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MikeEnRegalia 
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 Topic: Recording music on the PC ...
Posted: July 22 2008 at 05:19 |
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Recently I began to experiment with music recording ... I recorded some tracks almost 10 years ago, but it was a totally different situation back then, there was no USB, no ASIO etc..
Yesterday I installed a demo version of Ableton Live 7 and did some tests with my trusty old Line6 GuitarPort. It turned out that I can't seem to get the ASIO latency below 10ms (buffer size 512, 96khz, 24bit) ... I was wondering whether some of you managed to get a better value.
I also ordered a Line6 KB37 ... an audio interface with integrated midi keyboard. Maybe - since it is much newer than the GuitarPort - I will get better latency with it.
BTW: The cool thing about the Line6 devices is that with their ToneDirect (TM) monitoring you have almost zero latency when recording audio through the Line6 interface inputs, so the latency of the ASIO drivers is no problem there. But as soon as you start recording other instruments - drums, keyboards etc. - the output latency of the device becomes a problem.
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Posted: July 22 2008 at 05:39 |
If you monitor the Line 6 directly, you'll get 0 latency - it's only if you plug your phones or speakers into the PC that you get monitoring latency.
I'm making the assumption that the GuitarPort uses the same technology as my TonePort, which I use as input for all instruments.
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MikeEnRegalia
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Posted: July 22 2008 at 06:10 |
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^ thanks - I know about direct monitoring, but when you record drums or keyboards in your DAW application (using software instruments) you can't use direct monitoring anymore, and the latency of the audio output becomes a decisive factor.
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Certif1ed
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Posted: July 22 2008 at 06:22 |
 MikeEnRegalia wrote:
^ thanks - I know about direct monitoring, but when you record drums or keyboards in your DAW application (using software instruments) you can't use direct monitoring anymore, and the latency of the audio output becomes a decisive factor. |
Really?
It's hard to believe that's a limitation of the GuitarPort (although not impossible, of course) - ultimately, it's just a remote soundcard. If you bypass the internal one, then you can direct monitor everything - at least, the TonePort does 
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MikeEnRegalia
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Posted: July 22 2008 at 07:03 |
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^ every soundcard has latency ... it's not a limitation of the GuitarPort as such. The direct monitoring is possible because the software processes the input internally ... it goes from the input through the usb cable to the Line6 driver software, there it gets processed and goes back through the USB cable to the output. The signal never needs to be transferred to another software running on the computer - that's why it's so fast. But if the audio is generated by another VST instrument it needs to be transferred to the Line6 driver, and that's where ASIO comes into play. It's not a big problem for tracks which you play back through the device ... the DAW software compensates for that and simply starts the tracks with a negative offset to compensate for the latency of the interface (and possibly for added latency by software instruments and effects). But obviously this isn't possible for monitoring situations ...
It's difficult to describe ... currently I got the output latency down to 10ms which is something you can work with, but I would fell more comfortable getting it down to 5ms.
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cobb2
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Posted: July 22 2008 at 07:44 |
Not exactly up on the GuitarPort, but isn't it just a virtual amp. Won't the Ableton software have trouble with it if it is not recognised as a vst plugin? I am talking from the prospective of Sonar here, where instrument interfaces must come with compatible driver interfaces. I use sonar connected to a Roland Soniccell for midi and any instrument or vocal I record by plugging from the mixer output to computer soundcard and use the soniccell for monitoring.
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MikeEnRegalia
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Posted: July 22 2008 at 08:02 |
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^ yes and no. You can use the GuitarPort as a virtual amp, but only with additional software (called "Gearbox Plug-In". But most people use it as a digital input. You simply plug your guitar into it, and via the Gearbox software (not the plug-in I mentioned above) both processed and unprocessed output of the guitar become available as ASIO inputs.
Of course using the plug-in is really cool because you can record the unprocessed input and always change the amp configuration later. But there are downsides ... you'll need a fast computer, ASIO becomes an issue, and often you need more expensive licenses for the DAW software because the cheaper versions usually only support a limited number of simultaneously running VST instruments and/or audio effects.
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cobb2
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Posted: July 22 2008 at 08:13 |
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Oh, I see, so it's just using the ASIO bus as a soundcard would. You shouldn't have any problems. Does the Apleton suite have a special utility tool for setting up soundcards as Sonar does?
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MikeEnRegalia
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Posted: July 22 2008 at 08:17 |
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^ it does have tools for setting up ASIO, and everything works fine. I just can't get latency below 10ms, that's the problem.
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cobb2
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Posted: July 22 2008 at 08:23 |
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What about dropping your recording sample rate back to 16 bit. You will have to do this anyway if you want to burn waves to CD because of the ISO standard for audio CD's
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MikeEnRegalia
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Posted: July 22 2008 at 09:49 |
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^ Maybe that would improve the latency, but I really want to record in 24bit/96khz. It's the standard for professional audio, and even if it gets downsampled in the end the high resolution is necessary during the production process. Most DAWs do their calculation in this resolution anyway, and using lower quality samples for recording only results in more down/upsampling calculations.
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rileydog22
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Posted: July 22 2008 at 16:21 |
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I avoid the latency issue altogether. I put a splitter into my output and send half of my signal to the computer and half to an amp.
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MikeEnRegalia
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Posted: July 23 2008 at 04:52 |
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^ the Line6 interface does that automatically ... that's not the point of this thread. The point is that direct monitoring is not possible when you want to record drums, keyboards etc. entirely within the DAW application, using a MIDI keyboard as input.
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cobb2
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Posted: July 23 2008 at 05:35 |
Did you fix your problem? What about your buffer size? Have you played with this? I think it works the smaller the buffer size the quicker the actual computer needs to be to process it in real time- so a larger buffer may work. (I think that is the right way around).
edit: read quicker computer as more powerful
Edited by cobb2 - July 23 2008 at 05:36
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MikeEnRegalia
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Posted: July 23 2008 at 05:55 |
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Of course a larger buffer works ... but larger buffer means higher latency, which is my problem.
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cobb2
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Posted: July 23 2008 at 06:10 |
Oh, okay- just bandying ideas around. From experience I know that it can very frustrating getting music recording software to play nice with audio hardware.
addition: have you emailed Albeton with the problem- if they see you as a potential customer, I'm sure they would be more than happy to help
Edited by cobb2 - July 23 2008 at 06:12
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MikeEnRegalia
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Posted: July 23 2008 at 06:24 |
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I'm still waiting for the KB37 to arrive ... if all else fails, maybe I'll write them an email, but they don't yet officially support Vista 64bit, so I guess that for the time being I'll have to find a solution myself.
One possible solution would be to decrease the buffer size while recording the midi instruments ... there would be slight errors in the playback, but it wouldn't affect the recorded midi information. Still, I'd like to have an environment where I can focus on playing/recording the music and not concern myself with buffer settings and hardware configurations all the time. After all, that's why people use DAWs in the first place ... because it's much more comfortable and efficient than using real studio hardware.
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cobb2
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Posted: July 23 2008 at 06:28 |
If you have a 64bit CPU you should be able to set your buffer to its lowest setting. Does Apleton support 64bit?
I think Vista has a different audio subsystem than XP- though this shouldn't affect you if you are using ASIO
Edited by cobb2 - July 23 2008 at 06:30
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MikeEnRegalia
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Posted: July 23 2008 at 06:44 |
^ I think I mentioned before that the lowest buffer setting doesn't work. Why else would I have created this thread? 
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cobb2
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Posted: July 23 2008 at 07:19 |
Yes, but I think that's the crux of your problem- it should on a 64bit machine. I take it you have at least 2GB of memory? Assuming Ableton is Vista ready and is a 64bit installation package, it sort of rules out the software and the computer hardware and can only point to the line 6 unit. Does that make sense, or am I jumping the gun somewhere?
addition: You have probably already seen this but it seems to have relevance. from the FAQ's section in Ableton
When do I need to adjust the "Overall Latency" setting in Live?
Suppose you are monitoring your voice or an instrument through an external device instead of Live (you have the "Monitor through Live" switch set to "no"). Now, as you sing or play along with the metronome or clips already in the Set, you are in fact playing along with "late" audio, because it takes some milliseconds for the audio to travel from the CPU to the audio outputs. This delay is referred to as "latency." Because you are hearing the audio from the program late, the audio you are recording into the program arrives late with respect to the program's internal timing. Live automatically compensates for this error by moving the recorded audio in song time so that you normally don't have to worry about any of this. If the recording you make sounds "untight" despite Live's effort to compensate for latency, then the audio hardware driver is probably the culprit: It is reporting an inaccurate latency value to Live. You can manually correct Live's latency assumption by adjusting the "Overall Latency" control in the Audio Preferences. The manual describes how to do this in the "Computer Audio Issues" chapter.
Edited by cobb2 - July 23 2008 at 07:50
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Recording music on the PC ...
