[linux-audio-user] emu10k1 multichannel support

[Shayne O'Connor]

Lee Revell wrote:

>I posted this to alsa-devel but since my previous post on this list
>generated a lot of interest, I am just reposting it here.
>
>As promised, here's an updated patch to add real multichannel playback
>support (and improved multichannel capture) to the emu10k1 driver.
>
>http://www.alsa-project.org/~rlrevell/emu10k1-multichannel-v001.patch
>
>Please test it and report any problems.  I am especially interested in
>any regressions that impact regular PCM playback (the hw:0,0 device).
>
>QuickStart:
>
>$ jackd -R -v -d alsa -P hw:0,3 -C hw:0,2 -S
>
>I tested this and it works well with 16in/16out at 128, 256, 512 frames.
>32 and 64 should work too but I can't test as I'm running a stock 2.6.10
>kernel for now ;-).  You can check that the routing is correct by
>connecting a JACK client to the playback ports corresponding to the FX
>buses described in Documentation/Audigy-mixer.txt and
>Documentation/SB-Live-mixer.txt, and verifying that the output appears
>on that channel (the FX buses are numbered from 0 but JACK numbers
>clients from 1).  For example (from SB-Live-mixer.txt):
>
>    name='Music Playback Volume',index=0
>
>    This control is used to attenuate samples for left and right MIDI FX-bus
>    accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
>    The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
>
>So "alsaplayer -o jack -d alsa_pcm:playback_5,alsa_pcm:playback_6"
>should output to FX buses 4 and 5, which you can test by lowering the
>'Music' control in alsamixer.  With an SBLive, use ports 1 and 2 for the
>front channels, 3 and 4 for the rear channels.  The Audigy uses
>different channels, see the above docs for more info.
>
>In addition to multichannel recording applications, this should also be
>useful for OpenAL implementations, which are currently restricted to
>using 21 sources due to the use of an extra voice per stereo PCM.  This
>should allow up to 63 sources.
>
>This also adds some new register info including a per channel half loop
>interrupt that I have discovered by reverse engineering the Windows
>drivers.
>
>Improvements over previous versions:
>
> - Routes the 16 channels to the 16 FX buses by default.
>
> - Enables the first 16 FX capture outputs by default, required for 
>full duplex operation at latencies lower than 512 frames.
>
> - Rewrote the voice allocator to use a more efficient round 
>robin algorithm, eliminating the need to reserve the
>first 16 voices for the multichannel device.  The next free voice 
>is maintained in the card record and the search starts from there.
>
> - Use an extra voice for playback timing rather than the EFX capture
>interrupt.  I was only ever able to get that to work at 64 frames.  Also
>there are definite advantages to being able to use the capture and
>playback devices independently.
>
> - Use the newly discovered per-channel half loop interrupt source for
>the extra voice rather than the channel loop interrupts.  For unknown
>reasons, this works better for multichannel playback, and does not seem
>to affect regular PCM playback at all.
>
>TODO:
>
> - Fix the send routing and volume controls for the multichannel device.
>The current (copy and paste) solution assumes either one or two voices
>per PCM.  So the default settings work fine but changing them with the
>mixer is likely to have unpredictable effects.  
>
> - EFX capture should capture output channels 16-32 (mostly unused now)
>by default, so that we only capture the sources the user has connected
>to the multichannel recording inputs in the DSP manager.  Typically FX
>buses 0-15 would be connected directly to FX outputs 16-32 so the
>capture channels would correspond directly to the playback channels.  In
>order for this to work the default DSP configuration has to be changed
>slightly.
>
>Lee
>
>  
>
dear me, thanx a bunch - now for the stupid newbie question ... how in
heck do i apply this patch?

i'm using fedora core 2 ccrma setup (2.6.10 kernel, 1.0.8 alsa, jack
etc) but i'm guessing i'd have to forsake my apt-installed alsa and
compile it from source? that could get a bit messy, what with installed
dependencies etc ...

how should i go about this? i *so* bad want that kx-style multi-channel
support :)

shayne