Re: [pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error

2010-11-25 Thread PulseAudio
#871: PA compile fails on MAC OS X due to bonjour error
---+
  Reporter:  spitfire  |   Owner:  lennart  
  Type:  defect|  Status:  closed   
 Milestone:|   Component:  module-zeroconf-*
Resolution:  fixed |Keywords:   
---+
Changes (by coling):

  * status:  new = closed
  * resolution:  = fixed


Comment:

 OK, I've pushed Daniel's patch now, so will close this bug. If you still
 get a problem, please reopen :)

-- 
Ticket URL: http://pulseaudio.org/ticket/871#comment:6
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Re: [pulseaudio-tickets] [PulseAudio] #716: PA loses mono audio support when remixing is disabled (was: PA loses microphone when remixing is disabled)

2010-11-24 Thread PulseAudio
#716: PA loses mono audio support when remixing is disabled
+---
  Reporter:  Sam Stone  |   Owner:  lennart   
  Type:  defect |  Status:  new   
 Milestone: |   Component:  daemon
Resolution: |Keywords:  microphone, remixing, mono
+---
Changes (by Captain Chaos):

  * keywords:  microphone, remixing = microphone, remixing, mono


Comment:

 Is anything happening on this?

 If I understand it correctly, a pulseaudio user with more than two
 loudspeakers currently has three choices:

 1. Accept that all audio that plays is sent to all speakers, whether or
 not that's appropriate
 2. Accept that mono audio does not play at all
 3. Jerry rig a complicated system with virtual devices, which will only
 partly solve the problem (namely only for programs which allow you to
 choose the device they play to)

 To me all these options are unacceptable. People with surround sound
 systems should not be punished by having their audio not work at all, or
 be played over the wrong speakers. Instead, pulseaudio should
 intelligently choose which speakers to direct the audio to.

 I find the suggestion that remixing is needed to allow mono sound to be
 played on 5.1 speakers a bit odd. Surely mono sound could be played over
 the center speaker, no remixing required? Pulseaudio currently already
 correctly plays stereo sound over just the front left and right speakers
 when remixing is turned off, surely it would not be hard to similarly
 direct mono channels to the center speaker?

-- 
Ticket URL: http://pulseaudio.org/ticket/716#comment:5
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Re: [pulseaudio-tickets] [PulseAudio] #859: racy crashes

2010-11-24 Thread PulseAudio
#859: racy crashes
+---
  Reporter:  jankratochvil  |   Owner:  lennart
  Type:  defect |  Status:  closed 
 Milestone:  0.9.22 |   Component:  clients
Resolution:  fixed  |Keywords: 
+---
Changes (by lennart):

  * status:  new = closed
  * resolution:  = fixed
  * milestone:  = 0.9.22


Comment:

 Applied now to stable-queue. This will go in the next release.

 Sorry for the delay.

-- 
Ticket URL: http://pulseaudio.org/ticket/859#comment:1
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Re: [pulseaudio-tickets] [PulseAudio] #843: Some PA strings are not translated in the Gnome Sound Properties window

2010-11-24 Thread PulseAudio
#843: Some PA strings are not translated in the Gnome Sound Properties window
---+
  Reporter:  kelemeng  |   Owner:  lennart  
  Type:  defect|  Status:  closed   
 Milestone:  0.9.22|   Component:  module-alsa-*
Resolution:  fixed |Keywords:  i18n 
---+
Changes (by lennart):

  * status:  new = closed
  * resolution:  = fixed
  * milestone:  = 0.9.22


Comment:

 Applied in stable-queue. This will be part of the next release.

-- 
Ticket URL: http://pulseaudio.org/ticket/843#comment:8
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Re: [pulseaudio-tickets] [PulseAudio] #845: PulseAudio not detecting sound card - Alienware laptop, ubuntu 10.4

2010-11-24 Thread PulseAudio
#845: PulseAudio not detecting sound card - Alienware laptop, ubuntu 10.4
-+--
  Reporter:  pacanukeha  |   Owner:  lennart  
  Type:  defect  |  Status:  closed   
 Milestone:  |   Component:  module-detect
Resolution:  elsewhere   |Keywords:   
-+--
Changes (by lennart):

  * status:  new = closed
  * resolution:  = elsewhere


Comment:

 The kernel is not aware of your sound card as it appears. Please report
 this bug against the kernel. Also, please direct issues like this to
 Ubuntu first:

 http://pulseaudio.org/wiki/UbuntuBugs

-- 
Ticket URL: http://pulseaudio.org/ticket/845#comment:1
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Re: [pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs

2010-11-24 Thread PulseAudio
#852: Pulseaudio doesn't map multiple inputs/outputs
+---
  Reporter:  LCID Fire  |   Owner:  lennart
  Type:  defect |  Status:  closed 
 Milestone: |   Component:  daemon 
Resolution:  elsewhere  |Keywords: 
+---
Changes (by lennart):

  * status:  new = closed
  * resolution:  = elsewhere


Comment:

 PA does not expose more than one input or more than one output per sound
 card, because we cannot properly detect how they interact. This is a
 limitation and unless we get a proper enumeration interface there this
 cannot be fixed.

 You can manually work-around this however, by loading module-alsa-sink
 with an explicit hw:x,y alsa device.

 Anyway, closing this since we cannot support this unless ALSA is fixed.

-- 
Ticket URL: http://pulseaudio.org/ticket/852#comment:2
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Re: [pulseaudio-tickets] [PulseAudio] #855: Illegal instruction..

2010-11-24 Thread PulseAudio
#855: Illegal instruction..
---+
  Reporter:  ganeshpetkar  |   Owner:  lennart  
  Type:  tracking  |  Status:  new  
 Milestone:  0.9.22|   Component:  gst-pulse
Resolution:|Keywords:   
---+

Comment(by lennart):

 What makes you think this is a PA bug? I'd guess this bug is somwhere
 hidden in the Gst pa driver, not in pa code itself.

 Please provide a backtrace for these (generate this with gdb's bt full)
 crashes, so that we can figure out where this is triggered.

-- 
Ticket URL: http://pulseaudio.org/ticket/855#comment:3
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Re: [pulseaudio-tickets] [PulseAudio] #862: snd_pcm_avail_delay() returned strange values

2010-11-24 Thread PulseAudio
#862: snd_pcm_avail_delay() returned strange values
+---
  Reporter:  oniram |   Owner:  lennart  
  Type:  defect |  Status:  closed   
 Milestone: |   Component:  module-alsa-*
Resolution:  elsewhere  |Keywords:   
+---
Changes (by lennart):

  * status:  new = closed
  * resolution:  = elsewhere


Comment:

 The message is pretty explicit and asks you to report this to the ALSA
 devs, not the PA devs. Please report the bug to the ALSA developers. Thank
 you!

-- 
Ticket URL: http://pulseaudio.org/ticket/862#comment:2
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Re: [pulseaudio-tickets] [PulseAudio] #864: use GNU readline in pacmd

2010-11-24 Thread PulseAudio
#864: use GNU readline in pacmd
--+-
  Reporter:  skierpage|   Owner:  lennart
  Type:  enhancement  |  Status:  closed 
 Milestone:   |   Component:  daemon 
Resolution:  invalid  |Keywords: 
--+-
Changes (by lennart):

  * status:  new = closed
  * resolution:  = invalid


Comment:

 nah, that's hardly possible to implement since pacmd is little more than a
 socat implementation. You are directly talking to the server there, and
 readline requires a TTY to work, not a socket.

-- 
Ticket URL: http://pulseaudio.org/ticket/864#comment:1
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Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults

2010-11-24 Thread PulseAudio
#865: Pulseaudio segfaults
---+
  Reporter:  rabbit64  |   Owner:  lennart 
  Type:  defect|  Status:  closed  
 Milestone:|   Component:  module-combine-*
Resolution:  invalid   |Keywords:  
---+
Changes (by lennart):

  * status:  new = closed
  * resolution:  = invalid


Comment:

 This is intended this way: whenever we run into an underrun we trigger a
 SIGTRAP so that we can debug things. This happens only if you compile
 things with DEBUG_TIMING.

 So, this is not a bug. This is intended behaviour. Don't use DEBUG_TIMING
 if you are not ready to use a debugger to handle its effects!

-- 
Ticket URL: http://pulseaudio.org/ticket/865#comment:5
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Re: [pulseaudio-tickets] [PulseAudio] #877: Bluetooth headset does not appear in the sound devices list

2010-11-24 Thread PulseAudio
#877: Bluetooth headset does not appear in the sound devices list
-+--
  Reporter:  dgvirtual   |   Owner:  lennart   
  Type:  defect  |  Status:  closed
 Milestone:  |   Component:  daemon
Resolution:  distrospecific  |Keywords:  bluetooth, headset
-+--
Changes (by lennart):

  * status:  new = closed
  * resolution:  = distrospecific


Comment:

 Please use bluez-gnome to connect headset with the system.

 Also, please make sure to file ubuntu bugs in the ubuntu bugzilla:
 http://pulseaudio.org/wiki/UbuntuBugs

-- 
Ticket URL: http://pulseaudio.org/ticket/877#comment:2
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Re: [pulseaudio-tickets] [PulseAudio] #834: Provide a well tested 32bit equivalent to padsp

2010-11-24 Thread PulseAudio
#834: Provide a well tested 32bit equivalent to padsp
--+-
  Reporter:  alsuren  |   Owner:  lennart
  Type:  enhancement  |  Status:  closed 
 Milestone:   |   Component:  padsp  
Resolution:  invalid  |Keywords: 
--+-
Changes (by lennart):

  * status:  new = closed
  * resolution:  = invalid


Comment:

 Uh? just compile PA twice and the padsp .so will be compiled twice, too.
 LD_PRELOAD will then ensure that the right version is loaded.

 This is a distro bug, not a PA bug.

-- 
Ticket URL: http://pulseaudio.org/ticket/834#comment:1
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Re: [pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs

2010-11-24 Thread PulseAudio
#852: Pulseaudio doesn't map multiple inputs/outputs
+---
  Reporter:  LCID Fire  |   Owner:  lennart
  Type:  defect |  Status:  closed 
 Milestone: |   Component:  daemon 
Resolution:  elsewhere  |Keywords: 
+---

Comment(by LCID Fire):

 I disagree. You could very well built in a virtual mapping at least for
 the time being - especially since there are not that many cards out there
 with multiple input ports.
 On the other hand - the devices are mostly studio cards where you'd like
 to work on data in realtime and for that one uses ALSA or Jack. So not
 urgent for me.

-- 
Ticket URL: http://pulseaudio.org/ticket/852#comment:3
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[pulseaudio-tickets] [PulseAudio] #877: Bluetooth headset does not appear in the sound devices list

2010-11-21 Thread PulseAudio
#877: Bluetooth headset does not appear in the sound devices list
+---
 Reporter:  dgvirtual   |   Owner:  lennart
 Type:  defect  |  Status:  new
Milestone:  |   Component:  daemon 
 Keywords:  bluetooth, headset  |  
+---
 Pairing the headset and the computer is successful. The connection
 initially is also established (after a fresh reboot). But the headset does
 not appear in Pulseaudio devices list (as checked via pavucontrol, kmix,
 KDE's Phonon devices list).

 Once the bluetooth headset is connected, pulseaudio sometimes freezes (if
 I played sound on Kaffeine, it sometimes stops).

 The headset is ok, works with my phone, I also managed to get it to work
 once with pulseaudio, but do not ask me how, it was accidental and I was
 not able to replicate that (and it was with packages from ppa:ubuntu-
 audio-dev/ppa for maverick ubuntu; since those froze more often, I
 downgraded, and these tests are done with default ubuntu maverick
 packages).

 System information:

 Bluetooth headset: Sony Ericsson HBH-610a

 $ hcitool info 00:16:B8:0A:14:3E

 Requesting information ...
 BD Address:  00:16:B8:0A:14:3E
 Device Name: HBH-610a
 LMP Version: 2.0 (0x3) LMP Subversion: 0xe1b
 Manufacturer: Texas Instruments Inc. (13)
 Features: 0xff 0xfd 0x29 0x78 0x18 0x18 0x00 0x80
 3-slot packets 5-slot packets encryption slot
 offset
 timing accuracy role switch hold mode sniff mode
 park state channel quality SCO link HV2 packets
 HV3 packets u-law log A-law log CVSD transparent
 SCO
 enhanced iscan interlaced iscan interlaced pscan
 inquiry with RSSI AFH cap. slave AFH class. slave
 AFH cap. master AFH class. master extended features


 OS: Kubuntu Maverick i386

 $ uname -a

 Linux bala 2.6.35-22-generic #35-Ubuntu SMP Sat Oct 16 20:36:48 UTC 2010
 i686 GNU/Linux

 $ pulseaudio --version
 pulseaudio 0.9.21-63-gd3efa-dirty

 Some info about the bluetooth adapter:

 $ hciconfig

 hci0: Type: BR/EDR Bus: USB

 BD Address: 00:11:67:D6:35:60 ACL MTU: 1021:4 SCO MTU: 48:10
 UP RUNNING PSCAN ISCAN
 RX bytes:4749 acl:81 sco:0 events:173 errors:0
 TX bytes:2725 acl:96 sco:0 commands:48 errors:0

 $ lsusb | grep Blue

 Bus 004 Device 002: ID 1131:1004 Integrated System Solution Corp.
 Bluetooth Device

 Likely related bugs reported on launchpad:

 https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/530264

 https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/327284

 I am attaching two files. One is filtered output of /var/log/messages of a
 bluetooth connection session.
 The ohter one is pulseaudio - output of the same session.

 I will gladly do any tests required for this bug to be fixed.

-- 
Ticket URL: http://pulseaudio.org/ticket/877
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Re: [pulseaudio-tickets] [PulseAudio] #876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel

2010-11-20 Thread PulseAudio
#876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current
Rygel
-+--
  Reporter:  stephe  |   Owner:  lennart
  Type:  defect  |  Status:  closed 
 Milestone:  |   Component:  daemon 
Resolution:  fixed   |Keywords: 
-+--

Comment(by stephe):

 Thanks. I will take a look at the dbus stuff when I get a chance.

 It sounds like you need to load module-http-protocol-tcp in PA so that
 Rygel can get the data stream from PA to serve up.

-- 
Ticket URL: http://pulseaudio.org/ticket/876#comment:3
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Re: [pulseaudio-tickets] [PulseAudio] #870: USB headphone is not listed in pulseaudio

2010-11-19 Thread PulseAudio
#870: USB headphone is not listed in pulseaudio
+---
  Reporter:  JeroenGeldhof  |   Owner:  lennart
  Type:  defect |  Status:  closed 
 Milestone: |   Component:  daemon 
Resolution:  fixed  |Keywords: 
+---

Comment(by coling):

 I suspect it's some variant of the bug I mentioned then. We'll get that
 sorted eventually... once we nail down the cause... as the saying goes,
 it's like nailing jelly to a fence post :D

-- 
Ticket URL: http://pulseaudio.org/ticket/870#comment:3
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Re: [pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error

2010-11-18 Thread PulseAudio
#871: PA compile fails on MAC OS X due to bonjour error
---+
  Reporter:  spitfire  |   Owner:  lennart  
  Type:  defect|  Status:  new  
 Milestone:|   Component:  module-zeroconf-*
Resolution:|Keywords:   
---+
Description changed by coling:

Old description:

 I tried to compile pulseaudio from git HEAD, and got errors about bonjour
 (of course I don't have avahi installed):


  In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such
 file or directory
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or
 '__attribute__' before '*' token
 ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such
 file or directory
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or
 '__attribute__' before '*' token
 ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token
 lipo: can't open input file: /var/tmp//cc4jQLPi.out (No such file or
 directory)
 make[3]: *** [module-bonjour-publish.lo] Error 1

New description:

 I tried to compile pulseaudio from git HEAD, and got errors about bonjour
 (of course I don't have avahi installed):

 {{{
  In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file
 or directory
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or
 '__attribute__' before '*' token
 ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file
 or directory
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or
 '__attribute__' before '*' token
 ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token
 lipo: can't open input file: /var/tmp//cc4jQLPi.out (No such file or
 directory)
 make[3]: *** [module-bonjour-publish.lo] Error 1
 }}}

--

-- 
Ticket URL: http://pulseaudio.org/ticket/871#comment:1
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Re: [pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error

2010-11-18 Thread PulseAudio
#871: PA compile fails on MAC OS X due to bonjour error
---+
  Reporter:  spitfire  |   Owner:  lennart  
  Type:  defect|  Status:  new  
 Milestone:|   Component:  module-zeroconf-*
Resolution:|Keywords:   
---+

Comment(by coling):

 Hi,

 Daniel Mack has done a lot of the OSX compatibility stuff so I suggest you
 contact him.

 I'll send him a mail to notify him on this ticket.

 Col

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #868: No digital out with Audiotrak Prodigy HD2

2010-11-18 Thread PulseAudio
#868: No digital out with Audiotrak Prodigy HD2
--+-
  Reporter:  anbello  |   Owner:  lennart 
  Type:  defect   |  Status:  new 
 Milestone:   |   Component:  daemon  
Resolution:   |Keywords:  Audiotrak Prodigy HD2 ICE1724 Envy24
--+-

Comment(by coling):

 Can you use the device via alsa as iec958:0 e.g. aplay -D iec958:0

 PA relies on lower level alsa constructs to make it work, so even if it
 works with hw:0,1, then it doesn't really help us unless it's wrapped up
 properly.

 You can hack in support by editing the mixer path scripts or even
 writing a custom mixer profile for the device (and assigning it via a udev
 rule) by fiddling with the files in /usr/share/pulseaudio/alsa-mixer/.
 They are all hand editable (tho' be careful not to change too much that
 you break it!) and do not require PA to be recompiled (just restarted).

 If the alsa level access works fine with the iec958 device, then please
 attach the output of amixer -c0 and pacmd list to this bug.

-- 
Ticket URL: http://pulseaudio.org/ticket/868#comment:1
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Re: [pulseaudio-tickets] [PulseAudio] #876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel

2010-11-17 Thread PulseAudio
#876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current
Rygel
-+--
  Reporter:  stephe  |   Owner:  lennart
  Type:  defect  |  Status:  closed 
 Milestone:  |   Component:  daemon 
Resolution:  fixed   |Keywords: 
-+--
Changes (by coling):

  * status:  new = closed
  * resolution:  = fixed


Comment:

 OK, I've merged this now. I wasn't able to get it working on my PS3 but
 then I'm not sure it worked before anyway (first time I tried). I get a
 DLNA error when I try to connect to the audio device :( But it does show
 up and I think that's good enough for a merge :D

 One comment about the patch, it seems to define some functions that are
 very dbus centric. If these are standard utilities, then I suggest they
 go instead into dbus-util.c. If you could look into that I'd be very
 greatful. Feel free to drop by on IRC or on the mailing list if it's
 easier.

 I'll also push this out to stable-queue branch so it'll be part of the
 next PA release. I think that's fair to target it at that.

-- 
Ticket URL: http://pulseaudio.org/ticket/876#comment:2
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Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%

2010-11-17 Thread PulseAudio
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
-+--
  Reporter:  murz|   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--

Comment(by murz):

 One of the method to solve this is to store ballance value of channels
 when setting volume to 0%, but it solve the problem only for 2-channels
 sinks, but for describe volumes in 4.1-7.1 sink we must create more than
 one ballance value.
 I think that better solution is store per-channel volumes in sink
 unchanged, and create new value - main sink volume (something like base
 volume). And via increase/decrease sink commands do the change only main
 volume and don't touch channel volumes config.

 And adding main sink volume will add new feature to pulseaudio: ability to
 manually set maximum volume for sink (user can set 50% volume of each
 channel in sink and after that he been able to not be afraid making sound
 too loud, increasing main volume to 100%).

 Because very often sound systems have many decibels that was never used,
 because on 100% volume the sound is very loud.

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #789: 100% CPU use

2010-11-15 Thread PulseAudio
#789: 100% CPU use
-+--
  Reporter:  porton  |   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--

Comment(by porton):

 {{{
 (gdb) thread apply all bt

 Thread 3 (Thread 0xb2d3cb70 (LWP 3711)):
 #0  0xb78b7424 in __kernel_vsyscall ()
 #1  0xb7489775 in ppoll () from /lib/i686/cmov/libc.so.6
 #2  0xb784fda0 in pa_rtpoll_run () from /usr/lib/libpulsecore-0.9.21.so
 #3  0xb2d5a4ac in ?? () from /usr/lib/pulse-0.9.21/modules/libalsa-util.so
 #4  0xb780e322 in ?? () from /usr/lib/libpulsecommon-0.9.21.so
 #5  0xb7554955 in start_thread () from /lib/i686/cmov/libpthread.so.0
 #6  0xb7496e7e in clone () from /lib/i686/cmov/libc.so.6

 Thread 2 (Thread 0xb253bb70 (LWP 3713)):
 #0  0xb78b7424 in __kernel_vsyscall ()
 #1  0xb7489775 in ppoll () from /lib/i686/cmov/libc.so.6
 #2  0xb784fda0 in pa_rtpoll_run () from /usr/lib/libpulsecore-0.9.21.so
 #3  0xb2d60c66 in ?? () from /usr/lib/pulse-0.9.21/modules/libalsa-util.so
 #4  0xb780e322 in ?? () from /usr/lib/libpulsecommon-0.9.21.so
 #5  0xb7554955 in start_thread () from /lib/i686/cmov/libpthread.so.0
 #6  0xb7496e7e in clone () from /lib/i686/cmov/libc.so.6

 Thread 1 (Thread 0xb7012700 (LWP 3670)):
 #0  0xb78b7424 in __kernel_vsyscall ()
 #1  0xb7489775 in ppoll () from /lib/i686/cmov/libc.so.6
 #2  0xb77ad4aa in pa_mainloop_poll () from /usr/lib/libpulse.so.0
 #3  0xb77aec43 in pa_mainloop_iterate () from /usr/lib/libpulse.so.0
 #4  0xb77aed14 in pa_mainloop_run () from /usr/lib/libpulse.so.0
 #5  0x08052a2e in main ()
 }}}

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[pulseaudio-tickets] [PulseAudio] #876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel

2010-11-14 Thread PulseAudio
#876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current
Rygel
+---
 Reporter:  stephe  |   Owner:  lennart
 Type:  defect  |  Status:  new
Milestone:  |   Component:  daemon 
 Keywords:  |  
+---
 This patch implements the MediaServer2 D-Bus interface,
 http://live.gnome.org/Rygel/MediaServer2Spec, so that PA can work with
 Rygel version 0.7.1 and higher.

 Tested with Rygel 0.8.2 on Fedora 14. MediaStreamer running on a Nokia
 N800 and a Sony BDP-S570 Blu-ray player were used as the upnp clients.

 A note about testing. Something in Rygel seems to be suffering from the
 recent change to memcpy in glibc. The LD_PRELOAD trick in Fedora bug
 https://bugzilla.redhat.com/show_bug.cgi?id=638477 works around the
 problem. Also, the DLNA/UPnP Streaming source is very distorted with
 crackling and other noises and seems to run too fast.

-- 
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[pulseaudio-tickets] [PulseAudio] #875: [patch] Add module for jack sense evdev input devices

2010-11-12 Thread PulseAudio
#875: [patch] Add module for jack sense evdev input devices
-+--
 Reporter:  joeshaw  |   Owner:  lennart
 Type:  enhancement  |  Status:  new
Milestone:   |   Component:  core   
 Keywords:   |  
-+--
 Sound modules in the Linux kernel have the ability to create input devices
 for sound card jacks through the snd_jack API.  (See sound/core/jack.c)
 These devices are exported as evdev devices which notify (and can be
 queried about) a few different switches.

 I'm attaching a new PA module which maps one of those devices to a source
 and/or sink.  Whenever the jack sense changes, a property is set on the
 source/sink.  Clients can watch for notifications and take action (such as
 muting, changing the default source/sink, migrating streams to that
 source/sink) when the property changes.

-- 
Ticket URL: http://pulseaudio.org/ticket/875
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Re: [pulseaudio-tickets] [PulseAudio] #875: [patch] Add module for jack sense evdev input devices

2010-11-12 Thread PulseAudio
#875: [patch] Add module for jack sense evdev input devices
--+-
  Reporter:  joeshaw  |   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  core   
Resolution:   |Keywords: 
--+-

Comment(by coling):

 Thank you very much for this. I just wish you'd discussed the
 implementation with us before starting development.

 Jack Sensing support is not something that should be handled as a module,
 but rather something built in to the alsa-sink/source itself.

 Ultimately Jack Sensing should result in a change of sink/source port
 e.g. changing from built in Speakers to Headphones port in the case.

 Clients should not need to do any watching of sink properties etc. however
 they may wish to take some action, however they will be made aware of this
 due to the port changes anyway.

 The main complications with regard to Jack Sensing comes from when users
 have specifically overridden the port manually - who do we want to take
 notice of, the user choice or the automatic information? I'd err on the
 side of the latter, but there are still some complications (e.g. some
 hardware has three ports - Speakers, Headphones and something else I
 can't remember off the top of my head, so it's not a simple binary
 choice.

 But certainly the ports system is the correct place to hook up jack
 sensing. I'd love it if you would come and discuss this with us on the IRC
 channel or via the devel list.

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Re: [pulseaudio-tickets] [PulseAudio] #875: [patch] Add module for jack sense evdev input devices

2010-11-12 Thread PulseAudio
#875: [patch] Add module for jack sense evdev input devices
--+-
  Reporter:  joeshaw  |   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  core   
Resolution:   |Keywords: 
--+-

Comment(by joeshaw):

 Is there a concept of a null port?  We would need the null port to be
 selected until another jack was sensed, because our clients make
 determinations about which sound card to use based on whether the jacks on
 one of them is used at all.

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%

2010-11-08 Thread PulseAudio
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
-+--
  Reporter:  murz|   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--

Comment(by murz):

 In sink I see the value Base Volume, maybe with it I can change the
 volume without break the balance. How can I change it via pactl or api?

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%

2010-11-08 Thread PulseAudio
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
-+--
  Reporter:  murz|   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--

Comment(by tanuk):

 Replying to [comment:3 murz]:
  In sink I see the value Base Volume, maybe with it I can change the
 volume without break the balance. How can I change it via pactl or api?

 Nope, base volume doesn't solve this problem. Base volume is the level
 that alsa reports as 0dB.

 I guess solving this problem requires changing the internal volume
 representation. Currently volume is represented as a list of individual
 channel volumes relative to a fixed 100% level. Maybe it should be changed
 so that there would be a separate reference level variable that would
 tell the level of the loudest channel, and other channels would be
 relative to this reference level. This way the balance would be preserved
 even when the reference level is set to 0%.

 This would require quite a lot of work, but it should be fairly
 straightforward work.

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[pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%

2010-11-07 Thread PulseAudio
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
+---
 Reporter:  murz|   Owner:  lennart
 Type:  defect  |  Status:  new
Milestone:  |   Component:  daemon 
 Keywords:  |  
+---
 I have setup different channel volume for left, right and other channels
 in output sink.

 When I changing volume to 0% and restore it (I try kmix in KDE and veromix
 plasmoid), my per-channel volume config is broken, all channels have the
 same volume and I need to do the config again.
 Here is video for kmix: http://www.youtube.com/watch?v=zjPbBd2CCzc

 What I can do to don't break per-channel volume when setting 0% volume?

 Does pulseaudio have any option in API or special function/method for
 solving this issue?

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%

2010-11-07 Thread PulseAudio
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
-+--
  Reporter:  murz|   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--

Comment(by murz):

 gnome volume manager have the same problem, here is the bug about this in
 ubuntu: https://bugs.launchpad.net/ubuntu/+source/gnome-media/+bug/672420

-- 
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[pulseaudio-tickets] [PulseAudio] #872: Loss of sound with two sources

2010-11-04 Thread PulseAudio
#872: Loss of sound with two sources
+---
 Reporter:  AdamK   |   Owner:  lennart
 Type:  defect  |  Status:  new
Milestone:  |   Component:  daemon 
 Keywords:  |  
+---
 I have one sound source with no heavy latency requirements (movie playing
 in VLC).

 Now, if another sound is played (system sound for example), the first time
 two sources have to be mixed there is a slight loss of sound (for less
 then second). It does not happen afterwards.

 $ pulseaudio --version
 pulseaudio 0.9.21-98-ga8d7-dirty
 (ubuntu beta)

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #872: Loss of sound with two sink inputs (was: Loss of sound with two sources)

2010-11-04 Thread PulseAudio
#872: Loss of sound with two sink inputs
-+--
  Reporter:  AdamK   |   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--

Comment(by coling):

 Could this be a result of artefacts from the Flat Volumes feature?
 (pulseaudio will always try and use h/w volume whenever possible to reduce
 overhead and thus power consumption etc - and thus when there is only one
 stream playing, the volume of that stream will always be the same as the
 underlying alsa volume - although the concept of a system volume is
 still maintained and presented to the user even if it's not really used
 like that underneath!)

 To test this, load up alsamixer -c0 and have a look at the bars for Master
 and maybe PCM too. When your second sink input (source is not the right
 term as a source is a recording device - e.g. a mic. I've fixed the
 summary to reflect this.) starts (and you hear your dropout), do you see
 them jump about a bit?

 If so, you can work around this by disabling flat-volumes in daemon.conf.
 Git master already has support to minimise the fallout from this - which
 stems from there being no way to synchronise mixer changes in alsa - so if
 this is the problem you are seeing, it's likely fixed already :)

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #821: volume mute at 15%

2010-11-03 Thread PulseAudio
#821: volume mute at 15%
+---
  Reporter:  yelo3  |   Owner:  lennart  
  Type:  defect |  Status:  closed   
 Milestone: |   Component:  module-alsa-*
Resolution:  elsewhere  |Keywords:   
+---
Changes (by tanuk):

  * status:  new = closed
  * resolution:  = elsewhere


Comment:

 Replying to [comment:2 starT_T]:
  I tested, set the volume to 14%, -51.23 dB, I am sure that I can hear
 the sound.

 Great.

  -48 dB isn't mute.

 Yep. It isn't.

  But which value is good for mute?

 In Pulseaudio volume scale (0-65536) 0 means mute. In decibel scale -inf
 dB means mute. Did that answer your question?

 I'm a bit confused about your message - are you just reporting that you
 can't reproduce this bug? This bug is actually an alsa bug, so whether you
 can reproduce this or not depends on your hardware.

 Hmm, since this is an alsa bug, I guess nobody has anything against me
 resolving this ticket as elsewhere.

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Re: [pulseaudio-tickets] [PulseAudio] #821: volume mute at 15%

2010-11-03 Thread PulseAudio
#821: volume mute at 15%
+---
  Reporter:  yelo3  |   Owner:  lennart  
  Type:  defect |  Status:  closed   
 Milestone: |   Component:  module-alsa-*
Resolution:  elsewhere  |Keywords:   
+---

Comment(by coling):

 Yes I think this is a valid closer Tanu.

 For reference, I recently helped to solve this problem on my own h/w.

 The error was a combination of three problems: one in the h/w itself, one
 in the alsa driver and one in the alsa userspace library.

  * The hardware did not follow the HDA spec and muted the device when it
 was the slider was set to 0.
  * The driver did not have a quirk defined for this.
  * The library did not pay attention to the quirk when it was added.

 So there are a few commits needed to fix it:

  * kernel:
 
[http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=de8c85f7840e5e29629de95f5af24297fb325e0b
 de8c85] and
 
[http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=a74ccea51d4314632a81d568d59bf885e5b09d93
 a74cce] (but obviously this is specific to my h/w - yours may need similar
 fixes if it doesn't have them already)
  * alsa-lib: [http://git.alsa-project.org/?p=alsa-
 lib.git;a=commit;h=2f6206da0c1ff88235e6eca0077343f22a4b43ee 2f6206]

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Re: [pulseaudio-tickets] [PulseAudio] #848: Loud noise for pa_stream_flush()

2010-11-02 Thread PulseAudio
#848: Loud noise for pa_stream_flush()
+---
  Reporter:  mschwendt  |   Owner:  lennart 
  Type:  defect |  Status:  new 
 Milestone: |   Component:  daemon  
Resolution: |Keywords:  very loud short sound on stream flushing
+---
Changes (by adi):

 * cc: a...@drcomp.erfurt.thur.de (added)


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Re: [pulseaudio-tickets] [PulseAudio] #821: volume mute at 15%

2010-11-02 Thread PulseAudio
#821: volume mute at 15%
-+--
  Reporter:  yelo3   |   Owner:  lennart  
  Type:  defect  |  Status:  new  
 Milestone:  |   Component:  module-alsa-*
Resolution:  |Keywords:   
-+--

Comment(by starT_T):

 {{{
  set-sink-volume 0 9175

 D: alsa-sink.c: Requested volume: 0:  14% 1:  14%
 D: alsa-sink.c: Got hardware volume: 0:  14% 1:  14%
 D: alsa-sink.c: Calculated software volume: 0:  99% 1:  99% (accurate-
 enough=yes)

  list-sinks
 1 sink(s) available.
   * index: 0
 name: alsa_output.0.analog-stereo
 driver: module-alsa-sink.c
 flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
 FLAT_VOLUME
 state: RUNNING
 suspend cause:
 priority: 9009
 volume: 0:  14% 1:  14%
 0: -51.23 dB 1: -51.23 dB
 balance 0.00
 base volume:  10%
  -60.00 dB
 volume steps: 65537
 }}}

 I tested, set the volume to 14%, -51.23 dB, I am sure that I can hear the
 sound. -48 dB isn't mute. But which value is good for mute?

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Re: [pulseaudio-tickets] [PulseAudio] #789: 100% CPU use

2010-11-01 Thread PulseAudio
#789: 100% CPU use
-+--
  Reporter:  porton  |   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--

Comment(by Ford_Prefect):

 If you are familiar with gdb, when CPU usage hits 100%, could you attach
 to the process and provide a backtrace? Roughly, this would required:

 gdb path-to-pulseaudio pid-of-pulseaudio
 thread apply all bt

 Also would help if you're using the latest version of PA available on your
 distro.

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Re: [pulseaudio-tickets] [PulseAudio] #822: Closing the last stream on an ALSA sink causes distortion

2010-11-01 Thread PulseAudio
#822: Closing the last stream on an ALSA sink causes distortion
+---
  Reporter:  adi|   Owner:  lennart
  Type:  defect |  Status:  closed 
 Milestone: |   Component:  daemon 
Resolution:  duplicate  |Keywords: 
+---
Changes (by Ford_Prefect):

  * status:  new = closed
  * resolution:  = duplicate


Comment:

 Marking as a duplicate as pointed out.

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Re: [pulseaudio-tickets] [PulseAudio] #782: loud woof when closing a player

2010-11-01 Thread PulseAudio
#782: loud woof when closing a player
-+--
  Reporter:  patra...@gmail.com  |   Owner:  lennart
  Type:  defect  |  Status:  closed 
 Milestone:  |   Component:  daemon 
Resolution:  duplicate   |Keywords: 
-+--
Changes (by Ford_Prefect):

 * cc: a...@accosted.net (added)
  * status:  new = closed
  * resolution:  = duplicate


Comment:

 Marking as duplicate of #848.

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Re: [pulseaudio-tickets] [PulseAudio] #822: Closing the last stream on an ALSA sink causes distortion

2010-11-01 Thread PulseAudio
#822: Closing the last stream on an ALSA sink causes distortion
+---
  Reporter:  adi|   Owner:  lennart
  Type:  defect |  Status:  closed 
 Milestone: |   Component:  daemon 
Resolution:  duplicate  |Keywords: 
+---
Changes (by Ford_Prefect):

 * cc: a...@accosted.net (added)


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Re: [pulseaudio-tickets] [PulseAudio] #789: 100% CPU use

2010-11-01 Thread PulseAudio
#789: 100% CPU use
-+--
  Reporter:  porton  |   Owner:  lennart
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  daemon 
Resolution:  |Keywords: 
-+--
Changes (by Ford_Prefect):

 * cc: a...@accosted.net (added)


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[pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error

2010-10-31 Thread PulseAudio
#871: PA compile fails on MAC OS X due to bonjour error
--+-
 Reporter:  spitfire  |   Owner:  lennart  
 Type:  defect|  Status:  new  
Milestone:|   Component:  module-zeroconf-*
 Keywords:|  
--+-
 I tried to compile pulseaudio from git HEAD, and got errors about bonjour
 (of course I don't have avahi installed):


  In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file
 or directory
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or
 '__attribute__' before '*' token
 ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file
 or directory
 In file included from modules/module-bonjour-publish.c:47:
 ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or
 '__attribute__' before '*' token
 ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token
 lipo: can't open input file: /var/tmp//cc4jQLPi.out (No such file or
 directory)
 make[3]: *** [module-bonjour-publish.lo] Error 1

-- 
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[pulseaudio-tickets] [PulseAudio] #868: No digital out with Audiotrak Prodigy HD2

2010-10-29 Thread PulseAudio
#868: No digital out with Audiotrak Prodigy HD2
--+-
 Reporter:  anbello   |   Owner:  lennart
 Type:  defect|  Status:  new
Milestone:|   Component:  daemon 
 Keywords:  Audiotrak Prodigy HD2 ICE1724 Envy24  |  
--+-
 I have an Audiotrak Prodigy HD2 sound card on a PC with ubuntu 10.10 (i
 have tried both 32bit and 64bit) and i am not able to use the digital
 spdif out with pulseaudio.

 In Sound Preferences - Hardware tab - Profile: combobox i have
 only Analog Stereo Input, Analog Stereo Output and Analog Stereo
 Duplex and no Digital Stereo * of any sort so i cannot use pulseaudio
 with spdif output.

 With another Envy24HT based sound card (Terratec Aureon 7.1 Space) i had
 no problems and in Profile: combobox i had Digital Stereo Output
 (IEC958) (or something similar).

 With alsa i have no problems:

 aplay -l[[BR]]
  List of PLAYBACK Hardware Devices [[BR]]
 card 0: HD2 [Audiotrak Prodigy HD2], device 0: ICE1724 [ICE1724][[BR]]
 Subdevices: 1/1[[BR]]
 Subdevice #0: subdevice #0[[BR]]
 card 0: HD2 [Audiotrak Prodigy HD2], device 1: ICE1724 IEC958 [ICE1724
 IEC958][[BR]]
 Subdevices: 1/1[[BR]]
 Subdevice #0: subdevice #0[[BR]]



 Thanks[[BR]]
 Andrea

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[pulseaudio-tickets] [PulseAudio] #869: 100% cpu use

2010-10-29 Thread PulseAudio
#869: 100% cpu use
+---
 Reporter:  mccann  |   Owner:  lennart
 Type:  defect  |  Status:  new
Milestone:  |   Component:  daemon 
 Keywords:  |  
+---
 Noticed my battery drain very very quickly.  Looked at top:
   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
 17959 mccann 9 -11  107m 2944 2320 R 99.7  0.1 850:13.90 pulseaudio
 Yikes.

 strace showed:
 read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device)
 ppoll([{fd=4, events=POLLIN}, {fd=38, events=POLLIN}, {fd=37,
 events=POLLIN}, {fd=27, events=POLLIN}, {fd=33, events=POLLIN}, {fd=10,
 events=POLLIN}, {fd=21, events=POLLIN}, {fd=7, events=POLLIN}, {fd=20,
 events=POLLIN}, {fd=26, events=POLLIN}, {fd=32, events=POLLIN}, {fd=36,
 events=POLLIN|POLLERR|POLLHUP}, {fd=36, events=0}, {fd=35, events=POLLIN},
 {fd=34, events=POLLIN}, {fd=14, events=POLLIN}, {fd=28, events=POLLIN},
 {fd=31, events=POLLIN}, {fd=22, events=POLLIN}, {fd=25, events=POLLIN},
 {fd=19, events=POLLIN|POLLERR|POLLHUP}, {fd=19, events=0}, {fd=15,
 events=POLLIN}, {fd=18, events=POLLIN}, {fd=8, events=POLLIN}], 25, NULL,
 NULL, 8) = 1 ([{fd=32, revents=POLLERR}])
 read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device)
 ppoll([{fd=4, events=POLLIN}, {fd=38, events=POLLIN}, {fd=37,
 events=POLLIN}, {fd=27, events=POLLIN}, {fd=33, events=POLLIN}, {fd=10,
 events=POLLIN}, {fd=21, events=POLLIN}, {fd=7, events=POLLIN}, {fd=20,
 events=POLLIN}, {fd=26, events=POLLIN}, {fd=32, events=POLLIN}, {fd=36,
 events=POLLIN|POLLERR|POLLHUP}, {fd=36, events=0}, {fd=35, events=POLLIN},
 {fd=34, events=POLLIN}, {fd=14, events=POLLIN}, {fd=28, events=POLLIN},
 {fd=31, events=POLLIN}, {fd=22, events=POLLIN}, {fd=25, events=POLLIN},
 {fd=19, events=POLLIN|POLLERR|POLLHUP}, {fd=19, events=0}, {fd=15,
 events=POLLIN}, {fd=18, events=POLLIN}, {fd=8, events=POLLIN}], 25, NULL,
 NULL, 8) = 1 ([{fd=32, revents=POLLERR}])
 read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device)
 ppoll([{fd=4, events=POLLIN}, {fd=38, events=POLLIN}, {fd=37,
 events=POLLIN}, {fd=27, events=POLLIN}, {fd=33, events=POLLIN}, {fd=10,
 events=POLLIN}, {fd=21, events=POLLIN}, {fd=7, events=POLLIN}, {fd=20,
 events=POLLIN}, {fd=26, events=POLLIN}, {fd=32, events=POLLIN}, {fd=36,
 events=POLLIN|POLLERR|POLLHUP}, {fd=36, events=0}, {fd=35, events=POLLIN},
 {fd=34, events=POLLIN}, {fd=14, events=POLLIN}, {fd=28, events=POLLIN},
 {fd=31, events=POLLIN}, {fd=22, events=POLLIN}, {fd=25, events=POLLIN},
 {fd=19, events=POLLIN|POLLERR|POLLHUP}, {fd=19, events=0}, {fd=15,
 events=POLLIN}, {fd=18, events=POLLIN}, {fd=8, events=POLLIN}], 25, NULL,
 NULL, 8) = 1 ([{fd=32, revents=POLLERR}])
 read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device)

 And gdb showed:
 Thread 4 (Thread 0xb753bb70 (LWP 17960)):
 #0  0x00efc416 in __kernel_vsyscall ()
 #1  0x006bdec5 in ppoll (fds=0x847aa58, nfds=2, timeout=value optimized
 out, sigmask=0x0) at ../sysdeps/unix/sysv/linux/ppoll.c:58
 #2  0x0059dea0 in pa_rtpoll_run (p=0x8477dc0, wait_op=true) at
 pulsecore/rtpoll.c:304
 #3  0x001d061c in thread_func (userdata=0x847a890) at modules/alsa/alsa-
 sink.c:1430
 #4  0x003bd442 in internal_thread_func (userdata=0x84c6410) at pulsecore
 /thread-posix.c:72
 #5  0x00510919 in start_thread (arg=0xb753bb70) at pthread_create.c:301
 #6  0x006c8cbe in clone () at ../sysdeps/unix/sysv/linux/i386/clone.S:133

 Thread 3 (Thread 0xb294eb70 (LWP 17961)):
 #0  0x00efc416 in __kernel_vsyscall ()
 #1  0x006bdec5 in ppoll (fds=0x847f958, nfds=2, timeout=value optimized
 out, sigmask=0x0) at ../sysdeps/unix/sysv/linux/ppoll.c:58
 #2  0x0059dea0 in pa_rtpoll_run (p=0x84bf188, wait_op=true) at
 pulsecore/rtpoll.c:304
 #3  0x001d6d86 in thread_func (userdata=0x84a0a10) at modules/alsa/alsa-
 source.c:1274
 #4  0x003bd442 in internal_thread_func (userdata=0x84d1290) at pulsecore
 /thread-posix.c:72
 #5  0x00510919 in start_thread (arg=0xb294eb70) at pthread_create.c:301
 #6  0x006c8cbe in clone () at ../sysdeps/unix/sysv/linux/i386/clone.S:133

 Thread 2 (Thread 0xb1dffb70 (LWP 17962)):
 #0  0x00efc416 in __kernel_vsyscall ()
 #1  0x006bdec5 in ppoll (fds=0x847e7b8, nfds=2, timeout=value optimized
 out, sigmask=0x0) at ../sysdeps/unix/sysv/linux/ppoll.c:58
 #2  0x0059dea0 in pa_rtpoll_run (p=0x847e790, wait_op=true) at
 pulsecore/rtpoll.c:304
 #3  0x001d6d86 in thread_func (userdata=0x847e5e0) at modules/alsa/alsa-
 source.c:1274
 #4  0x003bd442 in internal_thread_func (userdata=0x84d8ba8) at pulsecore
 /thread-posix.c:72
 #5  0x00510919 in start_thread (arg=0xb1dffb70) at pthread_create.c:301
 #6  0x006c8cbe in clone () at ../sysdeps/unix/sysv/linux/i386/clone.S:133

 Thread 1 (Thread 0xb7743700 (LWP 17959)):
 #0  0x00efc416 in __kernel_vsyscall ()
 #1  0x006bdec5 in ppoll (fds=0x84700b8, nfds=25, timeout=value optimized
 out

Re: [pulseaudio-tickets] [PulseAudio] #858: Pulseaudio uses all CPU on ALSA problems

2010-10-23 Thread PulseAudio
#858: Pulseaudio uses all CPU on ALSA problems
+---
  Reporter:  LCID Fire  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  daemon 
Resolution: |Keywords: 
+---

Comment(by LCID Fire):

 After upgrading to 2.6.36 I don't get the dmesgs anymore - but it still
 locks the whole machine. Even worse - apps that use Pulseaudio like
 Thunderbird and Empathy do the same.

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #851: pulsesink not playing

2010-10-20 Thread PulseAudio
#851: pulsesink not playing
+---
  Reporter:  LCID Fire  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  gst-pulse  
Resolution: |Keywords:  gst-launch pulseaudio pulsesink
+---

Comment(by LCID Fire):

 Using the released versions in Ubuntu 10.10 it works for me again, too.

-- 
Ticket URL: http://pulseaudio.org/ticket/851#comment:3
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Re: [pulseaudio-tickets] [PulseAudio] #851: pulsesink not playing

2010-10-20 Thread PulseAudio
#851: pulsesink not playing
+---
  Reporter:  LCID Fire  |   Owner:  lennart
  Type:  defect |  Status:  closed 
 Milestone: |   Component:  gst-pulse  
Resolution:  fixed  |Keywords:  gst-launch pulseaudio pulsesink
+---
Changes (by LCID Fire):

  * status:  new = closed
  * resolution:  = fixed


-- 
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[pulseaudio-tickets] [PulseAudio] #867: Please update POTFILES.in

2010-10-20 Thread PulseAudio
#867: Please update POTFILES.in
--+-
 Reporter:  kelemeng  |   Owner:  lennart 
 Type:  defect|  Status:  new 
Milestone:|   Component:  build-system
 Keywords:|  
--+-
 I just added Pulseaudio to Gnome's Damned Lies (so that Gnome translators
 can see the stats where they are looking for it:
 http://l10n.gnome.org/module/Pulseaudio/ ), and intltool is complaining,
 please fix this:

 There are some missing files from POTFILES.in:

 * src/daemon/pulseaudio-kde.desktop.in
 * src/modules/module-equalizer-sink.c
 * src/modules/module-virtual-sink.c
 * src/modules/module-virtual-source.c

 Also, it would be great to sync the master-tx and master branches :).

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults

2010-10-19 Thread PulseAudio
#865: Pulseaudio segfaults
---+
  Reporter:  rabbit64  |   Owner:  lennart 
  Type:  defect|  Status:  new 
 Milestone:|   Component:  module-combine-*
Resolution:|Keywords:  
---+

Comment(by rabbit64):

  So you've patched alsa-sink.c to define DEBUG_TIMING. Why? These
 segfaults are not segfaults, instead the daemon gets the debug trap
 signal.
 To get some more information about timing in log files. Raymond from alsa
 has advised me to do so. Here is link to the original thread:
 [https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5146]. I think you
 need an account in alsa bugtracker.
  So what to do? Should this ticket be closed? Should we remove the
 PA_DEBUG_TRAP line from check_left_to_play()?
 No, it is probably intentional. There may be a bug in emu10k1 driver and
 coredump should provide more information if I understand it correctly. If
 DEBUG_TIMING is disabled, pulseaudio recovers from this state by spawning
 new process. But after some respawns audacious audio player freezes and
 has to be SIGKILLed.

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Re: [pulseaudio-tickets] [PulseAudio] #851: pulsesink not playing

2010-10-19 Thread PulseAudio
#851: pulsesink not playing
+---
  Reporter:  LCID Fire  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  gst-pulse  
Resolution: |Keywords:  gst-launch pulseaudio pulsesink
+---

Comment(by starT_T):

 {{{

 s...@ubuntu:~$ gst-launch-0.10 -v pulsesrc ! pulsesink
 Setting pipeline to PAUSED ...
 /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: actual-buffer-time =
 2376
 /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: actual-latency-time = 8
 /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0.GstPad:src: caps = audio/x
 -raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16,
 depth=(int)16, rate=(int)44100, channels=(int)2
 Pipeline is live and does not need PREROLL ...
 Setting pipeline to PLAYING ...
 New clock: GstAudioSrcClock
 /GstPipeline:pipeline0/GstPulseSink:pulsesink0.GstPad:sink: caps = audio/x
 -raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16,
 depth=(int)16, rate=(int)44100, channels=(int)2
 /GstPipeline:pipeline0/GstPulseSink:pulsesink0: volume = 1.00
 /GstPipeline:pipeline0/GstPulseSink:pulsesink0: mute = FALSE
 ^CCaught interrupt -- handling interrupt.
 Interrupt: Stopping pipeline ...
 Execution ended after 56762866587 ns.
 Setting pipeline to PAUSED ...
 Setting pipeline to READY ...
 /GstPipeline:pipeline0/GstPulseSink:pulsesink0.GstPad:sink: caps = NULL
 /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0.GstPad:src: caps = NULL
 Setting pipeline to NULL ...
 Freeing pipeline ...
 s...@ubuntu:~$ pulseaudio --version
 pulseaudio 0.9.21-63-gd3efa-dirty
 s...@ubuntu:~$ gst-launch-0.10 --version
 gst-launch-0.10 version 0.10.28
 GStreamer 0.10.28
 https://launchpad.net/distros/ubuntu/+source/gstreamer0.10

 }}}

 works fine for me too

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Re: [pulseaudio-tickets] [PulseAudio] #855: Illegal instruction..

2010-10-19 Thread PulseAudio
#855: Illegal instruction..
---+
  Reporter:  ganeshpetkar  |   Owner:  lennart  
  Type:  tracking  |  Status:  new  
 Milestone:  0.9.22|   Component:  gst-pulse
Resolution:|Keywords:   
---+

Comment(by starT_T):

 {{{
 s...@ubuntu:~$gst-launch-0.10 filesrc location=/home/star/Desktop/bbb.wav
 ! wavparse ! audioconvert ! audioresample ! pulsesink
 Setting pipeline to PAUSED ...
 Pipeline is PREROLLING ...
 Pipeline is PREROLLED ...
 Setting pipeline to PLAYING ...
 New clock: GstPulseSinkClock
 Got EOS from element pipeline0.
 Execution ended after 3720897522 ns.
 Setting pipeline to PAUSED ...
 Setting pipeline to READY ...
 Setting pipeline to NULL ...
 Freeing pipeline ...
 }}}

 my GStreamer version is 0.10.28
 I played very well

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Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults

2010-10-17 Thread PulseAudio
#865: Pulseaudio segfaults
---+
  Reporter:  rabbit64  |   Owner:  lennart 
  Type:  defect|  Status:  new 
 Milestone:|   Component:  module-combine-*
Resolution:|Keywords:  
---+

Comment(by tanuk):

 So you've patched alsa-sink.c to define DEBUG_TIMING. Why? These
 segfaults are not segfaults, instead the daemon gets the debug trap
 signal.

 The check_left_to_play() function has this piece of code:

 {{{
 #ifdef DEBUG_TIMING
 PA_DEBUG_TRAP;
 #endif
 }}}

 Are you arguing that it's a bug to trigger the signal when DEBUG_TIMING is
 defined? I don't have an opinion myself, since it's not documented what
 DEBUG_TIMING means, so I don't know what's the intended behaviour - is
 triggering the signal a good thing to do or not.

 Anyway, this message is the reason why this code path is executed:

 {{{
 pulseaudio[14730]: (   3.543|   0.000) alsa-util.c: snd_pcm_avail()
 returned a value that is exceptionally large: 319392 bytes (1810 ms).
 pulseaudio[14730]: (   3.543|   0.000) alsa-util.c: Most likely this is a
 bug in the ALSA driver 'snd_emu10k1'. Please report this issue to the ALSA
 developers.
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c: snd_pcm_dump():
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c: Hardware PCM card 1
 'SB Live! 5.1 [SB0220]' device 0 subdevice 0
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c: Its setup is:
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   stream   :
 CAPTURE
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   access   :
 MMAP_INTERLEAVED
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   format   :
 S16_LE
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   subformat: STD
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   channels : 2
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   rate : 44100
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   exact rate   : 44100
 (44100/1)
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   msbits   : 16
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   buffer_size  : 4096
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   period_size  : 2048
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   period_time  : 46439
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   tstamp_mode  :
 ENABLE
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   period_step  : 1
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   avail_min: 2048
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   period_event : 1
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   start_threshold  :
 -1
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   stop_threshold   :
 4611686018427387904
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   silence_threshold: 0
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   silence_size : 0
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   boundary :
 4611686018427387904
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   appl_ptr : 49256
 pulseaudio[14730]: (   3.544|   0.000) alsa-util.c:   hw_ptr   :
 129104
 }}}

 So what to do? Should this ticket be closed? Should we remove the
 PA_DEBUG_TRAP line from check_left_to_play()?

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Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete

2010-10-16 Thread PulseAudio
 of silence already written to the
 hardware buffer, which means that it's possible to rewind 0.1 seconds.
 Your stream, however, does not know that there's only 0.1 seconds in the
 buffer. It should be informed about that, but the current implementation
 doesn't. Since it doesn't know that, it has to play safe, and assume that
 the whole hardware buffer can be rewound, i.e. 2 seconds.

 So, when the sink asks for more data, the stream notices that draining has
 been requested, but it checks the rewind buffer and sees that it's not
 empty - the rewind buffer contains all data from the last 2 seconds, and
 since the clip started 1.9 seconds ago and finished 1.4 seconds ago, it
 means that the stream must still wait 1.4 seconds until it can be sure
 that it's safe to send the drain acknowledgement.

 And so the sink fills the buffer again with silence and sleeps 1.9
 seconds. After that wait is over, the stream sees that the most recent
 data is 1.4 + 1.9 = 3.3 seconds old, which is more than the maximum rewind
 amount, so now it's safe to send the drain acknowledgement - 3.3 seconds
 late.

 That's how it works in principle - the calculations used a bunch of
 assumptions that aren't entirely correct, so don't take the numbers as
 exact, but as your experiments show, the extra delay is always measured in
 seconds, if the hw buffer is two seconds.

 So, who wants to implement the needed fixes?

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Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete

2010-10-15 Thread PulseAudio
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to
complete
-+--
  Reporter:  th  |   Owner:  lennart 
  Type:  defect  |  Status:  new 
 Milestone:  |   Component:  libpulse
Resolution:  |Keywords:  
-+--

Comment(by th):

 I ran the test program with pulse running with verbose logging, if you
 look at the Launchpad bug entry you can see it:

 [https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/660567] (also
 pasted below).

 It seems that pulse gets a buffer underrun at the time that I would expect
 pa_simple_drain() to return, but it doesn't, and waits for 2s after that.
 In simplistic terms, I guess pa_simple_drain() could be thought of as
 wait for buffer underrun, but for some reason it doesn't signal the
 client when the buffer runs out.

 {{{
 ...
 ( 0.034| 0.000) D: source.c: Processing rewind...
 [0.344269] play done
 [0.344328] not calling pa_simple_drain()
 [0.358380] playing 680 hz tone volume 50 for 500 ms
 [0.829451] play done
 [0.829489] not calling pa_simple_drain()
 [0.840411] playing 440 hz tone volume 50 for 500 ms
 [1.303775] play done
 ( 1.471| 1.437) D: protocol-native.c: Underrun on 'beep', 0 bytes in
 queue.
 [3.505551] pa_simple_drain() done
 [3.519491] playing 680 hz tone volume 50 for 500 ms
 ( 3.532| 2.060) D: protocol-native.c: Requesting rewind due to end of
 underrun.
 ( 3.532| 0.000) D: alsa-sink.c: Requested to rewind 10156 bytes.
 ( 3.532| 0.000) D: alsa-sink.c: Limited to 10156 bytes.
 ( 3.532| 0.000) D: alsa-sink.c: before: 2539
 ( 3.532| 0.000) D: alsa-sink.c: after: 2539
 ( 3.532| 0.000) D: alsa-sink.c: Rewound 10156 bytes.
 ( 3.532| 0.000) D: sink.c: Processing rewind...
 ( 3.532| 0.000) D: sink-input.c: Have to rewind 10156 bytes on render
 memblockq.
 ( 3.533| 0.000) D: source.c: Processing rewind...
 [3.848197] play done
 ( 4.016| 0.483) D: protocol-native.c: Underrun on 'beep', 0 bytes in
 queue.
 [6.050165] pa_simple_drain() done
 [6.062775] playing 440 hz tone volume 50 for 500 ms
 ...
 }}}

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Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te

2010-10-15 Thread PulseAudio
#790: pulsecore/svolume_arm.c fails to compile for armv5te
-+--
  Reporter:  ao2 |   Owner:  wtay
  Type:  defect  |  Status:  new 
 Milestone:  0.9.22  |   Component:  core
Resolution:  |Keywords:  
-+--
Changes (by dromi):

 * cc: alexandre.rela...@gmail.com (added)


Comment:

 You should be able to compile for a generic ARM architecture. It will
 compile this ARMv6 assembly code but it should not be called on an ARMv5
 processor.
 But since you want to test some patches, here are they ;)
 The two patches do the same, ie they remove this ARMv6 code if you compile
 for an ARMv5 specific architecture.
 In the first one, I use a macro that was created elsewhere but whose name
 is not explicit at all. The second one is a bit cleaner since the macro is
 renamed.
 I have tested the first one, not the second one.
 Any kind of feedback is welcome !

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Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te

2010-10-15 Thread PulseAudio
#790: pulsecore/svolume_arm.c fails to compile for armv5te
-+--
  Reporter:  ao2 |   Owner:  wtay
  Type:  defect  |  Status:  new 
 Milestone:  0.9.22  |   Component:  core
Resolution:  |Keywords:  
-+--

Comment(by flokli):

 Thanks for your patches! I applied the second one to the source code, and
 it compiled well on this device (see [http://pastebin.com/aj7VrDDm]).
 However, there are some ugly autoconf warnings...

 Will now test a build of pulseaudio with network/avahi support and post
 results...

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Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te

2010-10-15 Thread PulseAudio
#790: pulsecore/svolume_arm.c fails to compile for armv5te
-+--
  Reporter:  ao2 |   Owner:  wtay
  Type:  defect  |  Status:  new 
 Milestone:  0.9.22  |   Component:  core
Resolution:  |Keywords:  
-+--

Comment(by flokli):

 Ok I tested it, it works!
 I will need to experiment a bit with resampling methods and such because
 sound sometimes is stuttering (cpu a bit weak), but all in all everything
 works like it should :-)

 Can you commit this patch so that it will be in the next release?

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[pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults

2010-10-14 Thread PulseAudio
#865: Pulseaudio segfaults
--+-
 Reporter:  rabbit64  |   Owner:  lennart 
 Type:  defect|  Status:  new 
Milestone:|   Component:  module-combine-*
 Keywords:|  
--+-
 PA segfaults when combining analog stereo and digital output on my CT4780
 SBLive card. I have uploaded coredumps and corresponding messages to
 http://fi.muni.cz/~xsakalik/ .

 I have in default.pa
 {{{
 load-module module-combine
 slaves=alsa_output.pci-_05_00.0.iec958-stereo,alsa_output.pci-_05_00.0
 .analog-stereo resample_method=trivial sink_name=combined-trivial
 }}}

 daemon.conf:
 {{{
 daemonize = no
 allow-exit = no
 high-priority = yes
 nice-level = -11
 realtime-scheduling = yes
 realtime-priority = 5
 log-target = syslog
 log-level = debug
 log-time = yes
 }}}
 I have compiled pulseaudio from git (commit
 3de129f3ac8dd6cf51178b266837db4d5e4a1215) with this patch:

 {{{
 diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c
 index 1108a79..6de95b6 100644
 --- a/src/modules/alsa/alsa-sink.c
 +++ b/src/modules/alsa/alsa-sink.c
 @@ -59,7 +59,7 @@
  #include alsa-util.h
  #include alsa-sink.h

 -/* #define DEBUG_TIMING */
 +#define DEBUG_TIMING

  #define DEFAULT_DEVICE default

 diff --git a/src/modules/alsa/alsa-source.c b/src/modules/alsa/alsa-
 source.c
 index 5f12675..47f914b 100644
 --- a/src/modules/alsa/alsa-source.c
 +++ b/src/modules/alsa/alsa-source.c
 @@ -56,7 +56,7 @@
  #include alsa-util.h
  #include alsa-source.h

 -/* #define DEBUG_TIMING */
 +#define DEBUG_TIMING

  #define DEFAULT_DEVICE default
 }}}

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Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults

2010-10-14 Thread PulseAudio
#865: Pulseaudio segfaults
---+
  Reporter:  rabbit64  |   Owner:  lennart 
  Type:  defect|  Status:  new 
 Milestone:|   Component:  module-combine-*
Resolution:|Keywords:  
---+

Comment(by coling):

 Hi there. Thanks for the detailed report.

 Would you be able to get a backtrace from those coredumps? You should be
 able to load them in gdb and get the necessary backtraces from that. As
 you've compiled your own, the core dumps on their own will not really help
 remotes, but a nice detailed backtrace could help a lot.

 See the wiki:Community page for some bactrace tips.

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Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults

2010-10-14 Thread PulseAudio
#865: Pulseaudio segfaults
---+
  Reporter:  rabbit64  |   Owner:  lennart 
  Type:  defect|  Status:  new 
 Milestone:|   Component:  module-combine-*
Resolution:|Keywords:  
---+

Comment(by rabbit64):

 Hi, I've compiled it with -O2 so some debugging information are not there.
 I should have also said that two instances of pulseaudio (under different
 users) were running (this was under root). But I think it isn't a problem
 because it has segfaulted even when only one instance was running. I have
 uploaded backtraces on my webpage.

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Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete

2010-10-14 Thread PulseAudio
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to
complete
-+--
  Reporter:  th  |   Owner:  lennart 
  Type:  defect  |  Status:  new 
 Milestone:  |   Component:  libpulse
Resolution:  |Keywords:  
-+--
Changes (by th):

 * cc: tho...@horsten.com (added)
  * component:  daemon = libpulse


-- 
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Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete

2010-10-14 Thread PulseAudio
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to
complete
-+--
  Reporter:  th  |   Owner:  lennart 
  Type:  defect  |  Status:  new 
 Milestone:  |   Component:  libpulse
Resolution:  |Keywords:  
-+--

Comment(by th):

 Here's the test program output with automatic linewrap removed...:

 {{{
 $ ./beep
 [0.039147] playing 440 hz tone volume 50 for 500 ms
 [0.291107] play done
 [0.291116] not calling pa_simple_drain()
 [0.293467] playing 680 hz tone volume 50 for 500 ms
 [0.855440] play done
 [0.855459] not calling pa_simple_drain()
 [0.857808] playing 440 hz tone volume 50 for 500 ms
 [1.312107] play done
 [3.544309] pa_simple_drain() done
 [3.546687] playing 680 hz tone volume 50 for 500 ms
 [3.798973] play done
 [6.132183] pa_simple_drain() done
 [6.134535] playing 440 hz tone volume 50 for 500 ms
 [6.386808] play done
 [6.386824] not calling pa_simple_drain()
 [6.386851] playing 440 hz tone volume 0 for 500 ms
 [6.951772] play done
 [6.951789] not calling pa_simple_drain()
 [6.956659] playing 880 hz tone volume 50 for 1000 ms
 [7.912965] play done
 [10.144917] pa_simple_drain() done
 }}}

 And the gcc command was supposed to be:

 {{{
 gcc `pkg-config --cflags --libs libpulse-simple` -o beep beep.c
 }}}

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Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete

2010-10-14 Thread PulseAudio
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to
complete
-+--
  Reporter:  th  |   Owner:  lennart 
  Type:  defect  |  Status:  new 
 Milestone:  |   Component:  libpulse
Resolution:  |Keywords:  
-+--
Description changed by coling:

Old description:

 The libpulse-simple API has a function, pa_simple_drain() which is
 supposed to [w]ait until all data already written is played by the
 daemon.

 However with the version that ships with Ubuntu 10.10, it waits a *fair*
 bit more than that, approximately 2 seconds (see below). This is also the
 case for Fedora 14 according to a user on IRC who confirmed this issue,
 hence I am reporting it here.

 This means, that when I want to synchronize audio in my program ie.
 ensure that previously streamed audio has finished playing, e.g. before I
 start playing a new sound or just prior to exiting the program, there is
 a typically 2.2 second extra delay.

 Without this call, at the end of playback before exiting, the final sound
 will be clipped off at the end, since there is still unbuffered audio
 waiting to be sent, so it is needed to call this at the end of the
 program, but now causes an extra delay of several seconds before the
 program exits.

 I have attached a program that demonstrates this behaviour. Compile with:

 $ gcc `pkg-config --cflags --libs libpulse-simple` -o beep beep.c

 Here is the output from the program:

 $ ./beep
 [0.039147] playing 440 hz tone volume 50 for 500 ms
 [0.291107] play done
 [0.291116] not calling pa_simple_drain()
 [0.293467] playing 680 hz tone volume 50 for 500 ms
 [0.855440] play done
 [0.855459] not calling pa_simple_drain()
 [0.857808] playing 440 hz tone volume 50 for 500 ms
 [1.312107] play done
 [3.544309] pa_simple_drain() done
 [3.546687] playing 680 hz tone volume 50 for 500 ms
 [3.798973] play done
 [6.132183] pa_simple_drain() done
 [6.134535] playing 440 hz tone volume 50 for 500 ms
 [6.386808] play done
 [6.386824] not calling pa_simple_drain()
 [6.386851] playing 440 hz tone volume 0 for 500 ms
 [6.951772] play done
 [6.951789] not calling pa_simple_drain()
 [6.956659] playing 880 hz tone volume 50 for 1000 ms
 [7.912965] play done
 [10.144917] pa_simple_drain() done

 I have also reported this on Ubuntu's bug tracker:
 [https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/660567]

New description:

 The libpulse-simple API has a function, pa_simple_drain() which is
 supposed to [w]ait until all data already written is played by the
 daemon.

 However with the version that ships with Ubuntu 10.10, it waits a *fair*
 bit more than that, approximately 2 seconds (see below). This is also the
 case for Fedora 14 according to a user on IRC who confirmed this issue,
 hence I am reporting it here.

 This means, that when I want to synchronize audio in my program ie. ensure
 that previously streamed audio has finished playing, e.g. before I start
 playing a new sound or just prior to exiting the program, there is a
 typically 2.2 second extra delay.

 Without this call, at the end of playback before exiting, the final sound
 will be clipped off at the end, since there is still unbuffered audio
 waiting to be sent, so it is needed to call this at the end of the
 program, but now causes an extra delay of several seconds before the
 program exits.

 I have attached a program that demonstrates this behaviour. Compile with:

 $ gcc `pkg-config --cflags --libs libpulse-simple` -o beep beep.c

 Here is the output from the program:

 {{{
 $ ./beep
 [0.039147] playing 440 hz tone volume 50 for 500 ms
 [0.291107] play done
 [0.291116] not calling pa_simple_drain()
 [0.293467] playing 680 hz tone volume 50 for 500 ms
 [0.855440] play done
 [0.855459] not calling pa_simple_drain()
 [0.857808] playing 440 hz tone volume 50 for 500 ms
 [1.312107] play done
 [3.544309] pa_simple_drain() done
 [3.546687] playing 680 hz tone volume 50 for 500 ms
 [3.798973] play done
 [6.132183] pa_simple_drain() done
 [6.134535] playing 440 hz tone volume 50 for 500 ms
 [6.386808] play done
 [6.386824] not calling pa_simple_drain()
 [6.386851] playing 440 hz tone volume 0 for 500 ms
 [6.951772] play done
 [6.951789] not calling pa_simple_drain()
 [6.956659] playing 880 hz tone volume 50 for 1000 ms
 [7.912965] play done
 [10.144917] pa_simple_drain() done
 }}}

 I have also reported this on Ubuntu's bug tracker:
 [https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/660567]

--

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Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete

2010-10-14 Thread PulseAudio
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to
complete
-+--
  Reporter:  th  |   Owner:  lennart 
  Type:  defect  |  Status:  new 
 Milestone:  |   Component:  libpulse
Resolution:  |Keywords:  
-+--

Comment(by coling):

 Thanks for the test case. Replicated here also. Not sure what cases it,
 but hopefully I or someone else will find time to investigate soon.

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Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te

2010-10-14 Thread PulseAudio
#790: pulsecore/svolume_arm.c fails to compile for armv5te
-+--
  Reporter:  ao2 |   Owner:  wtay
  Type:  defect  |  Status:  new 
 Milestone:  0.9.22  |   Component:  core
Resolution:  |Keywords:  
-+--
Changes (by flokli):

 * cc: flo...@flokli.de (added)


Comment:

 I also see build failure on armv5tel-softfloat-linux-gnueabi
 gcc-4.4.4, glibc-2.11.2-r0, 2.6.36-rc7-00125-gd2a63db armv5tel

 in file pulsecore/svolume_arm.c with pulseaudio pulseaudio-0.9.21.2-r2.


 {{{
   CC libpulsecore_0.9.21_la-svolume_arm.lo
 {standard input}: Assembler messages:
 {standard input}:34: Error: selected processor does not support `ssat
 r0,#16,r0'
 {standard input}:48: Error: selected processor does not support `ssat
 r2,#16,r2'
 {standard input}:49: Error: selected processor does not support `ssat
 r3,#16,r3'
 {standard input}:50: Error: selected processor does not support `pkhbt
 r0,r3,r2,LSL#16'
 {standard input}:66: Error: selected processor does not support `ssat
 r2,#16,r2'
 {standard input}:67: Error: selected processor does not support `ssat
 r3,#16,r3'
 {standard input}:68: Error: selected processor does not support `ssat
 r4,#16,r4'
 {standard input}:69: Error: selected processor does not support `ssat
 r5,#16,r5'
 {standard input}:70: Error: selected processor does not support `pkhbt
 r0,r3,r2,LSL#16'
 {standard input}:71: Error: selected processor does not support `pkhbt
 r1,r5,r4,LSL#16'
 make[3]: *** [libpulsecore_0.9.21_la-svolume_arm.lo] Error 1
 }}}



 pasted build logs (link from ao2 doesn't work anymore)
 for -O2
 [http://pastebin.com/PTs1we1s]

 and -Os
 [http://pastebin.com/zSau03SJ]

 Having pulseaudio running on this device would be very nice (I want to use
 it as a music jukebox with network sound)

 Will test patches and stuff if needed :-)

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Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink

2010-10-12 Thread PulseAudio
#863: assign different device.description when more than one sink
+---
  Reporter:  skierpage  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  daemon 
Resolution: |Keywords: 
+---

Comment(by coling):

 Yeah in an ideal world I'd have done the renaming via that UI but it would
 require some mechanism (i.e. API) in Phonon which in turn I could proxy on
 to PA, so not as trivial as it may first seem.

 Chances are that something made PA crash (likely in the module-device-
 manager code that I wrote!) and it then autospawned again (as it's meant
 to do) but would lack the nice modules that are loaded at login via start-
 pulseaudio-x11 and start-pulseaudio-kde.

 KDE support is certainly a whole lot better than it was. I've still got
 more features to add and obviously fix up any bugs etc.

 Aside from the name of the devices do you experience any other major
 issues?

 I have tried the Kubuntu live CD and while IMO it would be better if the
 Ubuntu Pulseaudio had even more of the stable-queue fixes (I'm not 100%
 sure what they do and don't have but I'm pretty sure it's not the full
 compliment), it did seem to work fine for me, even with some exotic
 devices and bluetooth headsets etc. which is nice.

 Personally I find that Xine sucks with regards to it's PA output and still
 far prefer GStreamer backend for that, and hopefully VLC backend will
 prove more reliable in the future than the Xine one.

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Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink

2010-10-12 Thread PulseAudio
#863: assign different device.description when more than one sink
+---
  Reporter:  skierpage  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  daemon 
Resolution: |Keywords: 
+---

Comment(by coling):

 Oh incidentally, I noticed something you mentioned on my blog about the
 mute status being odd (PA apparently muting one of the two devices but
 kmix showing it as unmuted - to be honest this could be a bug in kmix - I
 did use names as an index at one point IIRC - will have to double check).
 The “deciding to mute” scenario is interesting. Not quite sure what the
 reason for that would be but I suspect it’s something non-obvious. I’d be
 very interested to get the “pacmd list” output for this muted device – and
 a screenshot from kmix at the same time when this misbehaviour is
 observed.

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Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink

2010-10-11 Thread PulseAudio
#863: assign different device.description when more than one sink
+---
  Reporter:  skierpage  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  daemon 
Resolution: |Keywords: 
+---

Comment(by skierpage):

 Thanks for responding!  pavucontrol is not on the Kubuntu CD.  Ideally
 System Settings  Multimedia  Phonon  preferred output dialog would let
 me right-click to rename outputs, which is another Launchpad bug to file.
 I updated bug #723 with my `lspci -vvv` info, hope it helps.

 I don't know why my fiddling in pacmd made Phonon and/or PulseAudio lose
 track of names and/or unload module-device-manager, if I can reproduce
 I'll file a Launchpad bug.

 You must be the Tolstoy who penned KDE + PulseAudio != Sucks; it sounds
 like PulseAudio and KDE work better in other distributions than Kubuntu,
 even though 10.10 made Phonon  Xine  PulseAudio the default. I don't
 know whether to upgrade to 10.10 or switch distros.

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Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink

2010-10-09 Thread PulseAudio
#863: assign different device.description when more than one sink
+---
  Reporter:  skierpage  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  daemon 
Resolution: |Keywords: 
+---

Comment(by coling):

 It's actually easier than that to change the description. If you run
 pavucontrol (assuming it has support for it which is not guaranteed), then
 you can just right click on a sink and select rename.

 The problem you get is really that both cards are being detected as being
 part of the the motherboard (i.e. built in).

 There is possibly something you can do with udev to stop this happening
 for one card, but I'm not sure and would have to read over the code which
 I can't do right this minute.

 Ultimately I think this is a duplicate of http://pulseaudio.org/ticket/723

 It doesn't seem to affect too many people overall (I've not hear anyone
 mention it for ages) but it's obviously still a problem.

 When you just see PulseAudio Sound Server in Phonon (note that it's NOT
 a backend, it's listed under devices) then it means that PA itself does
 not have the module-device-manager loaded. Simply running start-
 pulseaudio-kde will reload this module. It should be run automatically on
 login. There is a lot more information on the
 http://pulseaudio.org/wiki/KDE page.

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[pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink

2010-10-08 Thread PulseAudio
#863: assign different device.description when more than one sink
---+
 Reporter:  skierpage  |   Owner:  lennart
 Type:  defect |  Status:  new
Milestone: |   Component:  daemon 
 Keywords: |  
---+
 I'm trying to get sound working on Kubuntu 10.10 (KDE  Phonon  Xine 
 PulseAudio 0.9.21-63-gd3efa-dirty).  My computer has two sound outputs, a
 built-in VT8233/A/8235/8237 AC97 Audio Controller and a Creative Labs
 SB Audigy PCI card; the names in quotes are their device.product_names
 from `pacmd list-sinks`

 The problem is PA gives these sinks identical device.description strings,
 Internal Audio Analog Stereo.  These names appear in the UI of KDE's
 System Settings  Multimedia  Phonon  Backend , so it's impossible to
 tell which is which!

 It would be friendlier if PA disambiguated the devices by assigning
 different device.descriptions.  This sounds like bug #708 (unable to
 differentiate multiple identical soundcards), but in my case the sound
 cards aren't even the same.

 A workaround according to [wiki:Modules#Sinks] is to run `update-sink-
 proplist SINKNAME device.description=NEWNAME`.  That works for PA but it
 messed up KDE System Settings which now shows only PulseAudio Sound Server
 as a backend -- I probably have to restart KDE.

-- 
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[pulseaudio-tickets] [PulseAudio] #864: use GNU readline in pacmd

2010-10-08 Thread PulseAudio
#864: use GNU readline in pacmd
-+--
 Reporter:  skierpage|   Owner:  lennart
 Type:  enhancement  |  Status:  new
Milestone:   |   Component:  daemon 
 Keywords:   |  
-+--
 I'm using Kubuntu 10.10 RC (KDE  Phonon  Xine  PulseAudio 0.9.21-63
 -gd3efa-dirty).  `pacmd` is nice, but it's strange that up arrow and
 editing keys don't work in it, unlike most other command-line environments
 in Linux (ftp, gdb, python, etc.).

 I think the fix is to integrate GNU readline.  I dunno if the separation
 of pacmd from the pulseaudio server and module-cli-protocol-unix makes
 this harder or easier.

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #828: Flash causes PA to get into broken state; affects other apps

2010-10-07 Thread PulseAudio
#828: Flash causes PA to get into broken state; affects other apps
-+--
  Reporter:  lukehutch   |   Owner:  lennart
  Type:  defect  |  Status:  closed 
 Milestone:  |   Component:  daemon 
Resolution:  lackofresponse  |Keywords: 
-+--
Changes (by lukehutch):

  * status:  new = closed
  * resolution:  = lackofresponse


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[pulseaudio-tickets] [PulseAudio] #862: snd_pcm_avail_delay() returned strange values

2010-10-06 Thread PulseAudio
#862: snd_pcm_avail_delay() returned strange values
+---
 Reporter:  oniram  |   Owner:  lennart  
 Type:  defect  |  Status:  new  
Milestone:  |   Component:  module-alsa-*
 Keywords:  |  
+---
 I have a Dell Latitude E6410 and sound is working fine with pulseaudio.
 However, I get some error messages when I start pulseaudio:

 $ pulseaudio
 E: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is
 less than avail 24.
 E: alsa-util.c: Most likely this is a bug in the ALSA driver
 'snd_hda_intel'. Please report this issue to the ALSA developers.
 E: alsa-util.c: snd_pcm_dump():
 E: alsa-util.c: Soft volume PCM
 E: alsa-util.c: Control: PCM Playback Volume
 E: alsa-util.c: min_dB: -51
 E: alsa-util.c: max_dB: 0
 E: alsa-util.c: resolution: 256
 E: alsa-util.c: Its setup is:
 E: alsa-util.c:   stream   : CAPTURE
 E: alsa-util.c:   access   : MMAP_INTERLEAVED
 E: alsa-util.c:   format   : S16_LE
 E: alsa-util.c:   subformat: STD
 E: alsa-util.c:   channels : 2
 E: alsa-util.c:   rate : 44100
 E: alsa-util.c:   exact rate   : 44100 (44100/1)
 E: alsa-util.c:   msbits   : 16
 E: alsa-util.c:   buffer_size  : 16384
 E: alsa-util.c:   period_size  : 8192
 E: alsa-util.c:   period_time  : 185759
 E: alsa-util.c:   tstamp_mode  : ENABLE
 E: alsa-util.c:   period_step  : 1
 E: alsa-util.c:   avail_min: 15502
 E: alsa-util.c:   period_event : 0
 E: alsa-util.c:   start_threshold  : -1
 E: alsa-util.c:   stop_threshold   : 4611686018427387904
 E: alsa-util.c:   silence_threshold: 0
 E: alsa-util.c:   silence_size : 0
 E: alsa-util.c:   boundary : 4611686018427387904
 E: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice
 0
 E: alsa-util.c: Its setup is:
 E: alsa-util.c:   stream   : CAPTURE
 E: alsa-util.c:   access   : MMAP_INTERLEAVED
 E: alsa-util.c:   format   : S16_LE
 E: alsa-util.c:   subformat: STD
 E: alsa-util.c:   channels : 2
 E: alsa-util.c:   rate : 44100
 E: alsa-util.c:   exact rate   : 44100 (44100/1)
 E: alsa-util.c:   msbits   : 16
 E: alsa-util.c:   buffer_size  : 16384
 E: alsa-util.c:   period_size  : 8192
 E: alsa-util.c:   period_time  : 185759
 E: alsa-util.c:   tstamp_mode  : ENABLE
 E: alsa-util.c:   period_step  : 1
 E: alsa-util.c:   avail_min: 15502
 E: alsa-util.c:   period_event : 0
 E: alsa-util.c:   start_threshold  : -1
 E: alsa-util.c:   stop_threshold   : 4611686018427387904
 E: alsa-util.c:   silence_threshold: 0
 E: alsa-util.c:   silence_size : 0
 E: alsa-util.c:   boundary : 4611686018427387904
 E: alsa-util.c:   appl_ptr : 15528
 E: alsa-util.c:   hw_ptr   : 15528

 As suggested by the messages I reported the bug to ALSA. Here's the bug
 report:
 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5154

 The ALSA developers seemed to suggest that I get pulseaudio developers
 involved ... so here we are.

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #862: snd_pcm_avail_delay() returned strange values

2010-10-06 Thread PulseAudio
#862: snd_pcm_avail_delay() returned strange values
-+--
  Reporter:  oniram  |   Owner:  lennart  
  Type:  defect  |  Status:  new  
 Milestone:  |   Component:  module-alsa-*
Resolution:  |Keywords:   
-+--

Comment(by oniram):

 Let me try to fix that formatting mess:

 {{{
 E: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is
 less than avail 24.
 E: alsa-util.c: Most likely this is a bug in the ALSA driver
 'snd_hda_intel'. Please report this issue to the ALSA developers.
 E: alsa-util.c: snd_pcm_dump():
 E: alsa-util.c: Soft volume PCM
 E: alsa-util.c: Control: PCM Playback Volume
 E: alsa-util.c: min_dB: -51
 E: alsa-util.c: max_dB: 0
 E: alsa-util.c: resolution: 256
 E: alsa-util.c: Its setup is:
 E: alsa-util.c:   stream   : CAPTURE
 E: alsa-util.c:   access   : MMAP_INTERLEAVED
 E: alsa-util.c:   format   : S16_LE
 E: alsa-util.c:   subformat: STD
 E: alsa-util.c:   channels : 2
 E: alsa-util.c:   rate : 44100
 E: alsa-util.c:   exact rate   : 44100 (44100/1)
 E: alsa-util.c:   msbits   : 16
 E: alsa-util.c:   buffer_size  : 16384
 E: alsa-util.c:   period_size  : 8192
 E: alsa-util.c:   period_time  : 185759
 E: alsa-util.c:   tstamp_mode  : ENABLE
 E: alsa-util.c:   period_step  : 1
 E: alsa-util.c:   avail_min: 15502
 E: alsa-util.c:   period_event : 0
 E: alsa-util.c:   start_threshold  : -1
 E: alsa-util.c:   stop_threshold   : 4611686018427387904
 E: alsa-util.c:   silence_threshold: 0
 E: alsa-util.c:   silence_size : 0
 E: alsa-util.c:   boundary : 4611686018427387904
 E: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice
 0
 E: alsa-util.c: Its setup is:
 E: alsa-util.c:   stream   : CAPTURE
 E: alsa-util.c:   access   : MMAP_INTERLEAVED
 E: alsa-util.c:   format   : S16_LE
 E: alsa-util.c:   subformat: STD
 E: alsa-util.c:   channels : 2
 E: alsa-util.c:   rate : 44100
 E: alsa-util.c:   exact rate   : 44100 (44100/1)
 E: alsa-util.c:   msbits   : 16
 E: alsa-util.c:   buffer_size  : 16384
 E: alsa-util.c:   period_size  : 8192
 E: alsa-util.c:   period_time  : 185759
 E: alsa-util.c:   tstamp_mode  : ENABLE
 E: alsa-util.c:   period_step  : 1
 E: alsa-util.c:   avail_min: 15502
 E: alsa-util.c:   period_event : 0
 E: alsa-util.c:   start_threshold  : -1
 E: alsa-util.c:   stop_threshold   : 4611686018427387904
 E: alsa-util.c:   silence_threshold: 0
 E: alsa-util.c:   silence_size : 0
 E: alsa-util.c:   boundary : 4611686018427387904
 E: alsa-util.c:   appl_ptr : 15528
 E: alsa-util.c:   hw_ptr   : 15528
 }}}

-- 
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[pulseaudio-tickets] [PulseAudio] #861: audio stops working with multiple users

2010-10-03 Thread PulseAudio
#861: audio stops working with multiple users
---+
 Reporter:  sergei |   Owner:  lennart
 Type:  defect |  Status:  new
Milestone: |   Component:  daemon 
 Keywords:  stops working, multiple users  |  
---+
 I have discovered that when I switch to another user in Mint 9 (gnome
 edition) the audio stops working until I reboot the computer.

-- 
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[pulseaudio-tickets] [PulseAudio] #860: No audio through speakers on Dell M1330 Laptop

2010-10-01 Thread PulseAudio
#860: No audio through speakers on Dell M1330 Laptop
-+--
 Reporter:  jjardon  |   Owner:  lennart
 Type:  defect   |  Status:  new
Milestone:   |   Component:  daemon 
 Keywords:   |  
-+--
 I have no sound in my laptop. It only works if I choose Analog Surround
 4.0 in sound preferences.

-- 
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Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices

2010-09-30 Thread PulseAudio
#706: [PATCH] Automatically switch to new devices
--+-
  Reporter:  mterry   |   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  daemon 
Resolution:   |Keywords: 
--+-

Comment(by mgedmin):

 Two questions:

 * If a new network sink appears, will this module switch to it?  That
 would be undesirable. (Use case: I'm listening to music on my laptop via
 headphones when someone reboots the shared office jukebox machine.)

 * If I switch an audio stream to a different device (say, the above-
 mentioned network sink), and then I switch it back to my laptop's current
 default device (internal audio) -- will that keep the save_device flag set
 and prevent it from moving to a USB headset?  That would be inconvenient.

-- 
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[pulseaudio-tickets] [PulseAudio] #859: racy crashes

2010-09-29 Thread PulseAudio
#859: racy crashes
---+
 Reporter:  jankratochvil  |   Owner:  lennart
 Type:  defect |  Status:  new
Milestone: |   Component:  clients
 Keywords: |  
---+
 Goal: Five rows in Tetris: http://js1k.com.nyud.net:8080/demo/730

 firefox-3.6.10-1.fc13.x86_64
 pulseaudio-0.9.21-6.fc13.x86_64

 It crashes in 1 to 5 seconds.

 You must run no debugging tools as they reduce threading raciness.
 Happenning on i7-920 (4 cores + multithreading = 8 cores).

 Finally making so many rows in Tetris now!

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Re: [pulseaudio-tickets] [PulseAudio] #843: Some PA strings are not translated in the Gnome Sound Properties window

2010-09-24 Thread PulseAudio
#843: Some PA strings are not translated in the Gnome Sound Properties window
---+
  Reporter:  kelemeng  |   Owner:  lennart  
  Type:  defect|  Status:  new  
 Milestone:|   Component:  module-alsa-*
Resolution:|Keywords:  i18n 
---+

Comment(by kelemeng):

 Ping?

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Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices

2010-09-23 Thread PulseAudio
#706: [PATCH] Automatically switch to new devices
--+-
  Reporter:  mterry   |   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  daemon 
Resolution:   |Keywords: 
--+-

Comment(by mterry):

 I just updated a more recent version of the patch (against 0.9.22).  This
 one drops the intended-roles logic, as it didn't work right (BT devices
 had roles set, as I commented above).  It also adds logic to not switch to
 ISA or PCI devices in case they show up for some reason after the module
 loads.

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Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices

2010-09-23 Thread PulseAudio
#706: [PATCH] Automatically switch to new devices
--+-
  Reporter:  mterry   |   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  daemon 
Resolution:   |Keywords: 
--+-

Comment(by coling):

 Hi there,

 On first glance the module looks fine. It may ultimately go through some
 changes once I've finished some other work but as I've been particularly
 tardy about that I see no reason to wait for that before this goes in!

 Couple points re the patch itself:
  1. It's (c) Canonical. Do you work for them or have you just copied it
 from elsewhere and forgot to update it?
  2. If you want to retain authorship can you post your full name and email
 address (for the git commit feel free to obfuscate email address here but
 it'll go into the git history so will likely be listed on the web
 somewhere). If prefer to remain anonymous, that's fine too :)

 Cheers

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Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices

2010-09-23 Thread PulseAudio
#706: [PATCH] Automatically switch to new devices
--+-
  Reporter:  mterry   |   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  daemon 
Resolution:   |Keywords: 
--+-

Comment(by mterry):

 Uh, that was at canonical.com.  Trac tried to help me out there.

-- 
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[pulseaudio-tickets] [PulseAudio] #858: Pulseaudio uses all CPU on ALSA problems

2010-09-22 Thread PulseAudio
#858: Pulseaudio uses all CPU on ALSA problems
---+
 Reporter:  LCID Fire  |   Owner:  lennart
 Type:  defect |  Status:  new
Milestone: |   Component:  daemon 
 Keywords: |  
---+
 Using Pulseaudio with a USB device I frequently get dmesg like:
 cannot submit datapipe for urb 0, error -28: not enough bandwidth

 after which Pulseaudio goes to 100% CPU utilization and stays there. The
 way to get the process to behave again is to kill it.

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Re: [pulseaudio-tickets] [PulseAudio] #799: libpulse use of Xlib is not thread-safe

2010-09-20 Thread PulseAudio
#799: libpulse use of Xlib is not thread-safe
+---
  Reporter:  courmisch  |   Owner:  coling  
  Type:  defect |  Status:  new 
 Milestone: |   Component:  module-x11-*
Resolution: |Keywords:  
+---
Changes (by coling):

  * owner:  lennart = coling


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[pulseaudio-tickets] [PulseAudio] #857: Gstreamer audio glitches when playback unpaused

2010-09-18 Thread PulseAudio
#857: Gstreamer audio glitches when playback unpaused
+---
 Reporter:  ettlz   |   Owner:  lennart
 Type:  defect  |  Status:  new
Milestone:  |   Component:  daemon 
 Keywords:  |  
+---
 The audio from Gstreamer-based applications (tested with Rhythmbox and
 Totem) does not unpause cleanly. Typically there is a glitch a fraction of
 a second into the unpaused audio. This is on Intel HDA audio with a
 ALC-883 codec.

 pulseaudio-0.9.21-6.fc13.x86_64
 rhythmbox-0.12.8-4.fc13.x86_64
 totem-2.30.2-1.fc13.x86_64
 gstreamer-0.10.30-1.fc13.x86_64
 gstreamer-0.10.30-1.fc13.i686
 gstreamer-plugins-good-0.10.25-1.fc13.x86_64

 Attached is the output of pulseaudio -.

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Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free

2010-09-14 Thread PulseAudio
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly
without glitch-free
-+--
  Reporter:  erich   |   Owner:  erich  
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  module-rtp-*   
Resolution:  |Keywords:  latency,glitch-free
-+--
Changes (by jw):

 * cc: devnull.pulseau...@molb.org (added)


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Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free

2010-09-14 Thread PulseAudio
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly
without glitch-free
-+--
  Reporter:  erich   |   Owner:  erich  
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  module-rtp-*   
Resolution:  |Keywords:  latency,glitch-free
-+--

Comment(by coling):

 jw, are you using all the patches on stable-queue git branch (which is
 0.9.21 + lots of fixes)? There are some rtp related fixes there that fix
 header parsing that could have previously broken things. Not certainly
 they'll fix this issue, but all the same it's worth trying.

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Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free

2010-09-14 Thread PulseAudio
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly
without glitch-free
-+--
  Reporter:  erich   |   Owner:  erich  
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  module-rtp-*   
Resolution:  |Keywords:  latency,glitch-free
-+--

Comment(by jw):

 Replying to [comment:4 coling]:
 You mean commit 678f12d? I will check that out and report afterwards!

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Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free

2010-09-14 Thread PulseAudio
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly
without glitch-free
-+--
  Reporter:  erich   |   Owner:  erich  
  Type:  defect  |  Status:  new
 Milestone:  |   Component:  module-rtp-*   
Resolution:  |Keywords:  latency,glitch-free
-+--

Comment(by jw):

 I updated pulseaudio (both sender and receiver), but the speed still
 changes pretty often and randomly (LAN connection). With vlc on the
 receiver side it plays normal...

-- 
Ticket URL: http://pulseaudio.org/ticket/477#comment:6
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Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)

2010-09-08 Thread PulseAudio
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient
network problem (networking / virtualization)
--+-
  Reporter:  peter|   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  clients
Resolution:   |Keywords:  network virtual
--+-

Comment(by coling):

 @gauss: Just don't talk directly to the remote side. Always create a local
 PA daemon even if there is no physical h/w. Pulseaudio automatically loads
 a Null sink when no real hardware is present (I could easily make it do
 the same with sources if there were a null source module!).

 By always talking to the local PA daemon there is a much reduced
 likelihood of any problems. You can then load a tunnel sink to the remote
 machine and output to the right place as you desire. If there are any
 network issues, the sink will unload (TCP connection broken) and the auto-
 null sink will kick in (i.e. exactly what you suggest).

-- 
Ticket URL: http://pulseaudio.org/ticket/194#comment:4
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Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)

2010-09-08 Thread PulseAudio
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient
network problem (networking / virtualization)
--+-
  Reporter:  peter|   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  clients
Resolution:   |Keywords:  network virtual
--+-

Comment(by coling):

 So you are suggesting that we essentially just accept that PA crashes
 very often and create some kind of wrapper system to deal with this?
 This seems fundamentally wrong: why not just fix the crashes?

 And I really don't think that PA itself crashes very often anyway. I've
 not seen any crash related tickets for a long time. If you have any back
 traces available, I'll be happy to take a look, but without some kind of
 information about the crashes you refer to, there is little we can do to
 help fix them.

 Now some kind of auto-reconnect logic is not necessarily a bad idea (I
 would find it useful when debugging) but this is really something up to
 the individual clients to implement. I've implemented such features in the
 various clients I've written, so really other clients need to do the same.

-- 
Ticket URL: http://pulseaudio.org/ticket/194#comment:6
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Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)

2010-09-08 Thread PulseAudio
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient
network problem (networking / virtualization)
--+-
  Reporter:  peter|   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  clients
Resolution:   |Keywords:  network virtual
--+-

Comment(by gauss-gs):

 We just mean that pulseaudio CAN crash (of course crashes need to fix) but
 it need to give smoother user experience by default(if pulseaudio stated
 as a consumer, not а professional sound server), with allow (but not
 force) client developers to deal with reconnections.

-- 
Ticket URL: http://pulseaudio.org/ticket/194#comment:7
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[pulseaudio-tickets] [PulseAudio] #856: found existing unlinked compatible pad pulsesink0:sink

2010-09-08 Thread PulseAudio
#856: found existing unlinked compatible pad pulsesink0:sink
--+-
 Reporter:  ganeshpetkar  |   Owner:  lennart  
 Type:  defect|  Status:  new  
Milestone:|   Component:  gst-pulse
 Keywords:|  
--+-
 Hi,
 i m using below gst pipeline

 gst-launch filesrc location=/home/Test.wav ! wavparse ! audioconvert !
 audioresample ! pulsesink

 find the log message after execution

 0:00:21.157652439   996   0x416050 DEBUG   GST_ELEMENT_PADS
 gstutils.c:1671:gst_element_link_pads: looping through allowed src and
 dest pads
 0:00:21.158231708   996   0x416050 DEBUG   GST_ELEMENT_PADS
 gstutils.c:1674:gst_element_link_pads: trying src pad audioresample0:src
 0:00:21.158841464   996   0x416050 DEBUG   GST_ELEMENT_PADS
 gstutils.c::gst_element_get_compatible_pad: finding pad in pulsesink0
 compatible with audioresample0:src
 0:00:21.159481708   996   0x416050 LOG GST_ELEMENT_PADS
 gstutils.c:1130:gst_element_get_compatible_pad: examining pad
 pulsesink0:sink
 0:00:21.178048781   996   0x416050 DEBUG   GST_ELEMENT_PADS
 gstutils.c:1155:gst_element_get_compatible_pad: found existing unlinked
 compatible pad pulsesink0:sink
 0:00:21.196920732   996   0x416050 DEBUG   GST_ELEMENT_PADS
 gstutils.c:1688:gst_element_link_pads: linked pad audioresample0:src to
 pad pulsesink0:sink
 Setting pipeline to PAUSED ...
 Illegal instruction


 I m using
  pulseaudio-0.9.21
  gst-plugins-good_0.10.23
  gst-plugin-pulse_0.9.7


 Please do the needful.

 Regards,
 ganesh

-- 
Ticket URL: http://pulseaudio.org/ticket/856
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Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)

2010-09-07 Thread PulseAudio
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient
network problem (networking / virtualization)
--+-
  Reporter:  peter|   Owner:  lennart
  Type:  enhancement  |  Status:  new
 Milestone:   |   Component:  clients
Resolution:   |Keywords:  network virtual
--+-
Changes (by gauss-gs):

 * cc: gauss...@gmail.com (added)


Comment:

 I think small sound gap is VERY MUCH better than -KILL-ing and restarting
 freezed apps (damn skype won the 1st prize for this, second is padsp/alsa-
 pulse :-!) while pulseaudio crash or network disconnect happens. Maybe
 just make sound go /dev/null until connection is established by default
 and let app override this behavior if needed? This is definitely need to
 fix.

-- 
Ticket URL: http://pulseaudio.org/ticket/194#comment:3
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[pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs

2010-09-03 Thread PulseAudio
#852: Pulseaudio doesn't map multiple inputs/outputs
---+
 Reporter:  LCID Fire  |   Owner:  lennart
 Type:  defect |  Status:  new
Milestone: |   Component:  daemon 
 Keywords: |  
---+
 I have a soundcard with multiple inputs and outputs.

 {{{
 cat /proc/asound/devices
 }}}

 shows

 {{{
  13: [ 2- 0]: digital audio playback
  14: [ 2- 0]: digital audio capture
  15: [ 2]   : control
  16: [ 2- 1]: digital audio playback
  17: [ 2- 1]: digital audio capture
  18: [ 2- 2]: digital audio capture
 }}}
 But in Pulseaudio only the first capture and playback get mapped. The
 second is not accessible via Pulseaudio.

-- 
Ticket URL: http://pulseaudio.org/ticket/852
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[pulseaudio-tickets] [PulseAudio] #853: Illegal instruction..

2010-09-03 Thread PulseAudio
#853: Illegal instruction..
--+-
 Reporter:  ganeshpetkar  |   Owner:  lennart  
 Type:  tracking  |  Status:  new  
Milestone:  0.9.22|   Component:  gst-pulse
 Keywords:|  
--+-
 Hi,
 i m trying to play wav file through gstreamer and pulseaudio as a sink but
 it is giving error as '''Illegal instruction'''
 But i m not able to weather my pipe line is correct or not.
 Find the pipe line and log messages

 gst-launch filesrc location=/home/test.wav ! wavparse ! audioconvert !
 audioresample ! pulsesink


 0:00:20.388993903  1008   0x416050 LOG  GST_ELEMENT_FACTORY
 gstelementfactory.c:441:gst_element_factory_make:elementfactory37 found
 factory 0x434d10
 0:00:20.488414635  1008   0x416050 DEBUGGST_ELEMENT_FACTORY
 gstelementfactory.c:233:gst_element_register:registry0 update existing
 feature 0x434d10 (pulsesink)
 0:00:20.490823171  1008   0x416050 DEBUGGST_ELEMENT_FACTORY
 gstelementfactory.c:233:gst_element_register:registry0 update existing
 feature 0x434da0 (pulsesrc)
 0:00:20.492621952  1008   0x416050 DEBUGGST_ELEMENT_FACTORY
 gstelementfactory.c:233:gst_element_register:registry0 update existing
 feature 0x434e30 (pulsemixer)
 0:00:20.493628050  1008   0x416050 INFO GST_ELEMENT_FACTORY
 gstelementfactory.c:363:gst_element_factory_create: creating element
 pulsesink
 0:00:20.504664635  1008   0x416050 INFOGST_ELEMENT_PADS
 gstelement.c:698:gst_element_add_pad:gstbases...@0x472078 adding pad
 'sink'
 Illegal instruction

-- 
Ticket URL: http://pulseaudio.org/ticket/853
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[pulseaudio-tickets] [PulseAudio] #854: Illegal instruction..

2010-09-03 Thread PulseAudio
#854: Illegal instruction..
--+-
 Reporter:  ganeshpetkar  |   Owner:  lennart  
 Type:  tracking  |  Status:  new  
Milestone:  0.9.22|   Component:  gst-pulse
 Keywords:|  
--+-
 Hi,
 i m trying to play wav file through gstreamer and pulseaudio as a sink but
 it is giving error as '''Illegal instruction'''
 But i m not able to weather my pipe line is correct or not.
 Find the pipe line and log messages

 gst-launch filesrc location=/home/test.wav ! wavparse ! audioconvert !
 audioresample ! pulsesink


 0:00:20.388993903  1008   0x416050 LOG  GST_ELEMENT_FACTORY
 gstelementfactory.c:441:gst_element_factory_make:elementfactory37 found
 factory 0x434d10
 0:00:20.488414635  1008   0x416050 DEBUGGST_ELEMENT_FACTORY
 gstelementfactory.c:233:gst_element_register:registry0 update existing
 feature 0x434d10 (pulsesink)
 0:00:20.490823171  1008   0x416050 DEBUGGST_ELEMENT_FACTORY
 gstelementfactory.c:233:gst_element_register:registry0 update existing
 feature 0x434da0 (pulsesrc)
 0:00:20.492621952  1008   0x416050 DEBUGGST_ELEMENT_FACTORY
 gstelementfactory.c:233:gst_element_register:registry0 update existing
 feature 0x434e30 (pulsemixer)
 0:00:20.493628050  1008   0x416050 INFO GST_ELEMENT_FACTORY
 gstelementfactory.c:363:gst_element_factory_create: creating element
 pulsesink
 0:00:20.504664635  1008   0x416050 INFOGST_ELEMENT_PADS
 gstelement.c:698:gst_element_add_pad:gstbases...@0x472078 adding pad
 'sink'
 Illegal instruction


 Please do the needful.

 Regards,
 Ganesh

-- 
Ticket URL: http://pulseaudio.org/ticket/854
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[pulseaudio-tickets] [PulseAudio] #855: Illegal instruction..

2010-09-03 Thread PulseAudio
#855: Illegal instruction..
--+-
 Reporter:  ganeshpetkar  |   Owner:  lennart  
 Type:  tracking  |  Status:  new  
Milestone:  0.9.22|   Component:  gst-pulse
 Keywords:|  
--+-
 Hi,
 i m trying to play wav file through gstreamer and pulseaudio as a sink but
 it is giving error as '''Illegal instruction'''
 But i m not able to weather my pipe line is correct or not.
 But if i use alsa as sink it is working fine.
 Find the pipe line and log messages

 gst-launch filesrc location=/home/test.wav ! wavparse ! audioconvert !
 audioresample ! pulsesink




 Please do the needful.

 Regards,
 Ganesh

-- 
Ticket URL: http://pulseaudio.org/ticket/855
PulseAudio http://pulseaudio.org/
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Re: [pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs

2010-09-03 Thread PulseAudio
#852: Pulseaudio doesn't map multiple inputs/outputs
+---
  Reporter:  LCID Fire  |   Owner:  lennart
  Type:  defect |  Status:  new
 Milestone: |   Component:  daemon 
Resolution: |Keywords: 
+---

Comment(by coling):

 Please attach output from {{{pacmd list}}}

-- 
Ticket URL: http://pulseaudio.org/ticket/852#comment:1
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Re: [pulseaudio-tickets] [PulseAudio] #854: Illegal instruction..

2010-09-03 Thread PulseAudio
#854: Illegal instruction..
---+
  Reporter:  ganeshpetkar  |   Owner:  lennart  
  Type:  tracking  |  Status:  closed   
 Milestone:  0.9.22|   Component:  gst-pulse
Resolution:  duplicate |Keywords:   
---+
Changes (by coling):

  * status:  new = closed
  * resolution:  = duplicate


Comment:

 Dupe of #853

-- 
Ticket URL: http://pulseaudio.org/ticket/854#comment:1
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