Re: [pulseaudio-discuss] No startup detection with module-udev-detect in PulseAudio 0.9.18
On Fri, Oct 02, 2009 at 03:58:50AM +0100, Matthew W. S. Bell wrote: Hi, I recently upgraded to PulseAudio 0.9.18 on a Debian testing/unstable mix. On starting pulseaudio, no ALSA cards were detected by pulseaudio, though there are two present. With debug level information being output, pulseaudio stated only: I: module-udev-detect.c: Found 0 cards. I have talked to a couple of other people who seem to be having this problem. Please note that the pulseaudio mailing list isn't the Debian bugtracking system :) Everyone i've seen using Debian with this problem were either running a too old kernel or a too old udev (latest pulse packages have their depends bumped). Judging from the discussion on this list, your udev is new enough, which leads to the question which kernel are you running? Sjoerd -- Fundamentally, there may be no basis for anything. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] No startup detection with module-udev-detect in PulseAudio 0.9.18
On Mon, 2009-10-05 at 10:00 +0100, Sjoerd Simons wrote: On Fri, Oct 02, 2009 at 03:58:50AM +0100, Matthew W. S. Bell wrote: Hi, I recently upgraded to PulseAudio 0.9.18 on a Debian testing/unstable mix. On starting pulseaudio, no ALSA cards were detected by pulseaudio, though there are two present. With debug level information being output, pulseaudio stated only: I: module-udev-detect.c: Found 0 cards. I have talked to a couple of other people who seem to be having this problem. Please note that the pulseaudio mailing list isn't the Debian bugtracking system :) Noted. Please note that I'm not reporting a bug in Debian? Everyone i've seen using Debian with this problem were either running a too old kernel or a too old udev (latest pulse packages have their depends bumped). Judging from the discussion on this list, your udev is new enough, which leads to the question which kernel are you running? 2.6.31 Matthew W.S. Bell ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] passthrough audio (eg. AC3 / DTS / WM9)
On Mon, 05.10.09 05:11, Sean McNamara (smc...@gmail.com) wrote: I would surmise that PulseAudio is probably not for you if you want to do this. There is no optional mixing in PulseAudio; everything is mixed in software. I don't even know if PA has a way to tell client applications sorry, but the soundcard is being used excuslively by another PA client right now. Uh, not sure why this would be desirable, but if it is, it is very easy to write a module for this, probably not more than 20 lines or so. Unless we had some way to do software mixing on other encodings like AC3 and WM9 (we'd have to code up special cases of the mixing logic for _each_ supported format), supporting arbitrary audio data formats would be eliminating the usefulness of 99% of PA's existing modules and features, which expect PCM samples. Uh, not sure where you came up with 99%. Yes we currently only deal with PCM. But the parts dealing with PCM are limited nonetheless and the APIs can be extended to support encoded audio, too. As a matter of fact, it's relatively simple to write a daemon that accepts UNIX socket or TCP connections, reads audio data supported by a hardware decoder from the socket, and passes the data to ALSA. At least, I think it would be easier to develop such a daemon from scratch than it would be to modify PA to support what you're asking for. Uh. That is bogus. Encoded audio is in many ways just another type of audio. And hence it should be handled the same was as we already handle PCM. In fact, codec support in PA will come. Probably even in the near feature, as I started to hack on that on my train ride back from the BlueZ summit. In fact, at the BlueZ summit it became very clear that having codec support matters. We need it for SPDIF, we need it for HDMI, we need it for Bluetooth A2DP, we need it for UPnP AV/DLNA, we need it for embedded hw where there is specific sound hw that natively speaks codecs. So, it is not a question whether this will come, it's more a question when. The complexity in adding this stems from the fact that deducing timing information fromt the codec is not trivial. In fact the lower layers (alsa) don't really handle that properly either and for the PA case this becomes even more complex since we rely much more on that. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] passthrough audio (eg. AC3 / DTS / WM9)
On Mon, 05.10.09 14:30, Dave Moore (davewantsmo...@gmail.com) wrote: Hi Everyone. I'm really sorry if this has been covered before, I've read back through the past 6 months list history and found nothing and my OS (ubuntu) and application support forums/lists are not being very helpful. I would like to pass digital audio completely unaltered (bit-stream) to my SPDIF output - Can this be done with pulseaudio? No. Not at this time. (See other mails in this thread for what is planned) If you need ac3 pass-thru then you need to bypass PA. Just make sure you are not using the SPDIF port for PA (use g-v-c or pavucontrol and make sure the sound card is notconfigured for any of the 'digital iec985' modes). Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Fri, 02.10.09 18:30, Peter Onion (peter.on...@btinternet.com) wrote: It seems a bit fussy about how much data you write with each pa_stream_write, but it might be my code that is generating the samples that is wrong! It's up to you how much your write. However, it is definitely a good idea to write at least what has been requested (or is reqturned by pa_stream_get_writable_size()). If you write less you won't get another request callback for that, so you might experience drop outs. You may also write more, which often makes a lot of sense, i.e. if your app works with a different block size. If you write more than requested it will be buffered on the server side. Summary: writing less: seldomly makes sense writing exactly what requested: makes sense if your app's block size is 1 sample writing more: makes sense if your app's block size is 1 sample. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Fri, 02.10.09 21:24, Peter Onion (peter.on...@btinternet.com) wrote: On Fri, 2009-10-02 at 18:30 +0100, Peter Onion wrote: Next thing is to try and control the sound with some glade widgets to see it the latency is going to be a problem doing things this way. Things are looking good :-) Latency is OK and only a very occasional dropout. It was ok when pulseaudio was using ~8% of one core but it seems to have jumped up to 20%, dropouts are happening every couple of seconds and these messages are appearing in /var/log/messages Oct 2 21:20:47 NewHP pulseaudio[4492]: module-alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Most likely this is an ALSA driver bug. Please report this issue to the PulseAudio developers. What do you need to know ? How did you configure the buffer_attr struct? did you set it at all? Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Sun, 04.10.09 09:34, Peter Onion (peter.on...@btinternet.com) wrote: On Fri, 2009-10-02 at 23:40 +0100, Colin Guthrie wrote: 'Twas brillig, and Peter Onion at 02/10/09 21:24 did gyre and gimble: It was ok when pulseaudio was using ~8% of one core but it seems to have jumped up to 20%, dropouts are happening every couple of seconds and these messages are appearing in /var/log/messages Oct 2 21:20:47 NewHP pulseaudio[4492]: module-alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Most likely this is an ALSA driver bug. Please report this issue to the PulseAudio developers. IIRC that message was changed in 0.9.15... So I suspect you're using a pretty old PA. It's a 0.9.14 on a Fedora 10 machine. Are there any newer rpm's about for F10 X86_64 ? I think the current pulse releases need a newer libtool than is current with F10 ? This might just be a good enough reason for me to jump up to F11. Uh. F10. That's quite old. PA is very much in flux, I'd not suggest users using old versions like that. If you ask me, especially developers should live on the bleeding edge, so if you don't want to go all the way to rawhide (what I'd recommend), then at least make sure to run the latest released version. I know that some peope think it is a good idea to run distros in releases that are a year old or two, under the assumption they'd be better tested and more stable. That assumption is wrong. They are bitrotten and developers tend to fix bugs much more frequently in newer software. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Sun, 04.10.09 15:44, Peter Onion (peter.on...@btinternet.com) wrote: So, what are the steps to upgrade my pulseaudio to the latest version ? I assume I can't just remove the pulseaudio rpm as that will break dependencies ? PA is pretty tightly integrated into the system. Consider it part of the the OS itself. So it is only feasible to update the entire OS or nothing at all. Upgrading PA alone would also require you to update the kernel, udev and other things. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] passthrough audio (eg. AC3 / DTS / WM9)
If you need ac3 pass-thru then you need to bypass PA. Just make sure you are not using the SPDIF port for PA (use g-v-c or pavucontrol and make sure the sound card is notconfigured for any of the 'digital iec985' modes). If you use AC3 pass-thru, what you are really sending over the SPDIF interface is a PCM stream. The AC3 content is formatted to follow the recommendations of IEC 61937: the compressed content uses 16 of the 24-bit words, there are additional headers and lots of padding with zeroes. In terms of timing, this is really PCM. You may need to set a number of control bits in the PCM interface but nothing serious. As long as you don't mix and don't modify the bitstream, you could handle pass-thru w/ PulseAudio without major modifications. Indeed things are completely different when the content is not reformatted to meet the PCM bitrate required by SPDIF (BT, MP3, etc). Cheers - Pierre ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] redirecting audio from SAA7134 tv card
On Fri, 02.10.09 14:54, Luca DELLA GHEZZA (luca_...@hotmail.it) wrote: On vie, 2009-10-02 at 10:11 +0200, Lennart Poettering wrote: On Thu, 01.10.09 12:38, Luca DELLA GHEZZA (luca_...@hotmail.it) wrote: I can ear the tv audio from the standard output. That is fantastic, because I tried as well with the BT headset and it works. Sadly the audio has a very poor quality: 1) the audio is pulsing, it seems that it pump up and down the master volume Uh? You say the volume changes all the time? That is weird. Maybe this is due to reception quality? it is not correct to say that the volume changes, it is more correct to say that that the volume pump up and down always at the same levels and with the same time interval. Uh? What's the difference between volume changing and volume pumping? What do you mean by pumping? Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Setting latency of module-loopback
On Fri, 02.10.09 08:20, pl bossart (bossart.nos...@gmail.com) wrote: I checked the source code, and latency_msec is missing from the list of valid module arguments. Attaching a patch to add it. Yes. this is a known issue, I provided a fix last month. Lennart, you want to apply this patch... Oops. You patch is still in my queue. There were some issues I didn't perfectly like, but it was not straightforward to fix them. So lazy as I am I just added your patch to my queue, so that I look into it later. Sorry. I have now commited Tor-Björn's patch as it was quite trivial. Pierre-Louis, I'll look into your full patch soon. Promised. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Mon, 2009-10-05 at 23:04 +0200, Lennart Poettering wrote: On Sun, 04.10.09 15:44, Peter Onion (peter.on...@btinternet.com) wrote: So, what are the steps to upgrade my pulseaudio to the latest version ? I assume I can't just remove the pulseaudio rpm as that will break dependencies ? PA is pretty tightly integrated into the system. Consider it part of the the OS itself. So it is only feasible to update the entire OS or nothing at all. Upgrading PA alone would also require you to update the kernel, udev and other things. Lennart I didn't read this until I had sent my previous reply. Are you saying it's actually impossible to upgrade pulseaudio to anything newer than the current version in Fedora 11 ? This seems to make it hard for people trying to write real applications to get any support as they can't use the latest released versions. PeterO ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Mon, 05.10.09 22:38, Peter Onion (peter.on...@btinternet.com) wrote: Upgrading PA alone would also require you to update the kernel, udev and other things. Lennart I didn't read this until I had sent my previous reply. Are you saying it's actually impossible to upgrade pulseaudio to anything newer than the current version in Fedora 11 ? Not sure if impossible is the right word. But it's certainly not easy. And not recommended either. It's a job for a distributor, not a user. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On 10/06/2009 08:30 AM, Peter Onion wrote: On Mon, 2009-10-05 at 23:01 +0200, Lennart Poettering wrote: Uh. F10. That's quite old. PA is very much in flux, I'd not suggest users using old versions like that. If you ask me, especially developers should live on the bleeding edge, But that doesn't make sense if you are developing real applications. Real users are seldom on the edge and if your code has to work for them being a bit behind the curve makes more sense I think. Anyway I'm now on F11 . so if you don't want to go all the way to rawhide (what I'd recommend), then at least make sure to run the latest released version. I'm just building 0.9.19, but it's failed on the version of sndfile. Requested 'sndfile= 1.0.20' but version of sndfile is 1.0.17 If I'm going to have to upgrade lots of other packages I'm afraid it's a show stopper for my attempt to integrate with pulseaudio properly. I've got people lined up to test my applications and I can't expect them to manually upgrade their machines if the current versions of packages in their chosen distribution are too old. Fair point. For that issues above I just compiled libsndfile from source. I think it was the only dep that had changed from the standard Fedora packages. Cheers. Patrick Shirkey Boost Hardware Ltd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio, Ubuntu 9.10 Karmic, SigmaTel STAC9200
On Tue, 29.09.09 19:49, Kyle Mallory (kyle.mall...@utah.edu) wrote: I'm experiencing an odd problem with PulseAudio on Ubuntu Karmic 9.10-a6. I wasn't quite sure what was happening, but I made a video that hopefully better illustrates the problem. I'm hopeful that someone can point me to a simple solution, or recommend a bug-fix. There isn't any audio, which is fine, but in particular take note of what PulseAudio is doing to the ALSA mixer. http://www.vimeo.com/6825531 This is the problem I'm having on my laptop, with Karmic and PulseAudio on a SigmaTel STAC9200 audio chipset (Dell Precision M90). When using the volume keys on the front of the laptop, the volume levels on the ALSA mixer adjust the Master first, and then after muting Master, subsequently adjust the PCM and LFE levels. Yes, PA controls the entire pipeline from the PCM stream to your hw. We start with the 'outermost' mixer element, and set it to the dB value that is nearest but larger than the volume we want to configure the hw to. Then we go on to the next mixer element and set try to configure the remaining volume value, all the way to the last element. Normally that means that Master does the biggest volume adjustment, while PCM is set to some value next to 0dB. This should be ideal to get the best range and granularity of software control from the hw. On my laptop, this results in very little volume control, and an overly bassy response as the LFE is maintained at full volume, while the master levels are dropped. Near the bottom of the volume range on the Notify window, the master volume is muted. Eventually the PCM and LFE levels are lowered through the last few 'presses' of the volume keys, and finally muted. Any help, suggestions, or ideas would be appreciated. I use my laptop for video editing, and a fubared audio mixer is really a deal-breaker. IIUTC then you have one of those laptops which are equipped with an extra LFE speaker which signal is synthesized in hw from the stereo signal we output? We don't fully cover that special hw right now, sorry. And I am till now sure how we could ever cover that proplery. There have been requests in the past that the LFE should always be kept in sync with our volume. Which is what we do right now. Previously we were always setting it to -inf dB. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On 10/06/2009 08:48 AM, Lennart Poettering wrote: On Mon, 05.10.09 22:38, Peter Onion (peter.on...@btinternet.com) wrote: Upgrading PA alone would also require you to update the kernel, udev and other things. Lennart I didn't read this until I had sent my previous reply. Are you saying it's actually impossible to upgrade pulseaudio to anything newer than the current version in Fedora 11 ? Not sure if impossible is the right word. But it's certainly not easy. And not recommended either. It's a job for a distributor, not a user. I can't tell if you are referring to only users here or if you are saying it is not easy for anyone including developers? Either way the situation could be improved for genuinely interested developers by providing the complete build instructions which are currently missing from the docs online and in the package. On my F11 I have libsndfile-1.20 installed manually with ./configure --libdir=/usr/lib64; make; make install I then compile PA with ./configure --libdir=/usr/lib64; make; make install I would certainly appreciate someone providing explicit steps for building the 32 bit chroot on f11. It doesn't sound particularly hard but I haven't got round to figuring out the exact steps myself yet. Everything else on F11 meets the dep requirements for building the source. Cheer.s Patrick Shirkey Boost Hardware Ltd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Tue, 06.10.09 06:49, Ng Oon-Ee (ngoo...@gmail.com) wrote: On Tue, 2009-10-06 at 00:37 +0200, Lennart Poettering wrote: If you are a user then you should use tha PA version that is shipped with your distro. If you want a newer version, then upgrade your distro. If you are a developer who writes third party apps then you should stick to a released distro, too. But of course you should really make sure to run the latest one. Sorry to interrupt, but it seems to me developers have the (unfortunate?) necessity of their software working with the latest 'stable' version of a distro, which would necessitate (especially in extreme cases like debian) that they use a kernel, udev, and thus pulse many versions behind? What do you expect? You cannot have it both ways: have the newest code and have it totally stable. That is impossible. And that Debian is an extreme case is certainly true. But take that criticism to the Debian developers. It is not particularly surprising that most developers work on other distros that have a much faster development cycle than Debian, or if the do choose Debian then they stick to the 'testing' or even 'unstable' distribution. Using Debian 'stable' for developing appears very unwordly to me. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On 10/06/2009 09:49 AM, Ng Oon-Ee wrote: On Tue, 2009-10-06 at 00:37 +0200, Lennart Poettering wrote: If you are a user then you should use tha PA version that is shipped with your distro. If you want a newer version, then upgrade your distro. If you are a developer who writes third party apps then you should stick to a released distro, too. But of course you should really make sure to run the latest one. Sorry to interrupt, but it seems to me developers have the (unfortunate?) necessity of their software working with the latest 'stable' version of a distro, which would necessitate (especially in extreme cases like debian) that they use a kernel, udev, and thus pulse many versions behind? There is also the murky middle area. Testers, advanced users and developers who would like to have an fully integrated system and are prepared to wrangle with yum and overwrite /usr/lib64 as it often is easier to work with than building into /usr/local which creates a whole new set of problems to overcome. Uninstalling PA from the packaging system completely can lead to a heap of manual installation and env var tweaking of the required deps when installing into /usr/local. So while you are correct that best practice is to install in /usr/local it is also necessary in some cases to be able to build and install over existing packages to get the fastest return on time invested in corralling the system to a workable state. it also solves the hassle of occasionally getting double ups when continuing to use the packaging system that bring in dep updates you have forgotten about when doing manual installs. Therefore preventing the ensuing headaches. Alternatively there could be a more explicit set of steps provided for running PA from the build dir recommended specifically for devs/interested parties to work with... Patrick Shirkey Boost Hardware Ltd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] redirecting audio from SAA7134 tv card
On lun, 2009-10-05 at 23:08 +0200, Lennart Poettering wrote: On Fri, 02.10.09 15:09, Luca DELLA GHEZZA (luca_...@hotmail.it) wrote: On vie, 2009-10-02 at 18:30 +0300, Tanu Kaskinen wrote: to, 2009-10-01 kello 12:38 -0500, Luca DELLA GHEZZA kirjoitti: ok Tanu, first of all thx a lot for your reply. I used pactl list to localize the correct input device, as well for the output, but I'm not shure is the right one. this is the output device I choose: Sink #0 State: SUSPENDED Name: alsa_output.pci-_00_07.0.analog-stereo Description: Internal Audio Analog Stereo snip If the internal sound card is the card from which you want to hear the tv card, then it's the right one. As Lennart said, you can actually use pavucontrol to move the loopback stream wherever you want to. I didn't understand how to do this, but this has a minor importance, I definitevely want that the tv audio goes trough the standard output, just that. With a recent version of pavucontrol you can just use the menu on the loopback steram and move it to another device, during runtime. It's really not that hard. I have Pavucontrol 0.9.8+git20090701 (from ubuntu repositories), but there I don't find the loopback stream menu. Probably is not new enough. where CARD is the tv card name given by alsa. It can be found in /proc/asound/cards. For example, I have this in my /proc/asound/cards: 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xfe024000 irq 16 The stuff in the square brackets is the card name, so in my case I have a sound card with name SB. well, just to try everything I tried this solutions too, I added the line to the file /etc/pulse/default.pa, in this moment I can't reboot the system so I used sudo /etc/init.d/pulseaudo force_reload. But nothing has changed. PA is not a system service (unless configured in a non-recommended way). If your distribution configures it as such, ask them to stop that crack. PA when run as user service (which is recommend) canbe restarted by issuing 'pulseaudio -k' and then restarting it via 'start-pulseaudio-x11'. I understand what you mean, no in Ubuntu Pulseaudio is not a system service, is a user service, that is the answer I received trying pulseaudio restart in that way, but just now I understand what does it mean. Anyway, I understand well that there is no solution to my problem? In other words: SAA7134-alsa module is too much crappy so is I want to watch and listen tv I have to find out a tv card with another chip? If yes, which one? Thx so much for your patience. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Example using async API
On Tue, 06.10.09 09:59, Patrick Shirkey (pshir...@boosthardware.com) wrote: Alternatively there could be a more explicit set of steps provided for running PA from the build dir recommended specifically for devs/interested parties to work with... As mentioned: $ ./bootstrap.sh $ make -j4 $ cd src $ ./pulseaudio And of course one should be able to read the output of configure and be able to install the missing deps. All the lines above should be pretty standard and are basically the same for every Linux package. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] redirecting audio from SAA7134 tv card
On Mon, 05.10.09 18:02, Luca DELLA GHEZZA (luca_...@hotmail.it) wrote: I didn't understand how to do this, but this has a minor importance, I definitevely want that the tv audio goes trough the standard output, just that. With a recent version of pavucontrol you can just use the menu on the loopback steram and move it to another device, during runtime. It's really not that hard. I have Pavucontrol 0.9.8+git20090701 (from ubuntu repositories), but there I don't find the loopback stream menu. Probably is not new enough. There is no menu that would be in any way specific to loopback streams. Just find the loopback stream, use the normal stream menu and move it to another device, it isn't that hard. Anyway, I understand well that there is no solution to my problem? In other words: SAA7134-alsa module is too much crappy so is I want to watch and listen tv I have to find out a tv card with another chip? If yes, which one? Thx so much for your patience. I guess what is true for SAA7134 is true for the other TV cards, too: since there is no commercial interest pushing Linux support the drivers are limited. Also, I don't have a TV, no TV antenna, no TV cable, so this is really not among the things I regularly test. Sorry. Lennart -- Lennart PoetteringRed Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4 ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
