Re: [LAU] RME HDSP 9632 - amixer control - Linux-audio-user

Klaus Schulz

23 Oct 23 Oct

6:20 a.m.

New subject: RME HDSP 9632 - amixer control

Hi Giso and others. 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> 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

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Giso Grimm

8:30 a.m.

New subject: RME HDSP 9632 - amixer control

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

Reply

Klaus Schulz

24 Oct 24 Oct

10:46 a.m.

New subject: RME HDSP 9632 - amixer control

Hi Giso and others. I managed to get a basic routing working! I wrote a script which initializes the entire matrix, connects one after another every input to every output and plays a sound per setting. I identified my alsainputs on 16/17 and analog outputs on 12/13 for my analog extension board (out1 out2). This has not very much to do with the channel assignment I've seen everywhere. With the script I can easily scan my setup now. Finally I need to find out how to get more then 2 channels from Alsa to the same card without using Jack. The pity is really that I can not control the attenuation on the hw-outputs with amixer. I'll do that on the application side. THX for all the hints so far. Cheers \Klaus On Fri, Oct 23, 2009 at 9:16 AM, Giso Grimm <gg3137(a)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

Reply

Klaus Schulz

25 Oct 25 Oct

9:14 a.m.

New subject: RME HDSP 9632 - amixer control

Hi Paul. OK - got it. I mean at least from a user perspective, by looking at HDSPmixer, the HW-outputs seemed like single controls. What's confusing on the first glance is that these cover a different gain range and can be contolled independenly form the input controls. I considered the outputs the vertical and the inputs the horizantal layer of the matrix. Anyhow. Never mind. Perhaps you could explain why in HDSPmixer the input group controls have a range of -inf to +6db and the HW-outputs -inf to 0db. THX Klaus On Sun, Oct 25, 2009 at 2:29 AM, Paul Davis <paul(a)linuxaudiosystems.com>wrote;wrote:

...

On Sat, Oct 24, 2009 at 10:46 AM, Klaus Schulz <kls.schlz(a)googlemail.com> wrote:

The pity is really that I can not control the attenuation on the

hw-outputs

with amixer. I'll do that on the application side.

there is no such control in the hardware. the matrix mixer only provides gain control for the signal level of a particular source to a particular output. the idea of "turning down channel 13" doesn't exist as an individual idea in this h/ w design - you have to turn down everything that feeds channel 13 instead.

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