[music-dsp] WDM for low(er) latency?
iain at idcl.co.uk
Sun Mar 20 05:26:21 EST 2005
What latency are you aiming for?
You can get down to 20 - 30 mS with DirectShow or DirectSound. In both
cases the minimum latency is gated by the system mixer (KMixer if I recall)
which has a 20 - 30 mS buffer. The Audio input devices in Direct Show have
a typical latency of 500 MS, but you can change this by setting the
preferred buffer size IAMBufferNegotiation interface before connecting the
filter up. You can get the latency of the input device quite low, but you
are still faced with the 20 - 30 mS KMixer delay.
If you want to get down to say 2 mS then you will probably need to use the
kernel streaming drivers (WDM Streaming System devices / WDM Streaming
capture devices). These will work quite nicely with latencies as low as 2
mS. Though as you say there are questions about if this is the long term
model that MS will use (though they WILL be supported for quite some time, I
HOWEVER. This is black art area. The documentation is sparse even by
Direct Show standards and most of it is in the DDK area (which is obscure
even by MS standards!).
Some factors here are.
1. You will need to mess with thread or process priority to prevent
system operations from blocking the audio (causing breakup).
2. You will have to take exclusive control over the device. You cannot
mix streams from other processes (like an standard audio player).
3. You will see a great deal of device dependency. My tests were with
a midiman delta 1010 and the latency related properties seemed unsupported
instead you controlled latency through the control panel.
From: music-dsp-bounces at ceait.calarts.edu
[mailto:music-dsp-bounces at ceait.calarts.edu] On Behalf Of James Chandler Jr
Sent: 19 March 2005 17:06
Subject: [music-dsp] WDM for low(er) latency?
We have been using MME for a long time, which isn't low latency, but is
very reliable, and works for 99.9 percent of customers. Have lately been
our feet into ASIO, mainly for low latency softsynth playthru. ASIO seems
Would like to improve Windows Driver latency as much as possible without
a big science project out of it. Kernel Streaming support is probably too
learning curve, since ASIO will in most cases deliver equivalent results,
for folks who don't have ASIO soundcards and can't/won't use generic drivers
like ASIO4ALL. Anyway, Microsoft has been threatening to deprecate kernel
streaming for what, 4 years now?
Spent half of yesterday googling third-party references on WDM support, but
there isn't as much third-party info as expected, except for game sound.
the other half of the day re-reading the latest MS DirectShow and
On XP SP2, as best I can tell, the minimum practical latency achievable with
MME, on two soundcards I tested, appears to be in the ballpark of 200 mS
That figure was measured with a WOMDone callback function to flag returned
buffers, and a hi-res 1 millisecond timer thread polling for returned buffer
flags. No PostMessage delays that I can see in the test code. Maybe there
some soundcards that give better MME latency than that, but it sure looks
least on my computers) XP is using a 32768 byte MME sound output buffer, and
its output mixing buffer gets much emptier than 32768 bytes, you get
regardless of how many small buffers you use, and how promptly you
Perhaps Microsoft adjusted those details in SP2 or DirectX 9? Think I did
same tests a year ago showing the minimum MME latency closer to 100 mS (on
same test computers), but maybe remembering wrong.
It would be nice to at least get Windows sound latency below 50 mS for 'most
customers with generic soundcards', with extra points if able to dip below
for folks with decent soundcards.
As far as I can tell, other than 'Kernel Streaming', there are two routes to
more directly talk to WDM drivers, DirectSound and DirectShow. Are there
Both DirectSound and DirectShow seem about equally complex to implement
DirectShow MIGHT be less complicated to implement. For playback would only
to write an ASYNC pull-model source filter and wire it to a Render filter.
sound input, probably just wire a Capture filter to a push-model destination
filter I already have.
DirectSound is a bit off-putting because of the API's game-orientation. At
I already know a bit about DirectShow. Enough to know it can sometimes take
much hacking before DirectShow actually works as expected (GRIN).
Which approach is most likely to give low-latency without becoming a long
science project? Have tested some Softsynth plugins that use DirectSound in
standalone mode, which have pretty good realtime MIDI play-thru latency. So
apparently at least DirectSound is capable of doing the task? Haven't heard
sequencers/softsynths using DirectShow for realtime i/o, but that doesn't
it isn't common.
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