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Does Cakewalk have some type of latency compensation feature?
Old 12th March 2019
  #1
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Does Cakewalk have some type of latency compensation feature?

Cakewalk 2018.11 (Build 31, 64bit)

I have recorded some instrument tracks and I want to record a vocal track but when I record the vocal track it lags the instrument tracks by approximately 125 milliseconds. Is it possible to set latency compensation during the recording so that the vocal track will match the instrument tracks? When I record the vocal track I have the instrument tracks playing at a low volume through my monitor speakers and I use a unidirectional mic for the vocals pointed away from the speakers so it doesn’t pick up the instrument tracks.

Thank you in advance for any help,
yeto
Old 12th March 2019
  #2
Gear Nut
 

125ms is huge. I take it you are not recording with an ASIO interface?
Old 12th March 2019
  #3
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Quote:
Originally Posted by SluttyMcSlut View Post
125ms is huge. I take it you are not recording with an ASIO interface?
I tried using ASIO which gave me effective latency of11.2 msec but my computer mic wouldn't work. I am new at this and I don't know how to assign to get it to work. I am using WASAPI shared which gives me 10.0 msec effective latency. I use my computer for work so I need my audio to work for other applications also. I don't know if this is possible with ASIO. Is it strange for WASAPI to be faster versus ASIO?

I am getting the 125ms delay measurement from track view. If I "nudge" the vocal track approximately 125ms to the left the vocals line up with the instruments. I was hoping there was some way this could happen during recording so that I would not have to use the nudge tool.

Again, I am new at this. Any suggestions as to what I am doing wrong or a technique that could make this easier would be greatly appreciated.

Thank you for taking time to reply,
yeto
Attached Thumbnails
Does Cakewalk have some type of latency compensation feature?-asio.jpg   Does Cakewalk have some type of latency compensation feature?-wasapi.jpg  
Old 12th March 2019
  #4
Gear Nut
 

Quote:
Originally Posted by yeto View Post
I tried using ASIO which gave me effective latency of11.2 msec but my computer mic wouldn't work. I am new at this and I don't know how to assign to get it to work. I am using WASAPI shared which gives me 10.0 msec effective latency. I use my computer for work so I need my audio to work for other applications also. I don't know if this is possible with ASIO. Is it strange for WASAPI to be faster versus ASIO?

I am getting the 125ms delay measurement from track view. If I "nudge" the vocal track approximately 125ms to the left the vocals line up with the instruments. I was hoping there was some way this could happen during recording so that I would not have to use the nudge tool.

Again, I am new at this. Any suggestions as to what I am doing wrong or a technique that could make this easier would be greatly appreciated.

Thank you for taking time to reply,
yeto
If you are only going to use a computer mic, you could try changing the time ruler to milliseconds and then starting the recording 125ms before it needs to start. I'm not sure if that will work though?

If you plan on recording regularly I would recommend getting an ASIO recording interface. It will give you a much smaller delay, but it will also have automatic compensation, so everything will just line up without having to mess around. You can also direct monitor with headphones from an interface, so you won't have to worry about playback noise getting into a mic.
Old 12th March 2019
  #5
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Quote:
Originally Posted by SluttyMcSlut View Post
If you are only going to use a computer mic, you could try changing the time ruler to milliseconds and then starting the recording 125ms before it needs to start. I'm not sure if that will work though?

If you plan on recording regularly I would recommend getting an ASIO recording interface. It will give you a much smaller delay, but it will also have automatic compensation, so everything will just line up without having to mess around. You can also direct monitor with headphones from an interface, so you won't have to worry about playback noise getting into a mic.
I take it Cakewalk does not have a "latency compensation" feature/setting like some of the other available DAWs?

Thank you for taking time out of your day to reply,
yeto
Old 12th March 2019
  #6
Gear Nut
 

Quote:
Originally Posted by yeto View Post
I take it Cakewalk does not have a "latency compensation" feature/setting like some of the other available DAWs?

Thank you for taking time out of your day to reply,
yeto
Edit > Preferences (ensure "Advanced" is selected at the bottom) > Audio > Sync and Caching

Then go to: Record Latency Adjustment > Manual Offset

The adjustment is in samples. To calculate milliseconds to samples, you multiply the milliseconds of delay by the project sample rate.

So if your project is set to 44.1 and you are getting 125ms of delay, the formula is 125 X 44.1 = 5512.5

I don't think you can pick a decimal, it has to be a whole number so I guess you can decide to go with either 5512 or 5513.

I'm not sure if you will need the offset to be a negative or positive number? If you punch in say 5512 and the audio winds up being even more delayed, then make it -5512 and see how that goes.

Last edited by SluttyMcSlut; 12th March 2019 at 03:35 PM.. Reason: Needed a bit more.
Old 12th March 2019
  #7
Here for the gear
 

Quote:
Originally Posted by SluttyMcSlut View Post
Edit > Preferences (ensure "Advanced" is selected at the bottom) > Audio > Sync and Caching

Then go to: Record Latency Adjustment > Manual Offset

The adjustment is in samples. To calculate milliseconds to samples, you multiply the milliseconds of delay by the project sample rate.

So if your project is set to 44.1 and you are getting 125ms of delay, the formula is 125 X 44.1 = 5512.5

I don't think you can pick a decimal, it has to be a whole number so I guess you can decide to go with either 5512 or 5513.

I'm not sure if you will need the offset to be a negative or positive number? If you punch in say 5512 and the audio winds up being even more delayed, then make it -5512 and see how that goes.
This "almost" works but the highest number that Cakewalk will let me enter is 5000. I am thinking maybe I could get that number down if I could use ASIO but I can't get my computer mic to work with ASIO. Do you see anything in the attached photos that I could change to get the mic to work?

Thank you and thank you for taking time to help,
yeto
Attached Thumbnails
Does Cakewalk have some type of latency compensation feature?-asio.jpg   Does Cakewalk have some type of latency compensation feature?-asio-mic-setup.jpg  
Old 13th March 2019
  #8
Gear Nut
 

Quote:
Originally Posted by yeto View Post
I tried using ASIO which gave me effective latency of11.2 msec but my computer mic wouldn't work. I am new at this and I don't know how to assign to get it to work. I am using WASAPI shared which gives me 10.0 msec effective latency. I use my computer for work so I need my audio to work for other applications also. I don't know if this is possible with ASIO. Is it strange for WASAPI to be faster versus ASIO?
yeto
Are you on windows 10? I seem to recall that windows made some big improvements to WASAPI and vaguely recall some people saying they were getting as low as 4-5ms latency.

Also what kind of mic are using?
Old 13th March 2019
  #9
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Quote:
Originally Posted by SluttyMcSlut View Post
Are you on windows 10? I seem to recall that windows made some big improvements to WASAPI and vaguely recall some people saying they were getting as low as 4-5ms latency.

Also what kind of mic are using?
Yes, I am on Windows 10. The mic is an off brand and it is plugged into the 3.5mm jack on the back of my tower. Would a USB mic make a difference?
Old 13th March 2019
  #10
Gear Nut
 

Quote:
Originally Posted by yeto View Post
Yes, I am on Windows 10. The mic is an off brand and it is plugged into the 3.5mm jack on the back of my tower. Would a USB mic make a difference?
A USB mic might make a difference, as I know some of them have their own driver system that allows them to be seen/utilised by different capture modes. To be honest I have never tried to capture audio straight into a jack in the pc, so unfortunately I can't give any tips in that regard.

Here is the Cakewalk documentation for using WASAPI:

WASAPI support
SONAR provides support for WASAPI (Windows Audio Session API), which is the new standard for audio in Windows 7 and future Windows operating systems.
The primary advantages of WASAPI are:
*
Better compatibility with consumer audio devices:
*
Supports a wide variety of audio devices, including newer WaveRT devices and legacy Windows audio devices.
*
Lets you use consumer audio devices that don’t have ASIO drivers or drivers that don’t work well under WDM.
*
Provides low latency access to audio devices.

Using WASAPI
WASAPI has four different modes of operation. SONAR supports WASAPI Exclusive mode and WASAPI Shared mode.
Exclusive mode provides low latency audio and direct access to the audio driver, bypassing the Windows audio engine. The disadvantage is that only one application can access the driver in Exclusive mode.
Shared mode allows multiple applications to access the audio driver. When Shared mode is selected, Windows will mix the audio output from SONAR along with audio from other apps such as YouTube. One advantage of Windows 10 is that WASAPI Shared mode supports low latency playback and recording at buffer sizes down to 2 milliseconds. In Windows 7, WASAPI Shared mode is less efficient and requires higher latency since it uses polling to access the driver.
*
In order to use WASAPI in SONAR, your audio device must have WDM drivers that Windows can successfully communicate with. In order to use WASAPI Exclusive mode, the audio device must be enabled to work in Exclusive mode.
Exclusive mode is the default in Windows 7 and can be configured from your audio device’s taskbar or Properties dialog box in Device Manager. Consult your audio device’s manual if you need further information.
Exclusive mode requires exclusive access to the audio device. Another program will not be able to use the audio device simultaneously with SONAR.
*
Tip: If you select Suspend Audio Engine When SONAR Is Not in Focus in Edit > Preferences > Audio - Playback and Recording, SONAR will relinquish control of the audio device when the program loses Windows focus.
To enable WASAPI mode in SONAR
1. Go to Edit > Preferences > Audio - Playback and Recording.
2. In the Driver Mode list, select WASAPI Exclusive or WASAPI Shared.
3. Select the Audio - Devices page tab (Edit > Preferences > Audio - Devices).
4. In the Input Drivers and Output Drivers lists, select the desired input and output ports.
5. Click OK to close the Preferences dialog box.

To enable/disable built-in DSP
In Windows 10, WASAPI Shared mode has support for configuring built-in signal processing for inputs and output independently. Many onboard audio devices have built in DSP, such as Dolby audio enhancement of audio outputs and noise reduction for audio inputs. While these effects may be desirable for better quality audio from laptops (for example, for noise gating of microphone inputs), the DSP processing of these effects can cause latency.
SONAR allows you to enable or disable DSP effects on inputs and output devices in WASAPI Shared mode. This is done via the EnableWasapiDSP variable in Aud.ini. By default, SONAR enables endpoint DSP effects for input devices (for noise removal).
To configure the built-in signal processing for inputs and outputs:
1. Go to Edit > Preferences > Audio - Configuration File.
2. Under Configuration Settings, click Edit Config File.
The Aud.ini file opens in the default Windows text editor.
3. In Aud.ini, add a new entry called EnableWasapiDSP in the [Wave] section, and set its value to the minimum desired buffer sample value. For example:
[Wave]
EnableWasapiDSP=0
Valid values are as follows:
*
0 = Disable all signal processing for inputs and outputs (RAW mode)
1 = Enable signal processing for inputs (default)
2 = Enable signal processing for outputs
3 = Enable signal processing for inputs and outputs
4 = Save Aud.ini and close the Windows text editor.
5 = Click Reload Config Settings to reload the current audio configuration settings from Aud.ini.
*
Note: Setting the EnableWasapiDSP variable to 2 or 3 will result in a minimum latency of 5 milliseconds in SONAR when using WASAPI Shared mode.
Old 13th March 2019
  #11
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Quote:
Originally Posted by SluttyMcSlut View Post
A USB mic might make a difference, as I know some of them have their own driver system that allows them to be seen/utilised by different capture modes. To be honest I have never tried to capture audio straight into a jack in the pc, so unfortunately I can't give any tips in that regard.
First of all let me say thank you for taking time to help. I am new at this and I appreciate all your suggestions.

I have a very cheap USB mic that I tried and it took care of the latency problem but it seemed to introduce some static into the audio during playback until I disconnected the mic from the computer. Once the mic was disconnected the audio cleared up. I am going to try and find a better USB mic and see if that will make a difference. I will report back and share my findings.

Again, thank you for taking time out of your day to help,
yeto
Old 13th March 2019
  #12
Here for the gear
 

[QUOTE=SluttyMcSlut;13861432]A USB mic might make a difference, as I know some of them have their own driver system that allows them to be seen/utilised by different capture modes. To be honest I have never tried to capture audio straight into a jack in the pc, so unfortunately I can't give any tips in that regard.

Just an update.

I tried a different USB mic and got the same static effect using WASAPI Shared mode. Basically what is happening is I am getting a lot of static during playback but it is eliminating the latency problem. If I change the mode to MME (32 bit) the static disappears. MME, for my amateur projects, will be good enough for me. All my instrument tracks are entered using piano roll and I just need to sync 1 or at the most 2 vocal tracks to the instrument tracks.

Again, thank you for all your help. I learned a lot trying to sort this out and that will help me now and in the future.

Thank you for taking time out of your day to help,
yeto
Old 14th March 2019
  #13
Gear Nut
 

[QUOTE=yeto;13863218]
Quote:
Originally Posted by SluttyMcSlut View Post
A USB mic might make a difference, as I know some of them have their own driver system that allows them to be seen/utilised by different capture modes. To be honest I have never tried to capture audio straight into a jack in the pc, so unfortunately I can't give any tips in that regard.

Just an update.

I tried a different USB mic and got the same static effect using WASAPI Shared mode. Basically what is happening is I am getting a lot of static during playback but it is eliminating the latency problem. If I change the mode to MME (32 bit) the static disappears. MME, for my amateur projects, will be good enough for me. All my instrument tracks are entered using piano roll and I just need to sync 1 or at the most 2 vocal tracks to the instrument tracks.

Again, thank you for all your help. I learned a lot trying to sort this out and that will help me now and in the future.

Thank you for taking time out of your day to help,
yeto
No problems. Glad you have got it working!

In the future if you are looking to take the next step in quality for your productions, you will likely have to look into getting an audio interface. As pre-amps will give your input signal a lot more body and presence to help it standout in the mix.
Old 14th March 2019
  #14
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[QUOTE=SluttyMcSlut;13863794]
Quote:
Originally Posted by yeto View Post

No problems. Glad you have got it working!

In the future if you are looking to take the next step in quality for your productions, you will likely have to look into getting an audio interface. As pre-amps will give your input signal a lot more body and presence to help it standout in the mix.
Thank you. I will definitely look into getting an audio interface.
Old 30th March 2019
  #15
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johnnyv's Avatar
It is a huge waste of time to try and do any serious audio recording with any DAW without using a proper ASIO audio interface. It does not have to be top of the line, a $100 will do. You will be fighting a uphill battle if you try and cludge a workable audio system out of a $10 sound chip. Talk about Low quality audio. USB mikes are for pod casting where latency does not matter.
WASAPI works great for just puttering around editing already recorded material and working with midi. But overdubbing audio will always be out of sync in any other mode other than ASIO.
Old 31st March 2019
  #16
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Quote:
Originally Posted by johnnyv View Post
It is a huge waste of time to try and do any serious audio recording with any DAW without using a proper ASIO audio interface. It does not have to be top of the line, a $100 will do. You will be fighting a uphill battle if you try and cludge a workable audio system out of a $10 sound chip. Talk about Low quality audio. USB mikes are for pod casting where latency does not matter.
WASAPI works great for just puttering around editing already recorded material and working with midi. But overdubbing audio will always be out of sync in any other mode other than ASIO.
Thank you for your help. I will upgrade in the future.

yeto
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