04-22-2020, 02:20 PM
Good stuff GDJ. Thanks for posting. For anyone reading your post and looking at changing the bit rate, also note his Audio Frame Size is 1.0.
This setting will give you the lowest latency. The higher the audio frame size, the higher the latency you will have. Because we are dealing with round trip audio, the added latency is essentially double the audio frame size (that's my theory anyway - and you can confirm the numbers if you look at your interface latency before and after a change). So an audio frame size of 1 = 2ms added latency to your interface. Audio frame size of 2 = 4ms of added latency. 2.5 = 5ms of additional latency, and so on. If you have a newer powerful computer you should be able to run at frame size of 1, but test it out. If you hear clicks, pops, etc., you will need to change to a higher frame size. I am using a very powerful computer but it is 12 years old and I suspect the bus speed may be limiting this, but it could have something to do with the os (I'm stuck on 10.10.5 until I buy a new computer). So the lowest setting I can use is 2.5. If I upgrade and can get to an audio frame size of 1 I would shave off 3ms of latency. FWIW, everyone I know that is using a newer Macbook Pro has no issue running a frame size of 1.
I haven't seen a measurable change when switching to lower bit rate (I was the only one that changed it - while the others looked at my numbers), but it makes sense that if everyone did it things could be better. I will have everyone try it and see what happens. There is definitely a noticeable difference in quality as you change this setting. Low and high end will suffer digital artifacts more noticeably (especially for bass players and other instruments heavy in those frequencies). You will also notice a change in how forward the mids are. I actually prefer the 320 setting to the 512, all else being equal (that is for vocals and electric guitar anyway). Ultimately, if you can't play with someone else due to sync issues, it doesn't matter how good it sounds. So if lower quality helps you be able to play, that is what you need to do. I can deal with lower quality audio, but not big delay and time $fts.
One other quick note. These setting are found as GDJ's pic shows (network settings), but can also be found in Manage --> Audio Settings --> Audio Booster. They are the same window, just two different ways to get there.
And while we are at it, play with your interface's sample rate. One person I play with was able to make a considerable savings in his interface latency by changing his interface's sample rate from 44.1 to 48. And JK allows for each player to be a different sample rates. Not sure if this creates other issues though. I could see how it may increase JK's buffer needs or possibly jitter. I think that is because it is converting your audio to the bitrate you select (see above) and that is what is being sent. Not sure how it deals with mixed bit rates on the other end though. I have a feeling that the more settings everyone in a session has the same, the smother everything will work. Although, I have no proof of this as I haven't even tested it and don't know how the software is handling many things. But ultimately, if someone can shave 10 or more ms off their latency, that may be a better return then slightly reduced jitter or something. After all, I think the system has to cater to the worst setup in the group of players. That means that everyone's experience will significantly improve if one person can shave that much off of their latency. Good luck out there!
This setting will give you the lowest latency. The higher the audio frame size, the higher the latency you will have. Because we are dealing with round trip audio, the added latency is essentially double the audio frame size (that's my theory anyway - and you can confirm the numbers if you look at your interface latency before and after a change). So an audio frame size of 1 = 2ms added latency to your interface. Audio frame size of 2 = 4ms of added latency. 2.5 = 5ms of additional latency, and so on. If you have a newer powerful computer you should be able to run at frame size of 1, but test it out. If you hear clicks, pops, etc., you will need to change to a higher frame size. I am using a very powerful computer but it is 12 years old and I suspect the bus speed may be limiting this, but it could have something to do with the os (I'm stuck on 10.10.5 until I buy a new computer). So the lowest setting I can use is 2.5. If I upgrade and can get to an audio frame size of 1 I would shave off 3ms of latency. FWIW, everyone I know that is using a newer Macbook Pro has no issue running a frame size of 1.
I haven't seen a measurable change when switching to lower bit rate (I was the only one that changed it - while the others looked at my numbers), but it makes sense that if everyone did it things could be better. I will have everyone try it and see what happens. There is definitely a noticeable difference in quality as you change this setting. Low and high end will suffer digital artifacts more noticeably (especially for bass players and other instruments heavy in those frequencies). You will also notice a change in how forward the mids are. I actually prefer the 320 setting to the 512, all else being equal (that is for vocals and electric guitar anyway). Ultimately, if you can't play with someone else due to sync issues, it doesn't matter how good it sounds. So if lower quality helps you be able to play, that is what you need to do. I can deal with lower quality audio, but not big delay and time $fts.
One other quick note. These setting are found as GDJ's pic shows (network settings), but can also be found in Manage --> Audio Settings --> Audio Booster. They are the same window, just two different ways to get there.
And while we are at it, play with your interface's sample rate. One person I play with was able to make a considerable savings in his interface latency by changing his interface's sample rate from 44.1 to 48. And JK allows for each player to be a different sample rates. Not sure if this creates other issues though. I could see how it may increase JK's buffer needs or possibly jitter. I think that is because it is converting your audio to the bitrate you select (see above) and that is what is being sent. Not sure how it deals with mixed bit rates on the other end though. I have a feeling that the more settings everyone in a session has the same, the smother everything will work. Although, I have no proof of this as I haven't even tested it and don't know how the software is handling many things. But ultimately, if someone can shave 10 or more ms off their latency, that may be a better return then slightly reduced jitter or something. After all, I think the system has to cater to the worst setup in the group of players. That means that everyone's experience will significantly improve if one person can shave that much off of their latency. Good luck out there!