23 Oct
2009
23 Oct
'09
10:30 a.m.
Hi again,
a few more comments:
1. changes via amixer do not show up in hdspmixer. You have to rely on
your ears.
2. as I guessed before the output attenuation is only on the software
side. I looked at the hdspmixer-code, and there it is:
//--HDSPMixerWindow.cxx----------------------------------------------------------
attenuation_l =
(double)(outputs->strips[dst*2]->fader->posToInt(outputs->strips[dst*2]->fader->pos[0]))/65535.0;
attenuation_r =
(double)(outputs->strips[dst*2+1]->fader->posToInt(outputs->strips[dst*2+1]->fader->pos[0]))/65535.0;
left_val = attenuation_l* vol * (1.0 - pan);
right_val = attenuation_r* vol * pan;
muted:
snd_ctl_elem_value_set_integer(ctl, 0,
src*cards[current_card]->playbacks_offset+cards[current_card]->channel_map[idx-1]);
snd_ctl_elem_value_set_integer(ctl, 1, cards[current_card]->dest_map[dst]);
snd_ctl_elem_value_set_integer(ctl, 2, (int)left_val);
if ((err = snd_ctl_elem_write(handle, ctl)) < 0) {
//--HDSPMixerWindow.cxx----------------------------------------------------------
3. hdspmixer and amixer use the same underlying API, thus if your card
works with hdspmixer it definitely also will with amixer. It is just a
matter of finding the correct settings. I would suggest simply to try
out all possibilities (for a single software output, let's say channel
0), this is a maximum of 26 trials; with a steady software output signal
you can do that in less than a minute.
4. The factory default of ALSA drivers is to mute everything, and the
factory default for hdspmixer is a direct routing of software channels
to hardware channels (maybe with exceptions).
-- Giso
Klaus Schulz wrote:
If you read the thread at alsa-devel - you refered to
below - it seems
that more people were facing this issue already in 2003.
It seems that the whole thing works only, if you use the default
setting/routing.
Currently at factory default HDSPmixer routes the first playback
channel to the first Adat output, which happens to sit right below
playback out1/out2 in the HDSPmixer graphics.
Correspondingly this would mean I need to use an Alsaplayback channel
out 13/14 to get to AN3+4. I think it is worth a try.
But - how do I output data to alsa-playback channel 13/14 on my device 0
? I am using ecasound and alsaoutput 0,0 right now for out1/out2. No
idea how to assign 13/14.
Since the earlier discussed amixer method won't work -- at least in my
case - I am wondering if the commands
/usr/bin/amixer -c 0 cset numid=5 26,2,32768
/usr/bin/amixer -c 0 cset numid=5 27,3,32768
reflect the HDSP logic:
The logic in HDSPmixer to setup the route from Alsa Out1/Out2 to my
AN3+4 output:
1. Assign playback channels Out1/Out2 to AN3+4 output channels. This
will automatically mute playback channels out1/out2 . Now you need to
unmute the playback channels.
Only this action will activate the new outputs. ( This you'll see if you
look in the HW-output selection field)
The output channels AN3+AN4 themselves run at 100% or 0db at factory
default. No need to touch them. However would be nice to know how to
control these with amixer.
(Note: What's strange though playback output 15/16 is muted by default!?!)
Were not yet ready:
To get rid of a connection e.g. the default Adat routing, since now
traffic is routed to AN3+4 AND Adat A1+2 you:
2. Select the Adat channels A1/A2 on playback channel Out1/Out2. Now you
mute and then unmute manually playback channels Out1/Out2. Without this
manual action nothing will change.
No idea how to do that with amixer.
The mute/unmute of playback outputs seem to play fundamental role in the
routing process. Perhaps something in this area goes wrong. Again: Is
there a way to explicitely control channel
volume with amixer?
( Beside that I figured that "mute" on MASTER and Playback Outs, does
not work at all with HDSPmixer)
Cheers
\Klaus
On Fri, Oct 23, 2009 at 9:16 AM, Giso Grimm <gg3137(a)vegri.net
<mailto:gg3137@vegri.net>> wrote:
Hi Klaus,
I hope you don't mind that I copy the reply to LAU (there are a lot RME
hdsp-users who probably know more about it that I do).
Klaus Schulz wrote:
1. Could you please share your .asoundrc. I guess
it is the .asoundrc
where you set "DSP".
It is not defined in the .asoundrc, but on ALSA itself: Try "cat
/proc/asound/cards" and find out the ALSA-name of your card, e.g. here
on my office PC it is:
0 [PAD ]: Digi96 - RME Digi96/8 PAD
RME Digi96/8 PAD at 0xfd000000, irq 21
1 [DSP ]: H-DSP - Hammerfall DSP
RME Hammerfall HDSP 9632 at 0xfeae0000, irq 21
2 [ICH5 ]: ICH4 - Intel ICH5
Intel ICH5 with AD1985 at irq 17
(the hdsp9652 is at home).
2. 32768=0db refers to the playback controls,
They run at 0db@32768
and +6db@64k
However this seems to be somewhat different on the ouptuts.
0db=max
position. I'd guess this should be 64k by
default.
Question: What do I actually control with below commands? I guess
the playback-inputs only. How do I control the output channels?
Probably I need to set it to 0db to.
The mixer in the hdsp cards can control routing from hardware inputs to
hardware outputs (for 'zero' latency monitoring) and the software output
to hardware outputs.
Hardware inputs are channels 0-25, software outputs 26-51, hardware
outputs 0-25. I don't think you can control the playback attenuation
with amixer. It might be (but that is a guess) that hdspmixer attenuates
the output channels only in software (by adding the output attenuation
to all relevant input/playback channels); When controlling only via
amixer I do not have to configure the output attenuation.
The channel mapping is also discussed here:
http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg06019.html
3. I tried without success:
/usr/bin/amixer -c 0 cset numid=5 26,2,32768 #first
playbackchannel to
first output of analog extension board (ch2)
/usr/bin/amixer -c 0 cset numid=5 27,3,32768 #first
playbackchannel to
first output of analog extension board (ch3)
The channel numbering starts with zero. I do not know if the extension
board uses the same channels as ADAT out, maybe try also channels 8-15.
Cheers!
Giso