Surround via SPDIF with ALSA/emu10k1?

Surround via SPDIF with ALSA/emu10k1?

[Thomas Zehetbauer]

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Hi,

I am desperately trying to get surround sound working here. I have
- Creative Labs SB Live! 5.1 (emu10k1) card
- digital/SPDIF/coaxial connection
- Cambridge SoundWorks DTT2500
- linux-2.6.12.5
- alsa-lib-1.0.9rf-2

Digital and analog sound basically works. I can play music on the front
speakers or headphone and I can even play DVDs in DolbyDigital 5.1 with
ac3 passthrough.

Unfortunately I could not even get the rear/center/LFE connectors of my
card to work until I discovered the following page:
http://alsa.opensrc.org/index.php?page=emu10k1

Now I can either enable the 'SB Live Analog/Digital Output Jack' switch
and use the SPDIF connector to my DD5.1 surround system OR mute this
control and access/unmute the rear/center/LFE channels. Using analog
cabling is not an option as the DTT2500 has no connector for the
center/LFE channels.

Tom

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Re: Surround via SPDIF with ALSA/emu10k1?

[Robert Hancock]

Thomas Zehetbauer wrote:
> Now I can either enable the 'SB Live Analog/Digital Output Jack' switch
> and use the SPDIF connector to my DD5.1 surround system OR mute this
> control and access/unmute the rear/center/LFE channels. Using analog
> cabling is not an option as the DTT2500 has no connector for the
> center/LFE channels.

If the speaker system has no analog input for those channels, then there 
is no way to play back content through them unless it is Dolby Digital 
encoded. (There's no way to put more than 2-channel audio through SPDIF 
unless it's encoded such as by Dolby Digital or DTS.) The sound card is 
not capable of doing this, so if the content is not Dolby Digital 
already, this would have to be handled by some software.

-- 
Robert Hancock      Saskatoon, SK, Canada
To email, remove "nospam" from hancockr@nospamshaw.ca
Home Page: http://www.roberthancock.com/

Re: Surround via SPDIF with ALSA/emu10k1?

[Thomas Zehetbauer]

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I have now been told that SPDIF cannot support more than 2 channels
except with AC3 compression. Given the fact that we can send 580MBit/s
over USB2.0 I would not have even remotely considered this to be the
problem and find it an incredible shame that audio industry is using
such a crippled standard.

I have now solved my problem by buying and connecting an analog 5.1
speaker set. Unfortunately I get audible distortions when I turn both
the "PCM" and "Wave" mixers to the maximum setting. I wonder if anyone
can provide more insight what these controls really do and whether it's
better to turn down "PCM", "Wave" or both.

Tom

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Re: Surround via SPDIF with ALSA/emu10k1?

[Lee Revell]

On Sat, 2005-08-27 at 20:18 +0200, Thomas Zehetbauer wrote:
> I have now been told that SPDIF cannot support more than 2 channels
> except with AC3 compression. Given the fact that we can send 580MBit/s
> over USB2.0 I would not have even remotely considered this to be the
> problem and find it an incredible shame that audio industry is using
> such a crippled standard.
> 
> I have now solved my problem by buying and connecting an analog 5.1
> speaker set. Unfortunately I get audible distortions when I turn both
> the "PCM" and "Wave" mixers to the maximum setting. I wonder if anyone
> can provide more insight what these controls really do and whether it's
> better to turn down "PCM", "Wave" or both.

Off topic for LKML, this should only have been posted to the ALSA list.

Lee

Re: Surround via SPDIF with ALSA/emu10k1?

[Lennart Sorensen]

On Sat, Aug 27, 2005 at 08:18:59PM +0200, Thomas Zehetbauer wrote:
> I have now been told that SPDIF cannot support more than 2 channels
> except with AC3 compression. Given the fact that we can send 580MBit/s
> over USB2.0 I would not have even remotely considered this to be the
> problem and find it an incredible shame that audio industry is using
> such a crippled standard.
> 
> I have now solved my problem by buying and connecting an analog 5.1
> speaker set. Unfortunately I get audible distortions when I turn both
> the "PCM" and "Wave" mixers to the maximum setting. I wonder if anyone
> can provide more insight what these controls really do and whether it's
> better to turn down "PCM", "Wave" or both.

I thought the limitation on SPDIF was normally that most decoders only
support 2 channels of PCM, except for a few multimedia speaker systems
which support 3 pairs of PCM streams at the same time to support digital
out on some sound cards.

Most digital decoders do support AC3 of course and sometimes also DTS
both of which of course support encoding more than 2 channels.

So really I think it is just a problem of what standards the receivers of
the digital data know what to do with.

As for volume settings, I always try to keep the sound card mixers at
around 75 to 80% since it seems most amplifiers and mixer do distort a
bit when you max them out.  Why would you want them all at 100% anyhow,
then you might as well not have mixing control for the seperate audio
channels at all.

Len Sorensen

Re: Surround via SPDIF with ALSA/emu10k1?

[Lee Revell]

On Sat, 2005-08-27 at 17:17 -0400, Lennart Sorensen wrote:
> As for volume settings, I always try to keep the sound card mixers at
> around 75 to 80% since it seems most amplifiers and mixer do distort a
> bit when you max them out.  Why would you want them all at 100%
> anyhow, then you might as well not have mixing control for the
> seperate audio channels at all. 

For the best S/N ratio and dynamic range all mixer controls SHOULD be at
100%, assuming the volume control in your driver only attenuates
signals.  This is the case for the emu10k1 which implements all mixer
controls via DSP programs that run on the soundcard anyway, and handles
overflow itself.

Think about it, if you lower the mixer controls to 75%, you're not
getting the full 16 bits of dynamic range, it's probably more like 14 or
15.  16 bits is barely enough headroom anyway, so you really don't want
this.

Anyway the problem here is a bug in the emu10k1 driver, see alsa-devel
for the resolution.

Lee

Re: Surround via SPDIF with ALSA/emu10k1?

[Lennart Sorensen]

On Sat, Aug 27, 2005 at 05:47:08PM -0400, Lee Revell wrote:
> For the best S/N ratio and dynamic range all mixer controls SHOULD be at
> 100%, assuming the volume control in your driver only attenuates
> signals.  This is the case for the emu10k1 which implements all mixer
> controls via DSP programs that run on the soundcard anyway, and handles
> overflow itself.

Well I don't want them all at 100% since I do want to be able to have
even levels between the cd, midi and wave, which isn't always the case
at 100%.  They sure sounded worse.

I certainly know some simpler sound cards would have more distrosion
when you ran the amplifiers on the card at 100%.

> Think about it, if you lower the mixer controls to 75%, you're not
> getting the full 16 bits of dynamic range, it's probably more like 14 or
> 15.  16 bits is barely enough headroom anyway, so you really don't want
> this.

Bit 16 is used as soon as the signal passes 50%.  After all it is
required when the signal goes past 32767 and is on all the way to 65535.

So at 75% depending on how things are done internally I will be either
using the full range just not amplified to the same amount, or if truly
done entirely digitally, then I will be using 75% of the range, which is
still more than even 15bit could handle (since 15 bit has half the range
of 16bit).

> Anyway the problem here is a bug in the emu10k1 driver, see alsa-devel
> for the resolution.

Well that probably helps.

Len Sorensen