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* doxygen: Fix documentation typosDavid Fries2010-11-173-7/+8
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* sink-input: Fix commentColin Guthrie2010-11-171-1/+1
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* intended-roles: Mark devices with a form factor of 'headset' as being ↵Colin Guthrie2010-10-311-1/+2
| | | | | | | appropriate for 'phone' streams This was a result a report by Patrick Ben Koetter relating to his Logitech Wireless Headset.
* alsa-mixer: add profile for Native Instruments KorecontrollerDaniel Mack2010-10-313-1/+88
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* SSE/MMX/ARM: Fix high frequency noise with unusual number of channelsDavid Henningsson2010-10-133-13/+25
| | | | | | | | In the assembly optimized versions of SSE, a noise could occur when the number of channels were 3,5,6 or 7. For MMX and ARM, this could occur when the number of channels were 3. Signed-off-by: David Henningsson <david.henningsson@canonical.com>
* augment-properties: Fix debug messages and statement bracketing.Colin Guthrie2010-10-061-2/+3
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* suspend: Do not assert when checking for device suspended status and a ↵Colin Guthrie2010-10-052-2/+16
| | | | | | | | | | | | | | | | | | | | | | | | stream is not linked. When looping through the streams on a given device checking to see if the stream is 'active' there should be no assert if the stream is not linked, it should simply be ignored. This assert can be hit if a sink and a sink input are both created and setup but the final put calls are left to the end as is done in module-ladspa-sink. While the order of the calls in module-ladspa-sink could be altered, we should deal gracefully with the way it is now and not complain about ending up in this state. A trigger case was trivial: 1. Load a ladspa-sink. 2. Play a stream and move it to it. 3. Unload the module, then reload it. 4. Due to module-stream-restore and module-suspend-on-idle, the hook callbacks will ultimately hit this assert. Thanks to Kim Therkelsen for highlighting this issue.
* device-manager: Ensure that sinks/sources populate the device manager lists ↵Colin Guthrie2010-09-291-5/+59
| | | | | | | in order of their priority. Currently the order of the sinks is simply that of their position in the idxset which is certainly not what the user would want.
* augment-properties: Search for .desktop files in subfolders too.Colin Guthrie2010-09-291-5/+39
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* stream-restore: Clear the save_sink/save_source flags on apply_entry.Colin Guthrie2010-09-291-12/+36
| | | | | | | | | | | If the user specifically removes the device element from the stream restore rule, we have to clear the save_sink/save_source flag of the stream. This means that other stream routing systems (e.g. module-device-manager) can take over routing for this stream. In order to facilitate the reapplication of other routing rules, we fire a stream change event. Arguably the stream itself has not changed, but the rules governing its routing have, so I feel this is justified.
* Add a configure option to change 'udevrulesdir'Arun Raghavan2010-09-212-1/+7
| | | | | | | | | | | This patch serves two purposes: 1) Allows something other than the de-facto standard udev rules dir or /lib/udev/rules.d to be used (the udev build system allows you to customise this) 2) Allows a prefixed, non-root install (right now, the /lib/... path is hard-coded into the build system
* x11: Use the default screen for X11 properties.Colin Guthrie2010-09-185-69/+95
| | | | | This commit restores the functionality originally included in 65e807 by Leszek Koltunski.
* xcb: xcb_get_setup() can return 0, so make sure we check it before usingColin Guthrie2010-09-181-20/+48
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* xcb: Ensure the XCB connection is valid before using it.Colin Guthrie2010-09-182-0/+10
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* alsa: Only set the 'first' flag to false when we actually call snd_pcm_start()Colin Guthrie2010-09-141-1/+2
| | | | Previously, if work_done was false, we could conceivably not call snd_pcm_start().
* alsa-mixer: add profile for Traktor Kontrol S4Daniel Mack2010-09-143-1/+84
| | | | | This new audio interface from Native Instruments has 2 stereo channels for both input and output direction. This patch adds mappings for them.
* alsa: make defines for smoother configurationWim Taymans2010-09-142-4/+10
| | | | | | | | | | | | Make new defines for the smoother window size and adjust time constants instead of reusing some unrelated constant. Increase the smoother window size even more because the bigger it is, the better. Since we have a 200ms max update interval and the max smoother history is 64 entries, 10seconds is a good default. Decrease the smoother adjust time to 1 second. The previous value of 4 seconds was too much to adapt quickly after a resume.
* alsa: work around slightly broken _delay implementationsWim Taymans2010-09-144-6/+46
| | | | | | Use snd_pcm_avail_delay() in pa_alsa_safe_delay() so that we can check the delay value against the avail value and patch it up when it looks invalid. Only do this for capture.
* alsa-time-test: make test usable for capture tooWim Taymans2010-09-141-16/+43
| | | | Extend the test to also work for capture devices.
* alsa-source: refactor smoother and device startWim Taymans2010-09-141-8/+18
| | | | | | | Move the code to start the capture and the smoother closer together to improve smoother accuracy. Rework things to look more like the alsa sink where the device is started in only one place.
* smoother: avoid losing precisionWim Taymans2010-09-141-3/+1
| | | | | Avoid losing precision by subtracting uint64 values before converting them to doubles.
* alsa: Set the rewind safeguard proportionally to sample specColin Guthrie2010-09-041-2/+3
| | | | | | | | | | | | | Currently when rewinding alsa, a fixed value of 256 bytes is used, which represents 1.33ms @ 48kHz (2ch, 16bit). This is typically fine and due to DMA constraints we would not want to rewind less than this. However with more demanding sample specs, (e.g. 8ch 192kHz 32bit) 256 bytes is likely not sufficient, so calculate what 1.33ms would be and use which ever value is bigger. Discussed with David Henningsson and Pierre-Louis Bossart here: http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/7286
* alsa-sink: Get rid of a compiler warning regarding rewind_safeguard type.Tanu Kaskinen2010-09-041-2/+2
| | | | | GCC gave a warning, because the pointer given to pa_modargs_get_value_u32() had type size_t instead of uint32_t.
* add rewind-safeguard parameterPierre-Louis Bossart2010-09-042-5/+17
| | | | | | | | | | Rewinding the ring buffer completely causes audible issues with DMAs. Previous solution didn't work with tsched=0, and used tsched_watermark for guardband, which isn't linked to hardware and could become really high if underflows occurred. Added separate parameter that can be tuned to hardware limitations and size of DMA bursts.
* alsa: don't make use of tsched related variables when tsched is disabledLennart Poettering2010-09-042-22/+40
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* alsa: increase the smoother windowWim Taymans2010-09-031-2/+2
| | | | | 40ms for the smoother window is too small. Increase the size to 4 seconds, like we do for the sinks.
* alsa: resume smoother after unsuspendWim Taymans2010-08-231-1/+1
| | | | | | The smoother is paused when the device is suspended but never resumed on unsuspend. Pass the paused = FALSE flag to the pa_smoother_reset() call to make it unpause when unsuspending. This patch improves source timings quite a bit.
* x11: Partially convert to XCB.Colin Guthrie2010-08-158-90/+109
| | | | | | This commit mostly converts the X11 handling to XCB. There are still some uses of XLib to deal with the X11 session handling modules, however all client-side code should now be free of XLib and thus this should fix Bug #799
* udev: fix hex decodingLennart Poettering2010-08-151-2/+2
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* alsa: disable rewinds when using ALSA pluginsPierre-Louis Bossart2010-07-151-2/+12
| | | | | | | | | | This is required to when playing on a52: device, rewind is broken in those plugins. Credits to Michael Rans <mcarans@yahoo.co.uk> for finding this workaround, and Tanu Kaskinen <tanuk@iki.fi> for providing valuable feedback. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
* alsa: Fix assertion on mmap_write (triggered via a52 plugin)Colin Guthrie2010-07-151-0/+3
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* rtp: Fix bracketing in pa_rtp_recv.Colin Guthrie2010-05-081-2/+2
| | | | | | | | | | The syntactically correct error meant that the timestamp was always marked as found and only the first header was checked. In the case where the timestamp was the first header, things would have worked as expected. Thanks to pino for reporting via bug refs #818
* jack: Prevent crash on jack server shutdownDavid Henningsson2010-05-082-6/+7
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* thread: name all threads so that the names appear in /proc/$PID/task/$TID/commLennart Poettering2010-05-0323-28/+80
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* Revert "Fix crash on jack server shutdown"Lennart Poettering2010-03-221-3/+3
| | | | | | This reverts commit e756467434f787d392c45aac1327768bcf0b610c. The patch is incomplete and not entirely correct.
* socket-client: properly handle asyncns failuresJez Austin2010-03-221-1/+2
| | | | | | | | | We fail to detect when people disable IPv6 in there kernels. This patch makes sure we don't ignore this error condition. http://pulseaudio.org/ticket/752 https://bugzilla.mozilla.org/show_bug.cgi?id=533470 http://bugs.freedesktop.org/show_bug.cgi?id=25742
* intended-roles: Do not pick monitor sources when doing automatic role-based ↵Colin Guthrie2010-03-171-1/+12
| | | | device selection
* Fix crash on jack server shutdownDavid Henningsson2010-03-151-3/+3
| | | | | | | On sink unlinking, existing sink inputs are moved, which in turn calls a get latency callback, which references the jack client. Therefore, make sure the sink is unlinked before the client is closed. Failure to do so might lead to SIGSEGV.
* Handle 'Digital Mic' as an 'Input Source'Daniel T Chen2010-03-121-0/+4
| | | | | | | BugLink: https://launchpad.net/bugs/533877 Some laptops have 'Digital Mic' exposed as an 'Input Source', e.g., Dell XPS 1330, so handle these, too.
* core-util: ensure that we chmod only the dir we ourselves createdKees Cook2010-03-042-6/+35
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* core: rework how stream volumes affect sink volumesLennart Poettering2010-02-231-1/+41
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* esd,simple: use pa_memblockq_pop_missing()Lennart Poettering2010-02-232-2/+2
| | | | | | | | | | We need to use pa_memblockq_pop_missing() for all request handling, including the initial request, because otherwise the counters will be stay off during the entire runtime. This should fix: https://bugzilla.redhat.com/show_bug.cgi?id=559467
* core-util: introduce generic function pa_strip()Lennart Poettering2010-02-222-4/+25
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* alsa: reset max_rewind/max_request while suspendingLennart Poettering2010-02-221-1/+10
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* alsa-util: strip spaces from ALSA card/pcm namesLennart Poettering2010-02-221-5/+7
| | | | http://pulseaudio.org/ticket/778
* native: improve logging for buffer_attrsLennart Poettering2010-02-221-0/+18
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* suspend-on-idle: resume audio device even for initially corked streamsLennart Poettering2010-02-221-5/+3
| | | | | | | | | | | | We need to resume audio devices even for streams that are created in corked stat, so that the latency ranges of the audio device are known during the initial latency negotiation. If we don't the latency negotiation will be based on placeholder data and changed later on which clients do not expect. This should fix issues with Skype. https://bugzilla.redhat.com/show_bug.cgi?id=554929
* pacat: pass buffer_attr to recording streams tooLennart Poettering2010-02-211-1/+1
| | | | | | Pointed out by Colin Guthrie. https://tango.0pointer.de/pipermail/pulseaudio-discuss/2010-February/006698.html
* scache: when playing a sample from the cache make sure not queue them up ↵Lennart Poettering2010-02-215-19/+31
| | | | | | | when the sink is suspended libcanberra already sets the appropriate flags for uncached sample streams, we now need to make sure to set them for cached samples too.
* sample-cache: use the sample name as unmodified fallback in the properties ↵Lennart Poettering2010-02-211-1/+2
| | | | when playing a sample from the cache
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