exyll
Junior Member
Posts: 74
|
Post by exyll on Dec 14, 2009 22:48:19 GMT
Why is this important for you? Having low latency audio for playing video is not really an issue.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Dec 14, 2009 22:47:13 GMT
Which gives better sound quality ? Does someone know? You want to have the highest quality ASAP. That means that if you want to output in 48kHz that it is best to have the generator doing it for you or let it do a high quality conversion before passing to the audio out channels.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Dec 14, 2009 22:42:35 GMT
I have similar issues with live on my X-FI gamer but not on channel 1/2 (Left/Right)
I have my speaker setup configured for 5.1 1/2 Front Left Right 3/4 Front Center 5/6 Rear Left Right 7/8 Center Left Right
Channel 7/8 give me 'bleeding' channels as audio is now played over my front and rear speakers.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Dec 11, 2009 1:42:38 GMT
Hi all. My system: Vista Home, Realtek High Def Audio, Asio4ALL 2.9, guitar on line in plug, no windows system sounds. No guitar signal on A4A. Works fine on DirectX drivers. Ideas? Works fine on my Realtek HD Audio. You don't really give that much information to help you out.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Dec 9, 2009 18:17:38 GMT
I have *excellent* results with the latest beta. The USB latency is GONE and it is stable too as with 2.9 I built up more latency over time with my USB microphone.
I usually have the latency set to around 128 samples for my heaviest setups during live recording.
I really don't notice any delay during live monitoring my USB microphone and my realtek line-in guitar recordings.
Awesome job!
Hope that my configuration details help people optimize their system to get lower latency.
I have the following hardware setup: - AMD Phenom II X3 @ 3.25GHz - 4GB RAM - Samson CO3U USB microphone
Following software setup - Windows 7 x64 (system rating of 6.8) - Realtek HD Audio drivers latest version - Ableton Live 8.0.8 (other DAW's work fine too) - ASIO4ALL 2.10 beta 1
Following system configuration - Disabled screensaver - Disabled indexing/search services - Disabled Processor Powermanagement - Set processor scheduling to: Background services - Disabled (temporarily) the virus scanner - Disabled all software that I do not need during my audio sessions (MSN, Skype, email notifiers, twitter clients, etc.) - Set all audio devices to 48kHz 16 bit exclusive mode on the "advanced" tab of the audio ports in Windows "Playback" and "Recording" devices. Following DAW configuration: - Using ASIO and selected ASIO4ALL driver - Set DAW audio quality to 16bit / 48kHz as that is the usb microphone native sampling frequency
Following ASIO4ALL configuration: - Selected advanced view - Enabled only the ports I needed (realtek line-in, samson usb in, realtek audio out) - Unchecked "Always resample 44.1kHz<->48kHz" - Checked "Hardware Buffer" for my Samson USB mic - Checked "Allow Pull Mode (WaveRT)" on Realtek HD Audio
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Dec 9, 2009 17:54:21 GMT
Here it is working ok too on Windows 7 x64 with both a creative xtreme gamer (PCI) and Realtek HD Audio (onboard).
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Dec 9, 2009 17:51:33 GMT
Which versioning scheme you use is not that interesting.
Linux used to have even/odd minor numbering indicating stable or development releases. Some had postfixes like alpha, beta, gamma, delta.
At my current job we just use major.minor.revision. Added functionality means an increased minor and working to make that version stable will increase the revision for every release build that we publish/make available.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Jun 20, 2009 12:12:04 GMT
I think that you are mixing up some stuff Input and output is at 24 bit fixed point maximum. However, some DAW's have an internal engine that can be configured like in your situation where you can set it to 32 bit floating point. In the end when all channels have been mixed in 32 bit floating point modus it will need to "render" it live to the output specifications of your hardware. The reason that it is doing 32 bit floating point and not 24 bit floating point it that that's the way CPU's work. Cpu's do there work in 8, 16, 32, 64 or 128 bit operations. The reason its working in floating point instead of fixed is because of the higher precision while mixing audio channel data. In most situations it is sufficient to work in 44.1 or 48 kHz IN/OUT and if your hardware supports 24 bit then I suggest you use that too. I work in 48kHz / 24 bit when possible as my input gear its native sampling frequency if 48kHz. The human ear can hear at 22kHz max, you need twice that frequency for recoring such frequencies so then you get 44kHz. To be at the safe-side you could record at 48kHz. The bit depth will decide the quality of your recording. It is better to record at 48kHz/24b then 96kHz/16b as the human ear wont hear such frequencies except when you are recording stuff for your dogs ;-)
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Jun 20, 2009 12:01:04 GMT
BUT everytime I run Ableton Live, there are sudden cliks/pops/drops until I go to the ASIO4ALL settings and move on the buffer from 128 to somwhere and back to 128. Than it works ok.... I also use this 'reset' procedure often. I have a creative xtreme gamer and onboard realtek audio. I cannot use the onboard audio for lenghty sessions with ultralow latencies (like 64 samples), I really need to set it to 128-256 samples. The xtreme gamer is working better I usually have it set to below 96 samples but it also depends on the VST's loaded. Resource hungry VST's will result in clicks and pops that can be resolved by increasing the sample buffer. When I'm "mastering" then I often have it set to 256-512 samples because of all HQ settings that I have enabled and I just want to listenin HQ. In this mode I do not want to hear clicks and pops because I do not need the low output latencies as I'm not "monitoring" any inputs or playing MIDI instruments.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Jun 20, 2009 11:53:10 GMT
When you select ASIO4ALL does your audio hardware show up in the ASIO4ALL panel? And what status does your output have? If it has a red cross then quit ALL your applications except FLSTUDIO. Also browsers and such. Then reselect ASIO4ALL in FLSTUDIO and see if your output is still locked because another application is using it.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Jun 20, 2009 11:47:26 GMT
There is no "offline" control panel anymore with the latest version of asio4all. Just start the audio application that supports ASIO. Then select ASIO4ALL as the ASIO driver and then a icon tray icon will appear that lets you tweek the ASIO4ALL settings.
ASIO4ALL settings are stored for each user and each application.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Jun 20, 2009 11:44:19 GMT
Asio4all working fine at 64 samples here on Windows 7 RC (build 7100).
Only thing is that a small delay is build up requiring to "reset" one of the sliders and then it works fine again. But I don't know if that is related to the Windows 7 drivers of creative, the Windows 7 stack or Ableton.
Please note that you still need Windows 7 audio drivers. Asio4all is NOT a replacement as it stacks up to the audio drivers for your hardware.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Jun 15, 2009 9:37:54 GMT
Hi all,
I'm using ASIO4ALL to be able to monitor my audio input while recording.
PC: AMD Phenom II X3 / 4GB / Windows 7 RC x64 IN: Samson C03U USB microphone OUT: Creative X-FI Gamer Sequencer: Ableton Live 8 Audio frequency: 48kHz (native sampling frequency of the mic)
I enable hardware buffering for both devices and set the samples for both to 64.
I then select live monitoring in Ableton Live (from AUTO to IN) and I have a very low latency that very suited for live monitoring but as time goes by latency is increasing until it isn't suited for monitoring anymore.
Other people also experiencing similar behavior and any suggestions in fixing this?
When I click on the slider in the ASIO4ALL dialog then the latency is gone and slowly builds up again.
I disabled any effect VST's that could increase the delay/latency.
|
|
exyll
Junior Member
Posts: 74
|
Post by exyll on Jun 4, 2009 8:56:02 GMT
Therefore: No multi-client with ASIO4ALL - unless your audio device supports this in hardware. I have a Creative X-FI Gamer and using Windows 7 x64 RC and using the Creative ASIO drivers let me have low latency audio in my DAW and still have sounds in the background so my guess is that my hardware supports it. My guess here is that with "multi-client" you mean multiple audio applications that use the ASIO4ALL driver? As I don't have background audio when I select ASIO4ALL. It locks the hardware exclusively. But as far as I can remember is that I didn't have this with XP. Is this different between 2000, XP, Vista or 7?
|
|