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@xxxxxxxxx
> <mailto:gg3137@xxxxxxxxx>> 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@xxxxxxxxxxxxxxxxxxxxx/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
>
>
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@xxxxxxxxxxxxxxxxxxxx
http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user
