Latency in Playback

I am using Logic Pro 9 with 8 GB ram..
All of a sudden i have huge latency when u playback. The playhead is moving fine. Also this latency is occuring at different sections of the track not at the same place. So i am pretty sure its nothing to do with track itself.
I have checked all the forums available but none seem to help me get over this problem. I have changed all possible options for recording delay and pluginin delay compensation, buffer size and what not..
Please help. Almost finished the track to send to label and now this happens .
Kill me
Version: 9.1.8 (1700.67)
Model: MacBookPro8,1, BootROM MBP81.0047.B27, 2 processors, Intel Core i7, 2.8 GHz, 8 GB, SMC 1.68f99
Graphics: Intel HD Graphics 3000, Intel HD Graphics 3000, Built-In, 512 MB
Memory Module: BANK 0/DIMM0, 4 GB, DDR3, 1333 MHz, 0x80CE, 0x4D34373142353237334348302D4348392020
Memory Module: BANK 1/DIMM0, 4 GB, DDR3, 1333 MHz, 0x80CE, 0x4D34373142353237334348302D4348392020
AirPort: spairport_wireless_card_type_airport_extreme (0x14E4, 0xD6), Broadcom BCM43xx 1.0 (5.106.198.19.22)
Bluetooth: Version 4.0.8f17, 2 service, 18 devices, 1 incoming serial ports
Network Service: Wi-Fi, AirPort, en1
Serial ATA Device: TOSHIBA MK7559GSXF, 750.16 GB
Serial ATA Device: MATSHITADVD-R UJ-8A8
USB Device: FaceTime HD Camera (Built-in), apple_vendor_id, 0x8509, 0xfa200000 / 3
USB Device: hub_device, 0x0424 (SMSC), 0x2513, 0xfa100000 / 2
USB Device: Fast Track, 0x0763 (M-Audio), 0x2024, 0xfa130000 / 6
USB Device: BRCM2070 Hub, 0x0a5c (Broadcom Corp.), 0x4500, 0xfa110000 / 5
USB Device: Bluetooth USB Host Controller, apple_vendor_id, 0x821a, 0xfa113000 / 8
USB Device: Apple Internal Keyboard / Trackpad, apple_vendor_id, 0x0253, 0xfa120000 / 4
USB Device: hub_device, 0x0424 (SMSC), 0x2513, 0xfd100000 / 2
USB Device: IR Receiver, apple_vendor_id, 0x8242, 0xfd110000 / 3
FireWire Device: ATA Device 00, Seagate, 400mbit_speed

Thanks for the reply well it seems like this problem is specific to the project i am working on. Going to try freeze and see if that helps.
is having 70 channels a lot? Maybe because of that?
The project is saved on an external Firewire drive.
FireWire Device: ATA Device 00, Seagate, 400mbit_speed
By huge latency on PB i meant:
When i hit play the playhead is moving fine but the audio is skipping at places. Could this be because of my audio files in arrange maybe. I have about 15 percussion sounds and thats where the problem seems to be coming from while playback the percussion elements just stop and other keep playing and also on a few MIDI channel.
I accidently pressed "Insert Silence between Playhead" could this be by any chance the reason for this? I dont know how i can find out where is the silence.

Similar Messages

Problem with tracking relative to latency during playback

Hi all,
I'm pretty new to Logic but have been using software sequencers for a couple years now (Pro-tools/Reason/Ableton Live) so I'm wondering what I'm forgetting/doing incorrectly when I track any live sound and hear that it's offset from the global tempo when I play back.
In other words I'm pretty sure I have the basics set up correctly but I'm still having the latency issues with tracking live audio; the low latency mode is on, HD buffer size is at 256, I've installed the latest firmware for my interface (motu 8 pre-fire wire). Maybe the follow tempo button in inspector being checked would help... just read about that but I'm not at my computer to try.
Any trouble shooting suggestions would be greatly appreciated.
Thanks!

A Buffer I/O setting of 256 is a pretty high setting for tracking. That's giving you 5.8 ms of latency at 44.1K, and 5.3 ms latency at 48K. That's enough to throw off otherwise tight timing.
That's the first thing.
Second thing is that you should probably check your recording delay setting. If it's set to zero then you need to do a loopback test to assess the recording latency and then plug in the appropriate value so that your recorded tracks play back where they should.
Go to this link [http://discussions.apple.com/thread.jspa?messageID=5412800&#5412800] and look for my long post towards the bottom. The procedure for setting your recording delay is outlined in detail. Don't be put off by the number of steps --- it's a procedure you only need to do once.
And make sure that when you do the test that you start with a fresh Logic song/project with NO plugins anywhere, and have PDC switched off.
Still, I'd recommend setting your buffer size to 64 or better, 32 (if you can get away with it).

Latency Issue When Multi-tracking For Webstream Audio

We are using Audition on a iMac 24". The input to the iMac is from a Yamaha M7CL via a Dante network, then through Audition, then sent via a Focusrite Scarlett 6i6 to Newtek Tricaster 8000. There is noticeable latency. The audio is not syncing with the video. Our video output is in real time, but the audio output from Audition is not. What could be the problem?

Sounds like we're on to something...
Do you have the Sound Output on your Mac set to the
Ozone and have the monitors connected to it as well?
The only way to get zero-latency in the headphones
during recording is to use the headphone jack on the
Ozone. Problem is though, USB audio devices are worse
than FireWire when it comes to latency.
yeah the sound output is set to the ozone and the speakers are connected to the ozone. there isn't any latency when recording, but when i play it back there is a delay. are you saying that i should connect the headphones into the ozone instead of the speakers? will this prevent latency during playback? please let me know and thanks for your help, thus far.

Rode podcaster and runtime error.

hello, i hope someone can help me with this problem. i purchased adobe auditon 3.0 because i do podcasting and i needed a program that would support 2 usb microphones. well i have no problem getting one of my microphones to record (an MXL usb .006) but when i plug in my rode podcaster and try to use it (and this is with the MXL unplugged, so i havent even tried to use multiple usb mics yet) it gives me a error that reads "microsoft visual C++ runtime library" as the header, then it reads "runtime error! program:...rogram files (x86)\adobe\adobe audition3.0\audition.exe
this application has requested the runtime to terminate it in an unusual way. please contact the application 's support team for more information."
the laptop is running windows vista, the 64 bit version.
so can anyone give me an idea what to do next?

>and i have no clue what you mean when you say use the wdm usb driver rather then the asio ones, so any advice there would be appreciated.
Well, this is the simplest explanation I can manage that explains the important principles:
WDM drivers are the ones that Microsoft intend to work with the OS (Windows Driver Model), and they allow for the use of multiple spund devices - wherever and however they are attached. But, these drivers have inevitably got a higher latency (delay introduced in processing signals), because they sit nearer to the top of the processing heirarchy in the OS. ASIO gets around the latency problem by short-cutting most of the OS, and allows software using its engine to talk more or less directly to your sound device - singular. To use any more than a singular device would require the arbitration that the OS would have to provide, thus destroying the point of doing this in the first place.
What ASIO4ALL does is to take the ASIO-based stream from Audition, or any other software using the same engine, and convert that back to a WDM-based internal protocol. It is bi-directional, so signals coming in (like from your USB mics) can be converted from their WDM format into ASIO for Audition, as well. As a consequence, ASIO4ALL can redirect a multi-channel stream to different devices, and can also take multiple input streams from them as well, combining them into a concurrent ASIO stream. There are several penalties to be paid though - the increased latency is one, and also you could
potentially run into sync problems, but not with the application you have in mind, I think.
There is one important bit in the ASIO4ALL FAQ that you should note, but a comment on it is needed:
>After some initial troubles with earlier revisions, the driver now appears to work just fine with numerous USB devices. one thing to not do, I mean: not ever! is to check the DMA box with USB devices. If you do not need all the additional software gizmos like EAX, EQ, Fake3D-Whatever... it is recommended that you _not_ install the drivers shipped with your USB audio interface. Windows has a built-in generic USB audio driver that works with most devices. Thus, if you want to have honest, no-nonsense, low-latency audio playback/recording with your USB audio interface - by all means: stick with the Windows driver! This is especially true for consumer grade stuff, where you can be assured that the drivers shipped with it are most likely 'consumer grade' as well.
One of the problems with USB mics is that you may well need to install at least the control applet for it, otherwise it
may not be usable sensibly at all. But, USB mics vary - so what you would need to do in any given situation, I can't comment on. In the past though, it's been pretty apparent that without the control applet, at least some of them have no sensible gain control or routing. But I would emphasise that this is very much a YMMV situation.
>i freely admit i need to do more research and reading on the software, but i have been told by people using multiple usb mics that this is what they used, or at least earlier versions of adobe audition.
That would be true. The earlier version referred to would probably be 1.5, which was the last version to use the built-in WDM system for recording, although all earlier versions could do this too, at least in principle. I believe, though, that there were other difficulties recording from USB sources with earlier versions, so all bets are off unless you've actually tried what you want to do with a specific hardware/software combination and actually
know it works.

Asio 4 all FAQ for USB card IMPORTANT INFO

well since i see look like in this forum the admin and a technicaly support service TOO BUSSY TO reply BECUZ look like to BUSSY TO PLAYING GAMES IN HIS COMPUTER AND TOO BUSY TO PLAYING GAMES HIS CUSTOMER WHO LOOK LIKE BEING TRAP BY HIS MARKETING GAYS for buying his product X-FI hehehe...:angry:
i copy paste it from asio4all
Q: Any hints on using ASIO4ALL with USB devices?

A: After some initial troubles with earlier revisions, the driver now appears to work just fine with numerous USB devices. one thing to not do, I mean: not ever! is to check the DMA box with USB devices. If you do not need all the additional software gizmos like EAX, EQ, Fake3D-Whatever... it is recommended that you _not_ install the drivers shipped with your USB audio interface. Windows has a built-in generic USB audio driver that works with most devices. Thus, if you want to have honest, no-nonsense, low-latency audio playback/recording with your USB audio interface - by all means: stick with the Windows driver! This is especially true for consumer grade stuff, where you can be assured that the drivers shipped with it are most likely 'consumer grade' as well.
well fact is since i only see asio4all maker is genius guy ( he is the man) for driver
( i still not know asio steinberg support my card or not with their new asio tipe 2)
i will explain this
 : there still a bug need to fix on asio 4 all for usb card x-fi cuz IM NOT SURE is da card lock in 24 bit 48 khz and some missing input output, or is becuz asio 4 all still need to fix da problem
2: i put driver install from CD and all kinda things and just remove un needed things and i check up again IN fruity loops so i see 92khz 32bit int + input output 24 bit 96khz boths
3: problem is since this alien soundcard worx unknown
starting with mixer at 48khz with toons of vst plug goin well
starting with mixer at 96khz with toons of vst plug goin blow up CPU etc
rewire geting worst
this is an example for music production i make
http://rapidshare.com/files/27069609/club_fusion__original_cheapsoundcard_mix_by_tripto nez_.mp3.html
the st one without rewire
http://rapidshare.com/files/272273409/club_fusion__revision_sound_mix_by_triptonez_.mp3. html
the 2nd this is after mix up rewire and cut out the vst from FL becuz cpu might blow like hell
(JUST LIKE HERBIE HANCOCK SAYS music come st , technology come after thats)
IF U GOT problem with musician to making music with just a gaming soundcard well thats not BIG problem for me thats is your problem your mistakes being not honest




The headset has an independent C-Media audio chip that accepts USB on one side and has it's own AC97/DAC. Because are 2 devices (one USB controller and one AC97 sound chip) apears in 2 places in Windows devices! For more info look here:
http://www.cmedia.com.tw/?q=en/usb_audio
CM06-F+/ L+ is a highly integrated single chip USB audio solution. All essential analog modules are embedded in CM06F+/L+, including 8CH DAC and earphone buffer, 2CH ADC, microphone booster, PLL, regulator, and USB transcei'ver.
I would correct the affirmation made by somebody that USB audio devices are no soundcards - YES there are soundcards with an USB controller instead of the usual PCI one! The digital data flows on USB instead of PCI to the audio controller that send it to the integrated DAC's and to the headsets.
So trying to use the USB headset via the X-Fi is like you are trying to use the on-board soundcard via the X-Fi...
Message Edited by SoNic2367 on 06-25-2006 2:2 PM

Problem with playback distortion. Possible latency buffer issue?

I have noticed a few other posts in the same vain as this but with no clear answers so here is my take on the problem:
I am experiencing a distorted slowed down playback through logic pro 8 only occasionally which can only be fixed wit a full system restart. It sounds very much like a latency/buffer issue but to be honest I am not too great at fixing this myself. I have tried to change the buffer size to no avail. Recording seems to be fine mostly but it is just on playback. Although today i did experience the same sound issues when recording.
I am using a brand new macbook 2.4 ghz with a behringer xenix 1622 mixing desk and a usb audio device. I have only just switched to a mac and have recorded before on my pc with the same desk and interface for over 2 years without a problem, so I am positive it is not the external hardware that is at fault.
I am getting exactly the same problem when I use Garageband.
Is it just the apple products which do this?
Is this a common problem with an easy fix?
I just dont know to be honest with you, i dont pretend to be an expert in the field, it is infuriating.
Is there a problem with my machine? If so I can change it no problem.
Please if anyone knows how I can sort this please let me know.
I am gutted because I bought a mac thinking they were designed for music production, i dont want a pc again!
Thanks.

Can you be more specific on your usb device; Make & model... USB2? Visit the manufacturer's website, check compatibility & make sure you've got the latest driver. If you haven't already, run "Software Update" (under the Apple logo in the top left of your screen) & download the updates... this will take a while, so go & get yourself a coffee! Make sure you turn Airport off when you want to use Logic.
If you're still having prob's check out this link;
http://docs.info.apple.com/article.html?artnum=304970
It's written for Logic 7, but most of it's in the same place.
Hope this helps.
Cheers...

Midi latency problems during record/playback

I am starting to use Garageband for midi recording--I am going through an M-Audio Audiophile interface by firewire--when I preview the software instruments everything is fine but when I record then the recorded notes are dropping out or delayed during record and playback--how can I solve thhis problem?
I have already followed the Garageband help suggestions for settings
I would welcome any help!

Do they have the same interface as you? I get very little latency on my iBook G4 and I am using usb 2.
Before getting my external drive I had been recording successfully on my iBook's hardrive. Besides eating up hardrive space I had no problems. Much of what I do uses GB's excellent software instruments and MIDI.
Software instruments are strenuous on the cpu and ram. What is your ram situation?
I also lock tracks as I go along, especially with software instruments. Avoid using effects during the recording stage, add the effects afterwards.
Kurt

DPC Latency Spikes + MSI P55-GD55 + i5-750 + Clipping in playback & record

I'm getting spikes in DPC Latency causing clipping sounds for playback and record using the on-board Realtek. I have all the latest drivers/bios from nvidia, msi, etc.
System:
Windows 7 x64
MSI P55-GD65 @ BIOS 1.3 (Turned off EIST in BIOS, helped but spikes remain)
Intel i5-750
OCZ DDR3 4GB @ 7-7-7-21 1.65v
Nvidia GT220
Have other users mention anything about latency issues with this model? Maybe a BIOS update like this thread can fix it?
https://forum-en.msi.com/index.php?topic=129152.0
Please, I hope it's a BIOS issue cause I'm lost as to what the problem could be. I tried disable everything (sound/network) on the board and still latency spikes.

So MSI support gave me a new beta bios to test out v1.4B5, and the spikes are less frequent but still happens.
I can try disabling some services but not sure where to even start. It does look like a BIOS update might be the actual solution so I'm hoping support can eventually help me out.

TS4094 Have a problem with drums latency when i rec it al seems good but when i playback th drums have a big latency issue please help

Drums latency help

Sounds like the Graphics chip has a problem, which I think is a Mainboard replacement.
You can get a quote for the repair at an Authorized Service Provider. There is an ASP list on the Toshiba website.

I was trying to adjust the latency issues i was having. something happened and i hear a very loud squeal. deathening.now when i listen to playbach thru my headphones i hear crackling sounds. if i were to bounce and playback thru i-tunes, i don't hear it.

if i record w/out monitoring and playback on i-tunes. sounds great. monitoring recording thru headphones gets a crackling sound that has a bit of delay or reverb. i use tascam us-122mkII interface. help!

if i record w/out monitoring and playback on i-tunes. sounds great. monitoring recording thru headphones gets a crackling sound that has a bit of delay or reverb. i use tascam us-122mkII interface. help!

Audio and video playback latency ,since windows 10 help please

hi everyone I'm having a problem with my laptop ,video playback is lagging and sound to , sometimes making robotic sounds, and it all started when I upgraded to windows 10.my laptop: Lenovo ThinkPad x220 intel core i7 2.7 GHz processor 8 GB of ram previously had windows 7 pro now windows 10 proI tried resetting my computer , you know to have it clean and fresh, I thought this might help , maybe some old programs from windows 7 were causing this problem , but it didn't help , I also tried disabling and enabling the drivers from the device manager , is a trick I found on youtube , but it didn't work , I'd be very thankful if you could give me some advices .

Can you try disabling the Ethernet adapter in Device Manager?

Audio playback and recording latency

When I quit video/music or utilize recording features like the voice memo, I'm having to wait 10 seconds while holding a button or starting the feature associated. This also includes apps being updated to ios 5. Any way to speed this problem up?

A slight departure from previous behavior is that Audition no longer automatically changes the sample rate of your device. There were too many problems when switching between different applications (e.g. between Premiere Pro and Audition, or Audition and Windows Media Player, etc) and the clock setting changing on the device, and then the device "freaking out" (to put it politely).
So, if there's software that came with Echo where you can change the clock settings, you might want to set it to 96 kHz there first, and then Audition CS5.5 should just go along with 96 kHz.

A quick primer on audio drivers, devices, and latency

This information has come from Durin, Adobe staffer:
Hi everyone,
A common question that comes up in these forums over and over has to do with recording latency, audio drivers, and device formats. I'm going to provide a brief overview of the different types of devices, how they interface with the computer and Audition, and steps to maximize performance and minimize the latency inherent in computer audio.
First, a few definitions:
Monitoring: listening to existing audio while simultaneously recording new audio.
Sample: The value of each individual bit of audio digitized by the audio device. Typically, the audio device measures the incoming signal 44,100 or 48,000 times every second.
Buffer Size: The "bucket" where samples are placed before being passed to the destination. An audio application will collect a buffers-worth of samples before feeding it to the audio device for playback. An audio device will collect a buffers-worth of samples before feeding it to the audio device when recording. Buffers are typically measured in Samples (command values being 64, 128, 512, 1024, 2048...) or milliseconds which is simply a calculation based on the device sample rate and buffer size.
Latency: The time span that occurs between providing an input signal into an audio device (through a microphone, keyboard, guitar input, etc) and when each buffers-worth of that signal is provided to the audio application. It also refers to the other direction, where the output audio signal is sent from the audio application to the audio device for playback. When recording while monitoring, the overall perceived latency can often be double the device buffer size.
ASIO, MME, CoreAudio: These are audio driver models, which simply specify the manner in which an audio application and audio device communicate. Apple Mac systems use CoreAudio almost exclusively which provides for low buffer sizes and the ability to mix and match different devices (called an Aggregate Device.) MME and ASIO are mostly Windows-exclusive driver models, and provide different methods of communicating between application and device. MME drivers allow the operating system itself to act as a go-between and are generally slower as they rely upon higher buffer sizes and have to pass through multiple processes on the computer before being sent to the audio device. ASIO drivers provide an audio application direct communication with the hardware, bypassing the operating system. This allows for much lower latency while being limited in an applications ability to access multiple devices simultaneously, or share a device channel with another application.
Dropouts: Missing audio data as a result of being unable to process an audio stream fast enough to keep up with the buffer size. Generally, dropouts occur when an audio application cannot process effects and mix tracks together quickly enough to fill the device buffer, or when the audio device is trying to send audio data to the application more quickly than it can handle it. (Remember when Lucy and Ethel were working at the chocolate factory and the machine sped up to the point where they were dropping chocolates all over the place? Pretend the chocolates were samples, Lucy and Ethel were the audio application, and the chocolate machine is the audio device/driver, and you'll have a pretty good visualization of how this works.)
Typically, latency is not a problem if you're simply playing back existing audio (you might experience a very slight delay between pressing PLAY and when audio is heard through your speakers) or recording to disk without monitoring existing audio tracks since precise timing is not crucial in these conditions. However, when trying to play along with a drum track, or sing a harmony to an existing track, or overdub narration to a video, latency becomes a factor since our ears are far more sensitive to timing issues than our other senses.   If a bass guitar track is not precisely aligned with the drums, it quickly sounds sloppy. Therefore, we need to attempt to reduce latency as much as possible for these situations. If we simply set our Buffer Size parameter as low as it will go, we're likely to experience dropouts - especially if we have some tracks configured with audio effects which require additional processing and contribute their own latency to the chain. Dropouts are annoying but not destructive during playback, but if dropouts occur on the recording stream, it means you're losing data and your recording will never sound right - the data is simply lost.   Obviously, this is not good.
Latency under 40ms is generally considered within the range of reasonable for recording. Some folks can hear even this and it affects their ability to play, but most people find this unnoticeable or tolerable. We can calculate our approximate desired buffer size with this formula:
(Sample per second / 1000) * Desired Latency
So, if we are recording at 44,100 Hz and we are aiming for 20ms latency:   44100 / 1000 * 20 = 882 samples. Most audio devices do not allow arbitrary buffer sizes but offer an array of choices, so we would select the closest option. The device I'm using right now offers 512 and 1024 samples as the closest available buffer sizes, so I would select 512 first and see how this performs. If my session has a lot of tracks and/or several effects, I might need to bump this up to 1024 if I experience dropouts.
Now that we hopefully have a pretty firm understanding of what constitutes latency and under what circumstances it is undesirable, let's take a look at how we can reduce it for our needs. You may find that you continue to experience dropouts at a buffer size of 1024 but that raising it to larger options introduces too much latency for your needs. So we need to determine what we can do to reduce our overhead in order to have quality playback and recording at this buffer size.
Effects: A common cause of playback latency is the use of effects. As your audio stream passes through an effect, it takes time for the computer to perform the calculations to modify that signal. Each effect in a chain introduces its own amount of latency before the chunk of audio even reaches the point where the audio application passes it to the audio device and starts to fill up the buffer. Audition and other DAWs attempt to address this through "latency compensation" routines which introduce a bit more latency when you first press play as they process several seconds of audio ahead of time before beginning to stream those chunks to the audio driver. In some cases, however, the effects may be so intensive that the CPU simply isn't processing the math fast enough.   With Audition, you can "freeze" or pre-render these tracks by clicking the small lightning bolt button visible in the Effects Rack with that track selected. This performs a background render of that track, which automatically updates if you make any changes to the track or effect parameters, so that instead of calculating all those changes on-the-fly, it simply needs to stream back a plain old audio file which requires much fewer system resources. You may also choose to disable certain effects, or temporarily replace them with alternatives which may not sound exactly like what you want for your final mix, but which adequately simulate the desired effect for the purpose of recording.   (You might replace the CPU-intensive Full Reverb effect with the lightweight Studio Reverb effect, for example. Full Reverb effect is mathematically far more accurate and realistic, but Studio Reverb can provide that quick "body" you might want when monitoring vocals, for example.) You can also just disable the effects for a track or clip while recording, and turn them on later.
Device and Driver Options: Different devices may have wildly different performance at the same buffer size and with the same session. Audio devices designed primarily for gaming are less likely to perform well at low buffer sizes as those designed for music production, for example. Even if the hardware performs the same, the driver mode may be a source of latency. ASIO is almost always faster than MME, though many device manufacturers do not supply an ASIO driver. The use of third-party, device-agnostic drivers, such as ASIO4ALL (www.asio4all.com) allow you to wrap an MME-only device inside a faux-ASIO shell. The audio application believes it's speaking to an ASIO driver, and ASIO4ALL has been streamlined to work more quickly with the MME device, or even to allow you to use different inputs and outputs on separate devices which ASIO would otherwise prevent.
We also now see more USB microphone devices which are input-only audio devices that generally use a generic Windows driver and, with a few exceptions, rarely offer native ASIO support. USB microphones generally require a higher buffer size as they are primarily designed for recording in cases where monitoring is unimportant. When attempting to record via a USB microphone and monitor via a separate audio device, you're more likely to run into issues where the two devices are not synchronized or drift apart after some time. (The ugly secret of many device manufacturers is that they rarely operate at EXACTLY the sample rate specified. The difference between 44,100 and 44,118 Hz is negligible when listening to audio, but when trying to precisely synchronize to a track recorded AT 44,100, the difference adds up over time and what sounded in sync for the first minute will be wildly off-beat several minutes later.) You are almost always going to have better sync and performance with a standard microphone connected to the same device you're using for playback, and for serious recording, this is the best practice. If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device. Attempt to match the buffer sizes and sample rates as closely as possible, and consider using a higher buffer size and correcting the latency post-recording. (One method of doing this is to have a click or clap at the beginning of your session and make sure this is recorded by your USB microphone. After you finish your recording, you can visually line up the click in the recorded track with the click in the original track by moving your clip backwards in the timeline. This is not the most efficient method, but this alignment is the reason you see the clapboards in behind-the-scenes filmmaking footage.)
Other Hardware: Other hardware in your computer plays a role in the ability to feed or store audio data quickly. CPUs are so fast, and with multiple cores, capable of spreading the load so often the bottleneck for good performance - especially at high sample rates - tends to be your hard drive or storage media. It is highly recommended that you configure your temporary files location, and session/recording location, to a physical drive that is NOT the same as you have your operating system installed. Audition and other DAWs have absolutely no control over what Windows or OS X may decide to do at any given time and if your antivirus software or system file indexer decides it's time to start churning away at your hard drive at the same time that you're recording your magnum opus, you raise the likelihood of losing some of that performance. (In fact, it's a good idea to disable all non-essential applications and internet connections while recording to reduce the likelihood of external interference.) If you're going to be recording multiple tracks at once, it's a good idea to purchase the fastest hard drive your budget allows. Most cheap drives spin around 5400 rpm, which is fine for general use cases but does not allow for the fast read, write, and seek operations the drive needs to do when recording and playing back from multiple files simultaneously. 7200 RPM drives perform much better, and even faster options are available. While fragmentation is less of a problem on OS X systems, you'll want to frequently defragment your drive on Windows frequently - this process realigns all the blocks of your files so they're grouped together. As you write and delete files, pieces of each tend to get placed in the first location that has room. This ends up creating lots of gaps or splitting files up all over the disk. The act of reading or writing to these spread out areas cause the operation to take significantly longer than it needs to and can contribute to glitches in playback or loss of data when recording.

There is one point in the above that needed a little clarification, relating to USB mics:
_durin_ wrote:
If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device.
If you are going to spend that much, then you'd be better off putting a little more money into an external device with a proper mic pre, and a little less money by not bothering with a USB mic at all, and just getting a 'normal' condensor mic. It's true to say that over the years, the USB mic class of recording device has caused more trouble than any other, regardless.
You should also be aware that if you find a USB mic offering ASIO support, then unless it's got a headphone socket on it as well then you aren't going to be able to monitor what you record if you use it in its native ASIO mode. This is because your computer can only cope with one ASIO device in the system - that's all the spec allows. What you can do with most ASIO hardware though is share multiple streams (if the device has multiple inputs and outputs) between different software.
Seriously, USB mics are more trouble than they're worth.

HOWTO: Low DPC latencies ( 100 us) on bootcamped Macbooks (Pro)

Here is a small HOWTO for getting the lowest possible DPC latencies (<100 us) on bootcamped Macbooks Pro (late 2008):
Disclaimer: I did all tests on my late 2008 Macbook Pro Unibody 2.8 GHz model with NVidia chipset and graphic. Most of the following suggestions should apply to standard Macbook models and likely older generation as well.
First of all Intel Speedstep can lead to dropouts and higher DPC latencies on small load! Unfortunately all tools that are supposed to manually switch Speedstep off don't seem to run on the late Macbooks (Pro) while on OS X you can use "Coolbook".
Your only way to make sure your processor is clocked high enough and not dynamically switching is to put up a constant load (like running your DAW pretty hot or running Prime95 at "Idle/Lowest" process Priority in the background). I will keep investigating if I can find a tool to switch Speedstep off.
Most importantly (to get rid of really bad DPC latency spikes):
Kill the process "KBDMGR.EXE"!
That's Apple's driver for controlling brightness and keyboard lighting via the function keys and setting tap options for the trackpad. It seems to have broken multithreading!
You can also change the CPU affinity of KBDMGR.EXE to CPU1 (not CPU0!) which will help decreasing DPC Latencies alot, but there will still be Audio dropouts.
Here's a small toolkit I put together that allows you to conviniently enable/disable Apple's "Boot Camp" tray application (KBDMGR.EXE) via an icon link and/or keyboard shortcut. Optionally it will switch the function of the F-Keys automatically for you depending on whether Boot Camp is loaded or not.
Furthermore it automatically turns Boot Camp's CPU priority to "Idle" and CPU affinity to CPU1 in order to turn down the bug induced DPC Latencies and prevent dropouts with Windows sounds and Media Player playback. Professional Audio users will find that only turning off Boot Camp will allow low audio latency usage. Installation instructions are included in the README.TXT for your convinience.
Boot CampED download page
Direct Download:
Boot CampED.zip - 3.3 kb
Turn off the Broadcom 802.11N WLAN driver via Device-Manager or update to the latest drivers via Microsoft Update Catalog.
Like on OS X the Airport module can lead to audio dropouts. The DPC Latencies produced by the Broadcom driver are less regular than the KBDMGR thing, alot higher in value. Best thing is to try for your own needs.
Update:Meanwhile a new Broadcom drivers was published via Microsoft's Update Catalog named "Broadcom - Network - Broadcom 4322AG 802.11a/b/g/draft-n Wi-Fi Adapter " (4322 is the chip used). This one comes with both low DPC latencies and finally the ability to use the full rate upto 300 mbit/s. Go get it! For safety you might still want to turn WLAN off during critical audio work though.
Change the graphic-card driver to "Standard VGA Driver" via Device-Manager or use RIVATUNER to enforce a fixed clock-rate and performance mode.
Update:The dynamic clock-rate switching happening with NVidia drivers in order to save power and keep temperatures low leads to extreme DPC spikes for each switch and constantly high DPC latencies when it settles in low performance 2D mode. RIVATUNER's "Enforce Performance Mode" option can be used to set the card to a fixed clock-rate. I recommend using "Low Power 3D" for audio work.
User of XP might think that they don't need this, but be aware that on XP the NVidia driver keeps running at highest clock-rates in "Performance 3D Mode" all the time. Via RIVATUNER you can switch to "Low Power 3D".
Turn off the ACPI compliant Battery driver via Device-Manager
This driver polls the battery for its current load status and produces a small, single, short spike exactly every 15 seconds. In my own tests I found that it doesn't seem to affect low latency audio performance. Furthermore turning it off will remove monitoring of your current battery status. But if you are running on power-chord anyway and want to make absolutely sure you can turn it off.
All other devices don't add much if anything to DPC latencies, but can savely be turned off if you don't need them (like Nvidia LAN, Bluetooth, Onboard High Definition Audio).
Attention: Removing the Battery while the power chord is connected results in permanently reduced CPU clock (downto the lowest clock setting possible). According to Apple this is done to prevent overloading the power-supply during heavy load as it needs the assistance of the battery from time to time.

I'd like to underline that these are workaround. Now that the Broadcom drivers are fixed it is up to Apple to fix KBDMGR and to get the NVidia drivers fixed!
Furthermore it seems as if only Vista 32-bit and OS X are heavily affected by Intel Speedstep, Vista 64-bit and Windows 7 (32/64) work alot better in this regard. XP is a mixed bag.
Here are some screenshots to prove that the workarounds do help:
DPC Latency before applying the workarounds:
DPC Latency Vista 64-bit (Idle, Speedstep enabled) after applying the workarounds:
DPC Latency Windows 7 64-bit (Idle: Speedstep enabled) after applying the workarounds:
As you can see Vista's DPCs run well below 100 us once everything is optimized, Windows 7 is a bit worse, XP is even better. But practically you get the same results when using all three for professional Audio work.
Message was edited by: T1mur

Thinkpad Edge E440 Random Lags, Stuttering, Latency Problem.

Dear All, I bought Thinkpad Edge E440 almost two weeks ago. The problem is it randomly lags and stutter without reason. For example I was writing documents on Microsoft Word, plays music on foobar2000 and doing some light browsing (only few tabs) but randomly the music lags for 1-2 seconds. It doesn't skip, it lags or stutter (sorry I don't know the right term). Also sometime it took awhile to open a new tab on firefox web browser while usually it took instantly.
Another example is I play two online games, World of Tanks (WoT in short, a tank simulation game) and Ghost Recon Phantoms (a shooting game). When I play both game, the gameplay went smooth but randomly it lags or stutter like 1 second on more, then it went smooth again and then stutter again. The game doesn't became slow (which is called FPS drop if I am not wrong), the display just stop moving and mouse/keyboard become unresponsive during the lag/stutter period. On the other hand I also played WoT on my Pentium 4 PC and it went smooth without any lags, though of course the settings is inferior to my E440 laptop (Ghost Recon Phantoms can't be played on my Pentium 4 so I won't compare it).
Also sometimes the audio sounds 'spiky'.
As I searched online, people said that this is called/related to Latency problem and I should use LatencyMon to check it. After using LatencyMon I found several 'culprits' which causes big latency such as NDIS.sys, ACPI.sys, ATAPORT.sys, storport.sys, etc. I also tried to disable cpu throttling, power management settings and such but the problem doesn't go away.
I am sure that there is something wrong with my hardware because if its caused by settings such as cpu throttling, power management (which I think is software-related) then won't this happen to everybody? I mean, I asked a fellow E440 user and he said that his laptop is fine and my problems doesn't occur on him.
I have phoned Lenovo Customer Service. I was using Windows 7 Ultimate. At first they told me to do clean reinstall and install the drivers in the order the CS give me. I did clean reinstall 4 times to try different driver and program installation sequence to see if I might find a way to fix my problem, but I found no solution (I tried Lenovo CS driver install order on the last clean reinstall). I even tried running LatencyMon after clean reinstall before installing any drivers or updates but the LatencyMon still reports big, 'unhealthy' latency.
Lenovo CS then told me to do hardware scan on Lenovo Solution Center. There are warning on the Nvidia GT740M stress test but the CS told me that the results are good. His last solution is that I should go to a Lenovo Service Center to check for the problem there.
I moved to Windows 8.1 to see if the problem will be solved but too bad it's not. I have ran hardware scan for 5 times, there are two times(or 3 times I forgot) that there is warning on Harddisk SMART short self-test, 2 times warning on Nvidia GT740M stress test, one time scan finished with no warning. Even so, the end result for all the tests is 'Passed'.
I think the problem lies within my harddisk (or something related to data transfer) since when I browse my local drive in windows explorer sometimes it took a long time for the icons to load but sometime it just loads instantly. And I am not sure what that Nvidia stress test warning has to do with.
Also I only use the stock 4GB Ram. Could this related to the problem? Though when I look at Windows 8.1 Task Manager the memory usage rarely goes above 60%.
Last but not least. I plug a wireless mouse on my laptop which I use mostly for playing games, but in the end I switches between mouse-touchpad-mouse and so on during working too. Sometime my touchpad suddenly went unresponsive for few seconds but my mouse is still usable during that moment. Then after I move my wireless mouse, my touchpad become responsive again. Any idea why this is happening? I think this is probably because touchpad driver not perfectly compatible with Windows 8.1 as I don't recall having this trackpad issue on WIndows 7.
Can somebody help me? Does anyone else ever had the same problem as I do?
I am sorry for the really long explanation. Your response will be greatly appreciated.
Thank you.
-Tahta

Hi All, sorry for being a while for a long time.
First of all, I should say that I am not sure if my trackpad/pointing device problem is related to the lag/stutter/latency problem so I should have made two different thread instead. Sorry if anyone got confused with this.
@rallygion, the first OS I used on this laptop was Windows 7 64bit. If I remember correctly I don't experience the trackpad/pointing device problem back then, maybe I was just forgotten tho. I am using Windows 8.1 starting sometime before I made this thread. Now the problem has gone worse, sometime the touchpad & red dot in the keyboard won't work for a whole day so I must use external mouse.
At first I thought its compability problem with Windows 8.1 but now I think the hardware has problem too. However, the keyboard is fully functional.
@waqs, thanks for your suggestion! Unfortunately it doesn't solve the lag/stutter/latency problem.
@ghasan, our problem is quite similar except I didn't use the M.2 SSD. One thing I notice is the disk activity sometimes jumps to 100% when the lags happened.
The problem doesn't lie with the laptop specification is inferior. Fyi, when I play music on foobar2000 it took few second for music to start playing (foobar says 'starting playback..') while on my pentium 4 desktop (which I still use) the music plays right after I clicked the song.
Also, eventhough I've closed all my applications, leaving only music player and microsoft word/firefox, the music which is playing lags (randomly stops for a moment).
The lags also happened when watching videos. I played video on Media Player Classic, turned the statistics on. There happen to be frame drops sometimes. Not framerate drop, just frame dropped. I mean, I think framerate dropping has relation with computer power while in my case the frame just randomly drops, which 'damages' the video, not slowing it down.
I am sure that this is hardware-related problem. I am going to take my laptop to service center but not now since I am in need on my laptop now.
I will give update later when I have gone to the service center.

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