signal with an amp simulator, a graphic eq, and the channel eq activated:



The latency has not changed:



The RME does not have general purpose VST processing. The signal has to go through the CPU in order for effects to be applied. This cannot happen unless hardware monitoring is disabled.

Granted there is a limit to how many and what kind of effects can be applied with a 48 sample buffer. That's why I generally stick with 64 samples while tracking.


@parityboy:
I apologize for derailing this thread. My posts were merely to illustrate that with the right hardware and drivers, very low latency can be achieved.

On topic: I really hope that there will be better sound hardware support in Linux soon. There are a lot of people (me included) who are interested in giving Linux audio production a shot but can't/don't want to abandon their hardware. I really want to give Ardour a try, but it would have to use my onboard sound.
Given that your latency is lower then the windows interupt qauntum, I am begging to doubt the accuracy of your scope. does it have a single clock for all channels or a clock for each channel ? also given that you added a DSP load, I really have a hard time belive that number. Also, audio is not a sqaure wave. you need to set that scope to Sine and AC coupling.
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