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Leif.
Today I did a test with Breakaway Live in one of our studios to see if it was possible to use it for the studio monitor and headphone feeds. They hate me when I get into "I’d like to try .." mode. 🙂
I patched the monitor output of that studio Klotz console to the input of Breakaway Live and then patched the output of Breakaway Live back to the monitoring input for the Klotz. Seemed OK to route that way. I had Live set to Low Latency (4ms) but the latency appeared to be greater than 4ms and the "stars" in the studio told me it was unusable due to the delay in their headphones.
Is the latency a product of the Buffer Size settings, and Buffer Count for the Input AND Output in the config, as well as the Latency Setting? If so, are there any suggested settings for "Live" use?
I’d like to experiment with this a little further later in the week as I am sure there are feeds where we could put Live and Broadcast to use.
Thanks,
ScottScotty, YES. The latency is a product of Sound Card input buffering, output buffering, SRC thread buffering, and the core latency itself. They all add up!
With the core in Low Latency core, the core only adds 4ms of latency (in phase linear, the core adds 100ms).
By using very small Kernel Streaming buffer settings, (for example 96×3 for input and output, if your sound card drivers can handle it) you can get reasonable latency, but it will still be 20ms or so at best with kernel streaming.
For true low latency, you need an ASIO capable sound card. Most semi-pro and pro cards are ASIO capable, and using ASIO you can have a total throughput of 10ms or less.
The cheapest ASIO capable card I know is ASUS Xonar D2X. That’s pro-sumer, and you can have one for under $100.
For semi-pro, look at ESI Juli@ or M Audio’s Delta / Audiophile cards ($150 or so). For pro, the sky is the limit (both functionality wise and price-wise).
Best,
///LeifLeif,
Thanks for the reply.
We have a combination of brands of soundcards based on what they are used, or supplied, in. I know we have a spare Audioscience card that will run with ASIO drivers that I can try. I also noticed today we have a spare Digigram card but I didn’t see any mention of ASIO for that but I am sure that it has ASIO drivers. One of the engineers in one of our recording/production studios has a setup with 2 RME 9652 cards that he runs with ASIO at 2 x 32 buffers for doing overdub vocals and he showed me today he is getting close to live throughput with those cards. Close enough that performers can do vocal overdubs with themselves without noticing any latency. I think they were ADAT in/out.
What is the lowest you recommend for the settings for Buffer Size and Count for In and Out for Breakaway Live?
I may be chasing my tail with this but I’ll be interested to see how close I can get. If nothing else they’re interesting toys to play with. 🙂
The main problem, as I notice you mention in another post in another topic, is everything you introduce into the signal path introduces more latency. 😯
Scott
With a current production RME of Analog IO… like:
HDSP(e) AIO
HDSP 9632
Fireface
Multiface IIyou should be able to get total latency down to like 5ms. 8)
Or if you wanna get into the high-end (and expensive) MADI stuff…
ADI-8 QS
Micstasy
(etc…)
with matching MADI networking and interface like
HDSP(e) MADI
HDSP MADIface
ADI-642
ADI-6432
ADI-648
MADI Bridge
MADI Converter
(etc…)Super low latency digital "HD" audio networking/routing over long distance fiber… with AES conversion too if you don’t like/want RME converters. 🙂 Amazing system to work with.
With ASIO, the lowest buffer size the Breakaway Core will support, is 48 samples, and in this mode the core will run in Extra Low Latency mode. 32 samples will not work under any circumstances.
With Kernel Streaming, 96/3 is the lowest I can recommend, and even that is not supported by most configurations.
///Leif
quote :With ASIO, the lowest buffer size the Breakaway Core will support, is 48 samples, and in this mode the core will run in Extra Low Latency modeWhat sort of tradeoffs are there for operating in the extra low-low latency modes versus phase linear? Is the difference audible? What is the lowest latency that can be achieved with phase linear mode?
On a similar note, what are the tradeoffs in BBP in the CPU efficiency mode (whatever it’s called)?
Thanks,
StuartHi Stuart!
Between Phase Linear and Low Latency, the difference is.. Well.. Phase linearity.
If you’re listening carefully, the difference is clearly audible while switching back and forth, especially with aggressive processing settings. If latency isn’t an issue, it’s a no-brainer — use Phase Linear. If latency IS an issue, again it’s a no-brainer — use Low Latency.
Extra low latency mode cuts down on the look-ahead time of the bass bands. In Normal Low latency / Phase Linear mode, the look-ahead for the bass bands (b1, b2) is 2ms. In Extra Low Latency mode this gets cut down to 1ms, translating to ever so slightly dirtier bass. However, if latency requirements are *that* tight, again it’s a no brainer!
BBP is always phase linear. CPU optimized vs Maximum Quality in BBP switches clipper modes. Maximum Quality uses more stages so that it can spread out the work, mask the distortion slightly better, and have ever so slightly better peak control. However, CPU optimized is very close is quality (let’s say 95% of the quality with 50% less cpu use) so if you’re short on CPU, this decision is also a no-brainer.
///Leif
Leif,
I’m having trouble getting any low latency settings to work with the Audioscience card. Maybe the card isn’t up to working with low latency settings in ASIO. In broadcast use, with our automation systems, I am sure we use them with the Wave drivers. I’ll venture to see if I can "borrow" one of the RME cards we use in production as, as JessieG points out, they seem happier to work with low latency in ASIO.
As an aside queston, in Breakaway Pipeline I see they are setup as 1ms per Interrupt. In VAC they are setup as 10ms per Interrupt. Does this do anything that adds to latency when used linking one application to BBP or Live?
Scott
Hi Scott!
The VAC default is 10ms per interrupt — I had him change it to 1 for my licensed version (Breakaway Pipeline). 1ms per interrupt enables lower latency through the pipeline.
Regarding the audioscience card, I’ve never tried it with Breakaway Live, and it’s not impossible that it’s using an ASIO data format not supported by Breakaway.
Breakaway currently supports 16 and 32 bits. 24 bits is currently not supported at all, and will yield silence. Most 24 bit cards actually deliver the audio in padded in 32 bit format, so it works. However, Asus Xonar DX, when set to 24-bit, really delivers 24-bit, and as such doesn’t work with Breakaway Live.
However, Asus Xonar DX can be set to 16-bit, and then it works perfectly. Perhaps the AudioScience card has a similar setting?
///Leif
Leif,
[quote author=”Leif”]The VAC default is 10ms per interrupt — I had him change it to 1 for my licensed version (Breakaway Pipeline). 1ms per interrupt enables lower latency through the pipeline.[/quote]Is there any problem with me setting the VAC ones to 1ms?
[quote author=”Leif”]Perhaps the AudioScience card has a similar setting?[/quote]I haven’t had any time to experiment with it further but should get some time in the next few days. I’ll investigate it further and let you know the final result.
Scott
Leif ,
[quote author=”Leif”]Nope, go ahead. VAC runs fine with 1ms timing.[/quote]I have changed all 3 VACs to 1ms. Does this reduce latency. I couldn’t figure out from the VAC help docs whether it did or not.
Playback still works fine with it at 1ms.
Scott
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