Re: [pulseaudio-discuss] Conexant CX20585 recording does not work
On 2011-03-08 00:26, Colin Guthrie wrote: 'Twas brillig, and Gabriel Schulhof at 07/03/11 19:56 did gyre and gimble: Hi! My mom has a laptop with the following sound card: lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40) driver: snd_hda_intel alsamixer: Card: HDA ATI SB Chip: Conexant CX20585 I can't record from this card no matter what I try. It seems that alsamixer has a lot more options than pulseaudio: Mic B Mic C Mic E Mic F Capture<0% -- 100% slider> Analog Mic Boost<0 db -- 40 db slider with 10 db increment> Nevertheless, I will attach the output from pacmd list-sources. I have tried to record from each mic in turn, I have tried the different boost levels (0 db - 40 db), but I have had no luck picking up anything. I have attached an external mic to the line-in, but no difference. When I say "I have tried to record" I mean that I logged into her computer, chose one of the mics from the above list, and then I told my mom to say something while looking at the input sound level indicator in the GNOME sound settings panel. On other computers, I have found this a sufficient indication of whether the mic works. Is this inability to record a known limitation of this sound card? If not, is there some special setting I have to modify to make it work? TIA for your help, Some mics do this weird out-of-phase thing whereby the recording level is effectively 0 when the stereo volume is synced. If you set left to say 5% and right to say 95% then it works fine. This may be the case with your h/w too. FWIW The "extra" options you see in ALSA are generally confusing and useless. PA attempts to consolidate it and make it a lot more user friendly. Normally you'd get a list of several "port" selectors in gnome-volume-control. Sadly as your card uses the incredibly helpful names of "Mic B" through "Mic F" these do not match the various "standard" names we probe for. David, have you come across this model on your travels? Many times, that's the fallback for many Conexants. Any ideas on how we could best support it? This one is one of few that lack an auto-parser of the pin default config, so for now - model quirks. Without seeing alsa-info or something similar, I can't tell which model would fit. In the long run we would benefit from an autoparser handling these as well. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Volume algorithm
'Twas brillig, and Tarantism at 07/03/11 20:19 did gyre and gimble: >>> > > The volume system on Maemo is a bit peculiar. The system volume doesn't >>> > > touch any sink volumes (except with bluetooth in HSP/HFP mode), all >>> > > volume control is done by adjusting stream volumes. If you want to use >>> > > the same volume as eg. the music player, set the media.role property of >>> > > the sink input to "x-maemo". That's a special role for pretty much >>> > > everything else than phone and event sound streams. >> > That figures. Thanks very much! > And works nicely. I won't say how long my dbus volume monitor class > work-around took to get working ... oh well. Ahh but the lessons learned are invaluable :) We've all been there! The value cannot be measured in lines of code of the final solution :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Conexant CX20585 recording does not work
'Twas brillig, and Gabriel Schulhof at 07/03/11 19:56 did gyre and gimble: > Hi! > > My mom has a laptop with the following sound card: > > lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) > (rev 40) > > driver: snd_hda_intel > > alsamixer: > Card: HDA ATI SB > Chip: Conexant CX20585 > > I can't record from this card no matter what I try. It seems that > alsamixer has a lot more options than pulseaudio: > > Mic B > Mic C > Mic E > Mic F > Capture <0% -- 100% slider> > Analog Mic Boost <0 db -- 40 db slider with 10 db increment> > > Nevertheless, I will attach the output from pacmd list-sources. > > I have tried to record from each mic in turn, I have tried the different > boost levels (0 db - 40 db), but I have had no luck picking up anything. I > have attached an external mic to the line-in, but no difference. > > When I say "I have tried to record" I mean that I logged into her > computer, chose one of the mics from the above list, and then I told my > mom to say something while looking at the input sound level indicator in > the GNOME sound settings panel. On other computers, I have found this a > sufficient indication of whether the mic works. > > Is this inability to record a known limitation of this sound card? If not, > is there some special setting I have to modify to make it work? > > TIA for your help, Some mics do this weird out-of-phase thing whereby the recording level is effectively 0 when the stereo volume is synced. If you set left to say 5% and right to say 95% then it works fine. This may be the case with your h/w too. FWIW The "extra" options you see in ALSA are generally confusing and useless. PA attempts to consolidate it and make it a lot more user friendly. Normally you'd get a list of several "port" selectors in gnome-volume-control. Sadly as your card uses the incredibly helpful names of "Mic B" through "Mic F" these do not match the various "standard" names we probe for. David, have you come across this model on your travels? Any ideas on how we could best support it? Gabriel, if the mono trick does not help (and the model= argument to the HDA alsa module (as suggested by Dark Shadow) does not yield any results either) perhaps try using alsamixer -c0 to flip those various Mic B through F to see if any of them make things work? Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Conexant CX20585 recording does not work
On Mon, Mar 7, 2011 at 12:56 PM, Gabriel Schulhof wrote: > Hi! > > My mom has a laptop with the following sound card: > > lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) > (rev 40) > > driver: snd_hda_intel > > alsamixer: > Card: HDA ATI SB > Chip: Conexant CX20585 > > I can't record from this card no matter what I try. It seems that > alsamixer has a lot more options than pulseaudio: > > Mic B > Mic C > Mic E > Mic F > Capture <0% -- 100% slider> > Analog Mic Boost <0 db -- 40 db slider with 10 db increment> > > Nevertheless, I will attach the output from pacmd list-sources. > > I have tried to record from each mic in turn, I have tried the different > boost levels (0 db - 40 db), but I have had no luck picking up anything. I > have attached an external mic to the line-in, but no difference. > > When I say "I have tried to record" I mean that I logged into her > computer, chose one of the mics from the above list, and then I told my > mom to say something while looking at the input sound level indicator in > the GNOME sound settings panel. On other computers, I have found this a > sufficient indication of whether the mic works. > > Is this inability to record a known limitation of this sound card? If not, > is there some special setting I have to modify to make it work? > > TIA for your help, > > > > Gabriel > > > > ___ > pulseaudio-discuss mailing list > pulseaudio-discuss@mail.0pointer.de > https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss > > Have you tried loading the module with different model parameters it can make the difference (for example my laptop by default only shows 2 channel stereo and mics don't work but with the right model I get mics and all the way up to 5.1ch surround sound) Run this command to see your options and trial and error them since some times another is close enough (I have to use one for a different laptop that happens to have the same sound setup) zcat /usr/share/doc/alsa-base/driver/HD-Audio-Models.txt.gz ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Volume algorithm
On Sun, 2011-03-06 at 22:55 +, Tarantism wrote: > On Sun, 2011-03-06 at 22:48 +0200, Tanu Kaskinen wrote: > > Are you planning to use the module outside Maemo 5 (I assume you're > > talking about Maemo 5)? I ask because Pulseaudio's internal API that > > modules use isn't stable, so modules that aren't part of Pulseaudio > > itself break very likely if you try to use them with any other version > > of Pulseaudio than what they were compiled against. > Yes, I am planning to use the module outside Maemo 5. > No, I wasn't aware that this plan was doomed to inevitable failure ... > I'll have to look into that. The interfaces that I'm using (those in module-sine.c) seem pretty stable (last major API change in July '07). I can keep up with that. > > The volume system on Maemo is a bit peculiar. The system volume doesn't > > touch any sink volumes (except with bluetooth in HSP/HFP mode), all > > volume control is done by adjusting stream volumes. If you want to use > > the same volume as eg. the music player, set the media.role property of > > the sink input to "x-maemo". That's a special role for pretty much > > everything else than phone and event sound streams. > That figures. Thanks very much! And works nicely. I won't say how long my dbus volume monitor class work-around took to get working ... oh well. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Conexant CX20585 recording does not work
Hi! My mom has a laptop with the following sound card: lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40) driver: snd_hda_intel alsamixer: Card: HDA ATI SB Chip: Conexant CX20585 I can't record from this card no matter what I try. It seems that alsamixer has a lot more options than pulseaudio: Mic B Mic C Mic E Mic F Capture <0% -- 100% slider> Analog Mic Boost <0 db -- 40 db slider with 10 db increment> Nevertheless, I will attach the output from pacmd list-sources. I have tried to record from each mic in turn, I have tried the different boost levels (0 db - 40 db), but I have had no luck picking up anything. I have attached an external mic to the line-in, but no difference. When I say "I have tried to record" I mean that I logged into her computer, chose one of the mics from the above list, and then I told my mom to say something while looking at the input sound level indicator in the GNOME sound settings panel. On other computers, I have found this a sufficient indication of whether the mic works. Is this inability to record a known limitation of this sound card? If not, is there some special setting I have to modify to make it work? TIA for your help, Gabriel moms-sound-card.list-sources Description: Binary data ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] patch for module-ladspa-sink
Hello Marius, First of all I'd like to apologise profusely for not getting back to you until now. Two months is a ridiculous delay and I'm sorry about that (especially as you mention that you are new to open source - it's not the best first impression to receive!) Secondly I'd like to thank you very much indeed for your contribution. Sadly, (and this is one of the reasons for the delay), there was a pending patch that updated the LADSPA module to give it multichannel support. I'm not really an expert in the land of LADSPA, which is why it was sitting in my moderation queue for so long and also why I'm only just now getting around to contacting you! As you are clearly interested in this work and obviously have knowledge of LADSPA and (now!) some knowledge of PA, I'd very much appreciate it if you could review the code now in current git master related to ladsap. Just to be clear, I've not used any of the code you attached to your mail, so it would mean starting with a clean slate for the review. I'm am sure that there is still room for improvement in the committed code, so any patches that you are able to make to it, I promise I'll review and push in a timely manner (or give you really good excuses if not :p) For clarity the main patch is this one: http://git.0pointer.de/?p=pulseaudio.git;a=commitdiff;h=6bd32e (although the commit date is from October, I only "pushed" it into our git master a week or so ago). If you do create further patches, I'd encourage you to clone the git master repository and make your changes. Then commit the results locally and supply the patches to us via the git format-patch command. This allows us to review the changes easily (as reading a list of the differences is much easier than reading the whole file) and also allows us to properly attribute the work to yourself in our changelog. I'd also very much encourage you to come on to #pulseaudio on IRC as a lot of the devs hang out there and we're a friendly and approachable bunch (usually!). It's not always super busy, so just hang out and talk occasionally and you'll soon see folks chat about stuff (I'm coling) I've not got much to say about your other comments but with regards to what you said about volume controls and why they are separate rather than just sharing one: this has been recently addressed by Tanu (and Marc-André!) with a feature known as "Volume Sharing" :) Again, apologises for the lack of response until now and I hope you'll still be keen to contribute - it would be very much appreciated. Cheers Col 'Twas brillig, and Marius Bjørnstad at 30/12/10 20:51 did gyre and gimble: > Dear PulseAudio community members, > > I am new to open source, but I wanted to extend the functionality of > module-ladspa-sink. I have created a new version of > module-ladspa-sink.c, based on the current version in git. I don't know > if it will be useful to include it in PulseAudio, so I will present it > here. I will summarise the changes, then provide some details, then list > some problems. I don't understand a lot of PA, but I tried to change > only the things I understood. > > Summary: > The motivation was to get bs2b ( http://bs2b.sourceforge.net/ ) working > in PulseAudio. The original module has a one-to-one mapping between > instances of the LADSPA plugin and channels. The new version works in > two modes: by default it has the same behaviour as the original module. > Alternatively, the option "cpmap" can be used to specify a mapping > between the channels of the stream in :Pulse, and input/output ports of > the plugin. In this mode, only one instance of the LADSPA plugin will be > created. > > Details: > > *** Parsing of the cpmap option *** > The cpmap option accepts a comma-separated list of port-channel > mappings. Each mapping contains values separated by colons (:). There > are three alternative formats that can be used: > > 1) inputport:outputport:channel > > 2) suffix:channel > uses Input suffix and Output suffix as ports, e.g. > "Input left" and "Output left" > > 3) :channel > Uses Input and Output as ports (used for the default value > of cpmap) > > The input and output port names are matched case sensitively. The > channel names are those used with pa_channel_map. > > The mappings are separated by commas , . It is not foreseen that port > names contain colons or commas (changing the separator, or even escaping > it, could be considered). > > Channels that do not have a listed mapping are passed through the module > unchanged (except for volume, see Problems). > > The keyword "all" can be used in place of a channel, and this invokes > the behaviour of the original module. The default value for cpmap is > thus ":all". "all" can not be mixed with other channels. > > > *** Other changes *** > The new module uses one buffer per channel, where the previous module > used a single buffer. In many places, the number of channels and the > number of plugin instances is disambi
