The csound messages you are getting are correct. There is only csound-out 
because
you are using -odac only and not -iadc
OK, this is what pulseaudio -v tells me when I run csound (FC9).Messages are 
from
the moment I start pulseaudio.
I: main.c: PolicyKit refuses acquire-high-priority privilige.

I: main.c: We're in the group 'pulse-rt', allowing real-time and 
high-priority scheduling.

I: core-util.c: Successfully gained nice level -11.

W: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted

I: main.c: This is PulseAudio 0.9.10

I: main.c: Page size is 4096 bytes

I: main.c: Fresh high-resolution timers available! Bon appetit!

I: module-hal-detect.c: Trying capability alsa

I: module-alsa-sink.c: Successfully opened device front:0.

I: module-alsa-sink.c: Successfully enabled mmap() mode.

ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0

I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory

I: alsa-util.c: Successfully attached to mixer 'hw:0'

I: alsa-util.c: Using mixer control "Master".

I: sink.c: Created sink 0 
"alsa_output.pci_8086_24c5_sound_card_0_alsa_playback_0" with sample spec 
"s16le 2ch 44100Hz"

I: source.c: Created source 0 
"alsa_output.pci_8086_24c5_sound_card_0_alsa_playback_0.monitor" with sample 
spec "s16le 2ch 44100Hz"

I: module-alsa-sink.c: Using 4 fragments of size 4408 bytes.

I: alsa-util.c: All 2 channels can be mapped to mixer channels. Using 
hardware volume control.

I: module-alsa-sink.c: Starting playback.

I: module.c: Loaded "module-alsa-sink" (index: #0; argument: "device_id=0 
sink_name=alsa_output.pci_8086_24c5_sound_card_0_alsa_playback_0").

I: module-alsa-source.c: Successfully opened device front:0.

I: module-alsa-source.c: Successfully enabled mmap() mode.

ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0

I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory

I: alsa-util.c: Successfully attached to mixer 'hw:0'

I: alsa-util.c: Using mixer control "Capture".

I: source.c: Created source 1 
"alsa_input.pci_8086_24c5_sound_card_0_alsa_capture_0" with sample spec 
"s16le 2ch 44100Hz"

I: module-alsa-source.c: Using 4 fragments of size 4408 bytes.

I: alsa-util.c: All 2 channels can be mapped to mixer channels. Using 
hardware volume control.

I: module.c: Loaded "module-alsa-source" (index: #1; argument: "device_id=0 
source_name=alsa_input.pci_8086_24c5_sound_card_0_alsa_capture_0").

I: module-hal-detect.c: Loaded 2 modules.

I: module.c: Loaded "module-hal-detect" (index: #2; argument: "").

I: module.c: Loaded "module-esound-protocol-unix" (index: #3; argument: "").

I: protocol-native.c: loading cookie from disk.

I: module.c: Loaded "module-native-protocol-unix" (index: #4; argument: "").

I: module.c: Loaded "module-volume-restore" (index: #5; argument: "").

I: module.c: Loaded "module-rescue-streams" (index: #6; argument: "").

I: module.c: Loaded "module-x11-bell" (index: #7; argument: 
"sample=x11-bell").

I: module.c: Loaded "module-x11-publish" (index: #8; argument: "").

I: main.c: Daemon startup complete.

I: client.c: Created 0 "Native client (UNIX socket client)"

I: protocol-native.c: Got credentials: uid=500 gid=500 success=1

I: protocol-native.c: Enabled SHM for new connection

I: client.c: Client 0 changed name from "Native client (UNIX socket client)" 
to "csound"

I: module-volume-restore.c: Restoring sink for 


I: module-volume-restore.c: Restoring volume for 


I: resampler.c: Forcing resampler 'copy', because of fixed, identical sample 
rates.

I: resampler.c: Using resampler 'copy'

I: resampler.c: Using float32le as working format.

I: sink-input.c: Created input 0 "csound-out" on 
alsa_output.pci_8086_24c5_sound_card_0_alsa_playback_0 with sample spec 
float32le 1ch 44100Hz and channel map mono

I: sink-input.c: Freeing output 0 "csound-out"

I: client.c: Freed 0 "csound"

I: protocol-native.c: connection died.


You can get loads more diagnostics if you run with --log-level=debug

Victor
----- Original Message ----- 
From: "cochedelaferte" 
To: 
Sent: Tuesday, November 25, 2008 1:37 AM
Subject: [Csnd] Pulseaudio realtime


>
> Hi Victor,
>
>
> Now investigating about why i don't get rt render on Pulseaudio (Solaris,
> last rtpulse change) except a little parasite for am.csd (fact which mean
> that any hope is possible), can you please provide to me some lights on
> several questions :
>
>
> 1) If i good understand, i should see at the run the launch message
> "PulseAudio client RT IO module..." ? Even using -m135, i don't see it. 
> Can
> that mean that something is wrong is my build ?
>
> 2) As related messages in my Csound terminal, i see exactly that :
>
>   rtaudio: pulseaudio module enabled
>   audio buffered in 1024 sample-frame blocks
>   PulseAudio output server: default
>   writing 2048-byte blks of shorts to dac
>
>   Is that fully OK ?
>
>
> 3) In my Pulseaudio terminal, i have that :
>
>   I: created 0 "Native client (UNIX socket client)"
>   I: client 0 changed name from "Native client (UNIX socket client)" to
> "csound"
>   I: created 0 "csound-out" on solaris_output with sample spec float32le
> 1ch 44100Hz
>   I: Creating new entry for 
>
>   But nothing about "csound-in". And no error messages. Only, Orch stops
> after some lines if i requires the "-o dac".
>
>
> I also asked the Pulseaudio list, but, assuming that the rtpulse module
> works without problem for most users (linux)... My hope is that comes a
> time, because now Pulseaudio is really knowed, by example on Fedora as i
> readed.
>
>
> Cheers,
>
>
> Sergio
>
> Sergio
>
>
>
>
>
>
>
>
>
>
> -- 
> View this message in context: 
> http://www.nabble.com/Pulseaudio-realtime-tp20673671p20673671.html
> Sent from the Csound - General mailing list archive at Nabble.com.
>
>
>
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