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by proportional multiple of the sample rate being changed If the buffer is unchanged.. Dimension features a powerful pairing of sample-playback and advanced synthesis technologies, capable of covering every aspect of contemporary music.
If you're referring to hardware latency when recording then the only reason I can think of is if you're using a SB Card and the default sampling rate for that card is 48 KHz, by setting the project also to 48 KHZ you would be saving the card built-in sampling rate conversion time which needs to happen for 44.. Say I record a guitar part to a click track from the computer metronome Then I record a second guitar part while listening to the 1st guitar part.
If I misunderstood the replies sorry for butting in ORIGINAL: boten Beagle you shouldn't be seeing this as sampling rate increase lowering the latency.
If you're referring to hardware latency when recording then the only reason I can think of is if you're using a SB Card and the default sampling rate for that card is 48 KHz, by setting the project also to 48 KHZ you would be saving the card built-in sampling rate conversion time which needs to happen for 44.. Does that sound right? I have the latency slider (WDM drivers) set anywhere between 10 to 30ms, Record one track at a time and never have a problem with track alignment on playback.. You can hear an example of Dimension Pro's Garritan Pocket Orchestra in the Dimension web player's 'Dimension Songs' category.. I'm not attempting to start anything, this is just my understanding and what my own research shows.. Whether you choose to do it straight up with in/out buffers or sample rates or combining both.. I haven't thought about it enough to figure out exactly why it works, but try it.. This will eliminate the perception of the delay It is at 10 now Is there any other thing I should be looking.. The reason for the effective latency definition you see in options -audio is that some cards set their latency based on numberof samples.. With the intuitive FocusBug controller it is just like using a tilt-shift or selective focus lens right inside of Adobe Photoshop.. in a shorter time as he is hauling the same amount of info(buffer) but recieving it more quickly.. I was talking about effective latency which differs from overal latency, the difference is that it is calculated based on the sampling rate.. The tracks will be aligned correctly Scook, I want to discuss this a bit Are you speaking only of monitoring latency while recording? I'm pretty sure with my setup that I have no monitoring latency.. A 10 ms delay would make the overall mix less than 'tight' Or is the latency delay only percieved by the monitors during recording? CPU load is monitored in cake walk and seems to be ok.. Preset Addicts With over 1,500 production-ready presets, you can access the perfect sound for almost any project.. In other words the delay (or the shift) will be the same because the buffer size is the same and the overall hardware latency which is not related to audio streaming but to the time it takes the audio device to physically drive A/D or D/A conversions is also constant for a given card (that should be lower than 2 msec to make things workable).. That's why you see the reduction of the effective latency when you change the sampling rate.. both recording and round trip This all being said Your picking your poison addressing any latency issues.. I'm not attempting to start anything, this is just my understanding and what my own research shows.. 1 effective latency multiplied by (44 1/88 2), because the program normalizes it.. both recording and round trip This all being said Your picking your poison addressing any latency issues.. What you're telling me, however, is that if I performed an audio loop back test (such as one described by zungle: ) for each of the scenarios above, that my latency would remain constant across all three? Regardless of what the 'effective latency' is reported as shown in the pix? I'll have to try it.. 1 KHz However, the system latency remains the same Hmmm well, I'll have to check it again when I get home, I haven't looked at that in months.. If your hardware doesn't support higher sample rates your entire system will have higher latency.. 7 with this setting(2 9ms in WDM) using 24/44 1 with sample rates over 48 00 you can get under 2ms and maintain reasonably good functionability.. That's what I meant above I'm a bit unclear? Am I reading that there is doubt about a sample rates affect on latency???? Higher sample rates definately reduce latency whether it be recording.. 1 effective latency multiplied by (44 1/88 2), because the program normalizes it.. (technical term) Is the timing on the track I am recording off by that much in refrance to the other tracks? I changed my sampling rate from 44100 to 192000.. With the new FocusBrush tool you can paint sharpness or blur right where you want it.. If these samples are more 'dense' because sampling rate has been increased, then the latency applied per sample will be reduced accordingly.. I will have to work on that with the sound card tech help to confirm I am doing that.. The fact that the sampling rate has been increased only increases the amount of information that needs to be processed by the system per unit of time, but the streaming buffers remain the same.. but its the unit of time that is changed with sample rate and buffer adjustments.. With Latency delay recording offset anything you wish to call it Your interface plays as much a part of that as the software app does.. With Latency delay recording offset anything you wish to call it Your interface plays as much a part of that as the software app does.. 2 7ms latency ETC The variations of these results shown by individual interfaces has to do with the way their audio drivers are written, what their DAC's are doing and whether or not DSP or virtual interfacing is being used.. If yours or others do NOT do this, I would be interested in knowing that and why mine does! Beagle, Your right.. latency will change with the sample rate change, whether its up or down Sample rate affects both recording latency and input monitoring as well as round trip system latency.. ORIGINAL: boten increasing the sampling rate will reduce your latency, Sorry Beagle I don't understand why 192 KHz improves latency on a system.. (technical term) Is the timing on the track I am recording off by that much in refrance to the other tracks? I changed my sampling rate from 44100 to 192000.. I am now able to set latency to 4 and when I use the 'input echo' button I get very little delay.. I want to confirm this is ok before I descide to leave it there Here are some system specs: (not sure what you info is applicable but here we go) Cakewalk Ver.. If you look at your effective latency you will see that for 88,8 the latency is 44.. The affect is latency reduction Heres 2 short articles that mention and support the affect of sample rates.. ASIO drivers get better hardware latency than WDM because they handle the data processing more efficiently than other drivers.. Both systems behave exactly the same on both softwares Here's what happens: They also behave the same way whether I'm using ASIO mode or WDM mode.. I'm always open to input The above statement isn't excactly true The buffers may stay the same.. 48kHz gives me about 1 5msec less latency than 44 1kHz and as you move up it gets lower.. If you have a small truck 5 yds and a large 5 yd bucket on your loader you will use more gas(CPU) because you will have to go and unload the truck so frequently.. I may have to redo or realign them But I think ASIO is the way to go I also plan to use my sound card mixer to monitor.. latency increase This is why many engineers that work with high sample rates will increase their buffers.. Maybe my rational for the extemely low latency is based on an incorrect assumption?I am just trying to get all my tracks lined up as 'tight' as possible.. Home Studio XL 4 01 Cake walk driver mode settings WDM/KX, No boxes checked in advanced tab.. Go into your M-AudioControl panel and find DMA settings I'm at work but I think its Control panel/advanced/hardware settings.. (I'm monitoring thru my mixer ) But I'm pretty sure I have 'recording latency' if that's a proper term.. latency increase This is why many engineers that work with high sample rates will increase their buffers.. FocalPoint 2 - create realistic selective focus, depth-of-field and vignette effects that tell your viewers right where to look.. I use the 'zero latency' monitoring feature on my interface, which is similar to what you are describing as monitoring with a mixer.. I am not sure how and increase I/O will effect me But I think using up disc space is not worth slightly lower latency if my final mix does not show it.. I am now able to set latency to 4 and when I use the 'input echo' button I get very little delay.. One does not exist without the other Change 1 you must change the other or it doesn't work.. Here is another that shows the math As far as hardware goes yes different interfaces have different results when changing the sample rate.. One thing I have notice is, now I can notice the delay in the tracks that I have been overlaying.. 1 KHz However, the system latency remains the same ORIGINAL: boten Not sure what you mean by 'latency is decreased'.. If I snap my fingers while recording i can hear the snap in the monitors a hair later.. Is the latency heard when recording the same when I play it back with the rest of the tracks? In otherwords.. If I misunderstood the replies sorry for butting in ORIGINAL: boten Beagle you shouldn't be seeing this as sampling rate increase lowering the latency.. 1 KHz However, the system latency remains the same Hmmm well, I'll have to check it again when I get home, I haven't looked at that in months.. The fact that the sampling rate has been increased only increases the amount of information that needs to be processed by the system per unit of time, but the streaming buffers remain the same.. The higher sample rate is more info and allows the buffers or forces the buffer to fill up more quickly.. But the system latency will remain unchanged, meaning your system is spending the same amount of time to perform commands only that when you increase your sampling rate it will have less time per sample to the process.. I'm always open to input The above statement isn't excactly true The buffers may stay the same.. raising the sample rate will lower latency But your latencies are high for your sample rates.. Say I record a guitar part to a click track from the computer metronome Then I record a second guitar part while listening to the 1st guitar part.. The delta change isn't the same, but that's expected I also tried changing the buffers in my M-audio panel for ASIO mode and again, the base number was different depending on the size of the buffer, but each time the sampling rate was changed, the latency responded inversely with the sampling rate.. It is at 10 now Is there any other thing I should be looking If I snap my fingers while recording i can hear the snap in the monitors a hair later.. 2 7ms latency ETC The variations of these results shown by individual interfaces has to do with the way their audio drivers are written, what their DAC's are doing and whether or not DSP or virtual interfacing is being used.. ORIGINAL: scook If you are not monitoring the signal you are recording through the DAW, latency should not be an issue.. Go into your M-AudioControl panel and find DMA settings I'm at work but I think its Control panel/advanced/hardware settings.. That's why you see the reduction of the effective latency when you change the sampling rate.. His experience and mastery have yielded an exciting array of synth basses, bold leads, thick stacks, pads, simulations and amazing effects.. input monitoring or round trip AD/DAW/DA The only way that sample rate does not affect latency is if the input buffer is also changed.. So to use less gas you would increase your truck size(buffer) to save on gas(CPU) I understand the math to be like this.. Electronica Collection Nico Herz of BigTone helped to create Dimension Pro's breathtaking electronica palette.. If you increase the sampling rate, your latency will decrease I have done so in SHS4.. The delta change isn't the same, but that's expected I also tried changing the buffers in my M-audio panel for ASIO mode and again, the base number was different depending on the size of the buffer, but each time the sampling rate was changed, the latency responded inversely with the sampling rate.. Not sure what you mean by 'latency is decreased' Typically latency is set by user either by settings the buffers size (samples) or directly through the effective latency slide in WDM mode.. The fact that the sampling rate has been increased only increases the amount of information that needs to be processed by the system per unit of time, but the streaming buffers remain the same.. I'm not using an SB card, but I was at one time, maybe that was my confusion Switching to ASIO seems to have fixed latency issues.. Buffer 128 divided by sample rate 44 1 2 9ms latency Buffer 128 divided by sample rate 48.. It was suggested to me to reduce my latency to 4 msec This will eliminate the perception of the delay.. 1 effective latency multiplied by (44 1/88 2), because the program normalizes it.. Using the large loader(higher sample rate) the truck may then drive up the road and unload and fill up again.. I gave a link in my earlier reply that shows the math and explains how and why the latency is decreased with a higher sample rate.. If you look at your effective latency you will see that for 88,8 the latency is 44.. by proportional multiple of the sample rate being changed If the buffer is unchanged.. ASIO drivers get better hardware latency than WDM because they handle the data processing more efficiently than other drivers.. in a shorter time as he is hauling the same amount of info(buffer) but recieving it more quickly.. as this lowers the CPU hit that can be caused by moving so much DATA so quickly You could Equate the CPU to gas.. Both systems behave exactly the same on both softwares Here's what happens: They also behave the same way whether I'm using ASIO mode or WDM mode.. Dimension Pro 1 5 VST DXi HYBRiD X86 Want to learn about the great things you can do with your new Surface Pro? Read these articles from Surface Support.. The Garritan Pocket Orchestra, included in Dimension Pro, contains samples of all the major instruments in a symphony orchestra-strings, brass, woodwinds, and percussion.. In other words the delay (or the shift) will be the same because the buffer size is the same and the overall hardware latency which is not related to audio streaming but to the time it takes the audio device to physically drive A/D or D/A conversions is also constant for a given card (that should be lower than 2 msec to make things workable).. Typically latency is set by user either by settings the buffers size (samples) or directly through the effective latency slide in WDM mode.. some are better than others but within each interface the math is generally consistent with all rates and buffer ratios within that device.
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