Re: [pulseaudio-discuss] [PATCH 5/6] alsa: create ucm profiles
Hi David,
On Wed, Apr 6, 2011 at 1:41 AM, David Henningsson
david.hennings...@canonical.com wrote:
On 2011-04-05 22:34, Margarita Olaya wrote:
+static pa_hook_result_t jack_insert_new_hook_callback(pa_core *c,
struct userdata *u) {
+ struct pa_alsa_ucm_config *ucm =u-ucm;
+ struct profile_data *d;
+ char *device_name;
+
+ pa_assert(u);
+
+ d = PA_CARD_PROFILE_DATA(u-card-active_profile);
+
+ pa_log_debug(Jack insert new hook callback);
+
+ device_name = strchr(d-profile-name, ':') + 2;
+ if (!device_name) {
+ pa_log(no device found for %s, d-profile-name);
+ return PA_HOOK_CANCEL;
+ }
+
+ if (strcmp(device_name, Headset.0) == 0)
+ return PA_HOOK_OK;
+
+ /* Set headset.0 device per default */
+ if (snd_use_case_set(ucm-ucm_mgr, _enadev, Headset.0) 0) {
+ pa_log(failed to set device Headset.0);
+ return PA_HOOK_CANCEL;
+ }
+
+ return PA_HOOK_OK;
+}
About this function:
1) pa_hook_cb_t defines three pointers as input, but this one only takes
two. Is that really working on all platforms?
This is a mistake in this version, It is fixed now.
2) There is no matching against what card sent the event, is that irrelevant
for UCM?
The UCM data is store per sound, I'm looking into a way of passing the
card name to the hook and compare it with the one passed in the
userdata.
3) Is Headset.0 a reserved identifier in UCM, or is this unfinished test
code? What if there isn't a headset, just a separate external mic, or
external speakers connected to line out?
It depends on the driver to report here what was inserted or removed.
+
+static pa_hook_result_t jack_remove_new_hook_callback(pa_core *c,
struct userdata *u) {
+ struct profile_data *d;
+ struct pa_alsa_ucm_config *ucm =u-ucm;
+ char *device_name;
+
+ pa_assert(u);
+
+ d = PA_CARD_PROFILE_DATA(u-card-active_profile);
+
+ pa_log_debug(Jack removed new hook callback);
+
+ device_name = strchr(d-profile-name, ':') + 2;
+ if(!device_name) {
+ pa_log(no device found for %s, d-profile-name);
+ return PA_HOOK_CANCEL;
+ }
+
+ /* Set current profile device */
+ if (snd_use_case_set(ucm-ucm_mgr, _enadev, Headset.0) 0) {
+ pa_log(failed to set device Headset.0);
+ return PA_HOOK_CANCEL;
+ }
+
+ return PA_HOOK_OK;
+}
Ehm, seems like you set it to the Headset device both on plug and unplug...?
hmm, here we will switch to device_name gotten from the current
profile, in case of failure the device will be disabled.
I will post the changes soon.
Regards,
Margarita
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
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Re: [pulseaudio-discuss] intermittent audio
I'm not sure if this is the right venue, but I'm not sure where else to ask. If there's a more appropriate place, please let me know. I recently upgraded my MythTV server from Fedora 11 to Fedora 14. Now the only way to get sound to work is to follow these steps: Start the audio source so that there's an audio stream Open the Pulseaudio Volume Control panel Click the Configuration tab, Change Internal Audio to any profile other than the current one Change Internal Audio back to the original profile ( Digital Stereo Duplex (IEC958) ) At this point, the audio works. But if I change from watching one show in MythTV to another, I have to go through the whole process again. It's like as soon as pulseaudio loses a signal, it has to be reset. The issue is odd enough that I don't even know what keywords to search for to see if this is a common problem - has anyone heard of this? Does anyone have any suggestions for how to fix or troubleshoot it? rm -rf ~/.pulse* pulseaudio -vvv 2 log.txt pastebin log.txt and send us the link ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] LAC2011: Linux Audio Conference - Are you going?
'Twas brillig, and David Henningsson at 07/04/11 12:52 did gyre and gimble: On 2011-04-06 23:46, Kurt Taylor wrote: It looks like I might be able to go to LAC. I can see about scheduling a PulseAudio workgroup time and space with the LAC folks or nearby. If so, you should coordinate with Colin Guthrie (coling on IRC). So, I need a headcount ASAP, are you going? I'll be going, I'll arrive Thursday evening and leave Sunday afternoon. Ditto on the timings. Lennart is also going but not yet sure of his schedule. Would you be interested in a PulseAudio meetup and working session? Yes. If so, what agenda items? Everything :-) Continuation on what and when for master/PA 1.0. Pretty much what David said :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] LAC2011: Linux Audio Conference - Are you going?
On Thu, Apr 7, 2011 at 5:18 PM, Colin Guthrie gm...@colin.guthr.ie wrote: 'Twas brillig, and David Henningsson at 07/04/11 12:52 did gyre and gimble: On 2011-04-06 23:46, Kurt Taylor wrote: It looks like I might be able to go to LAC. I can see about scheduling a PulseAudio workgroup time and space with the LAC folks or nearby. If so, you should coordinate with Colin Guthrie (coling on IRC). So, I need a headcount ASAP, are you going? I'll be going, I'll arrive Thursday evening and leave Sunday afternoon. Ditto on the timings. Lennart is also going but not yet sure of his schedule. Would you be interested in a PulseAudio meetup and working session? Yes. If so, what agenda items? Everything :-) Continuation on what and when for master/PA 1.0. FWIW, I'll be there, giving a talk about PulseAudio on Mac OS X. Daniel ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pa_simple_free
more informations about our problem:
in thread-mainloop.c
void pa_threaded_mainloop_stop(pa_threaded_mainloop *m) {
...
// freeze here:
pa_mutex_lock(m-mutex);
...
}
what would be the cause? it stop at pa_mutex_lock(m-mutex). what can i
do to fix this problem? any advice is more than welcome.
pat
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Re: [pulseaudio-discuss] pa_simple_free
On Thu, Apr 7, 2011 at 7:15 PM, patrick pured...@11h11.com wrote:
more informations about our problem:
in thread-mainloop.c
void pa_threaded_mainloop_stop(pa_threaded_mainloop *m) {
...
// freeze here:
pa_mutex_lock(m-mutex);
...
}
what would be the cause? it stop at pa_mutex_lock(m-mutex). what can i do
to fix this problem? any advice is more than welcome.
Are your calling pa_threaded_mainloop_stop() while the mainloop is
locked? IOW, did you call pa_threaded_mainloop_lock() without
pa_threaded_mainloop_unlock() before?
Quoting the documentation:
void pa_threaded_mainloop_stop( pa_threaded_mainloop *
m )
Terminate the event loop thread cleanly.
Make sure to unlock the mainloop object before calling this function.
Daniel
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Re: [pulseaudio-discuss] pa_simple_free
Does the parec-simple example work for you (it's part of the PulseAudio source tree). Hi Daniel, For first of all thank you for taking the time to help me track down our problem. parec-simple is working (nicely quitting). So the problem must be in this small code: http://pastebin.ubuntu.com/590602/ the thing is, all the other audio driver (portaudio, alsa, oss, esd) are working fine with the same code. but pulseaudio stop at pa_mutex_lock(m-mutex); does it means that 2 thread are conflicting. like you can read, i am not a good coder at all, i don't understand the reason of pa_mutex_lock. do you think that switching to the full API might help? is it possible to use the same kind of code / call (our audio read function is call, read the recording source and return the buffer size. pat Daniel ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pa_simple_free
Hi, On Fri, 2011-04-08 at 03:15 +0900, patrick wrote: Does the parec-simple example work for you (it's part of the PulseAudio source tree). Hi Daniel, For first of all thank you for taking the time to help me track down our problem. parec-simple is working (nicely quitting). So the problem must be in this small code: http://pastebin.ubuntu.com/590602/ I modified your code to compile independently and it works fine here. http://paste.pocoo.org/show/367406/ Are you using the API differently from this in your code? -- Arun ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] 2 minor issues noticed with padsp
Hi,
I have a really old application that uses oss interfaces, and noticed that
when attempting to run via padsp it would fail ( as opposed to with aoss ).
After talking with Ford_Prefect in #pulseaudio, we narrowed down the problem
to protocol-native.c
Attached is a patch which allows padsp's get_sink_info call to success on
the server side ( drop-invalid-check ).
He mentioned that name is allowed to be null; which makes me think it might
be less confusing to just do a *name check and then no others.
Additionally, I noticed that in padsp.c it was recieving two callbacks in
both sink and source_info_cb. As this appears to be the desired behavior in
introspect.c, I modified the _cb functions to only notify if success was not
already set.
-drew
diff -udPr --exclude=orig pulseaudio-0.9.21/orig/src/pulsecore/protocol-native.c pulseaudio-0.9.21/src/pulsecore/protocol-native.c
--- pulseaudio-0.9.21/orig/src/pulsecore/protocol-native.c 2009-11-22 22:57:07.0 -0500
+++ pulseaudio-0.9.21/src/pulsecore/protocol-native.c 2011-04-07 16:28:09.0 -0400
@@ -3125,7 +3125,6 @@
(command == PA_COMMAND_GET_SOURCE_INFO
pa_namereg_is_valid_name_or_wildcard(name, PA_NAMEREG_SOURCE)) ||
pa_namereg_is_valid_name(name), tag, PA_ERR_INVALID);
-CHECK_VALIDITY(c-pstream, idx != PA_INVALID_INDEX || name, tag, PA_ERR_INVALID);
CHECK_VALIDITY(c-pstream, idx == PA_INVALID_INDEX || !name, tag, PA_ERR_INVALID);
CHECK_VALIDITY(c-pstream, !name || idx == PA_INVALID_INDEX, tag, PA_ERR_INVALID);
diff -udPr --exclude=orig pulseaudio-0.9.21/orig/src/utils/padsp.c pulseaudio-0.9.21/src/utils/padsp.c
--- pulseaudio-0.9.21/orig/src/utils/padsp.c 2009-09-18 16:21:44.0 -0400
+++ pulseaudio-0.9.21/src/utils/padsp.c 2011-04-07 17:14:26.0 -0400
@@ -1205,8 +1205,8 @@
fd_info *i = userdata;
if (!si || eol 0) {
-i-operation_success = 0;
-pa_threaded_mainloop_signal(i-mainloop, 0);
+ if( ! i-operation_success )
+ pa_threaded_mainloop_signal(i-mainloop, 0);
return;
}
@@ -1227,8 +1227,8 @@
fd_info *i = userdata;
if (!si || eol 0) {
-i-operation_success = 0;
-pa_threaded_mainloop_signal(i-mainloop, 0);
+ if( ! i-operation_success )
+ pa_threaded_mainloop_signal(i-mainloop, 0);
return;
}
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Re: [pulseaudio-discuss] intermittent audio
'Twas brillig, and pl bossart at 07/04/11 15:48 did gyre and gimble: I'm not sure if this is the right venue, but I'm not sure where else to ask. If there's a more appropriate place, please let me know. I recently upgraded my MythTV server from Fedora 11 to Fedora 14. Now the only way to get sound to work is to follow these steps: Start the audio source so that there's an audio stream Open the Pulseaudio Volume Control panel Click the Configuration tab, Change Internal Audio to any profile other than the current one Change Internal Audio back to the original profile ( Digital Stereo Duplex (IEC958) ) At this point, the audio works. But if I change from watching one show in MythTV to another, I have to go through the whole process again. It's like as soon as pulseaudio loses a signal, it has to be reset. The issue is odd enough that I don't even know what keywords to search for to see if this is a common problem - has anyone heard of this? Does anyone have any suggestions for how to fix or troubleshoot it? rm -rf ~/.pulse* pulseaudio -vvv 2 log.txt pastebin log.txt and send us the link Also, are you sure MythTV is using PA? The MythTV guys insist on automatically suspending PA which can cause deadlocks if you configure MythTV to use the alsa:default ALSA device. I've tried to explain this on IRC before, but it has never quite stuck. I'd either configure your MythTV to use the device alsa:pulse (assuming you have such an alsa definition?) or to use the built in PulseAudio support. Or simply ask your local friendly Fedora packager to apply this patch... (which I still maintain should be applied upstream but I've given up trying...) http://svnweb.mageia.org/packages/cauldron/mythtv/current/SOURCES/0101-pulse-Do-not-suspend-PA-when-using-alsa-default.patch?view=markup HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] [RFC] Pulseaudio jack sense
Hiya Maggie, 'Twas brillig, and Margarita Olaya at 11/03/11 16:50 did gyre and gimble: The second thing is to figure out what we should really do once we have detected jack insertion or removal. Adding a hook event and let other modules act on that, seems reasonable to me. But we should also add a property to the property list (or something similar?) telling the current state of the jack - and btw, can this be figured out by reading from /dev/input/eventX at startup? atm the target is to used it along with the UCM module, so once we have detected jack insertion or removal the UCM will configure the sound card for it, so we need a (new?) hook to notify the other modules. Sorry for lack of replies from me of late... been a bit swamped at $DAYJOB :p Am I right in saying that you basically have to set appropriate verbs via UCM when the jack is inserted? In my mind, jack detection should basically change the sink/source ports only. Then the UCM module would hook into the fact that the port had changed to do it's magic (there are existing hooks for this already in place). That way jack detection is technically separated from UCM via the abstraction of sink/source ports in PA. We discussed a similar logic for setting verbs on IRC. http://colin.guthr.ie/meetings/pulseaudio-meeting/2011/pulseaudio-meeting.2011-02-24-21.02.log.html#l-146 This may not be entirely possible depending on what you actually need to do with UCM... but perhaps it will fit in nicely. I've not looked specifically at the code, but I'd have expected that jack detection would somehow be built into module-alsa-card or module-alsa-source/sink rather than shipped as a separate module (tho' I'm maybe not appreciating some intricacy here)... I'll try and review the code soon (although others may have beaten me to it anyway :D) Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio 0.9.22, after system suspends and resumes, alsasink is blocked
Running into a weird issue. I am doing the following on 0.9.22 version of pulseaudio start pulseaudio, and play and audio file. when the file is done playing, after 5 seconds, the alsa sink suspends. then I let the system go to power collapse. When I resume and play an audio file again, alsa sink is blocked somewhere, I do not see the message I: alsa-sink.c: Trying resume... The PA_SINK_SET_STATE message doesn't seem to be called to the alsa sink. Alsa is not blocked at this point, I can do an aplay to hw:0 and hear the output, bypassing pulseaudio. here is an example from pulseaudio with debug messages enabled, after suspend/resume: aplay -D media /usr/palm/sounds/phone.wav -vvv root@palm-webos:/var/home/root# I: client.c: Created 2 Native client (UNIX socket client) D: protocol-native.c: Protocol version: remote 19, local 19 I: protocol-native.c: Got credentials: uid=0 gid=0 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: no Playing WAVE '/usr/palm/sounds/phone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo D: module-suspend-on-idle.c: Sink pcm_output becomes busy. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 0 ALSA Playback on pcm_output with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: sink-input.c: media.name = ALSA Playback I: sink-input.c: application.name = ALSA plug-in [aplay] I: sink-input.c: native-protocol.peer = UNIX socket client I: sink-input.c: native-protocol.version = 19 I: sink-input.c: application.process.id = 3593 I: sink-input.c: application.process.user = root I: sink-input.c: application.process.host = palm-webos I: sink-input.c: application.process.binary = aplay I: sink-input.c: application.language = C I: sink-input.c: application.process.machine_id = palm-webos I: protocol-native.c: Requested tlength=500.00 ms, minreq=124.99 ms D: protocol-native.c: Early requests mode enabled, configuring sink latency to minreq. D: memblockq.c: memblockq requested: maxlength=4194304, tlength=88200, base=4, prebuf=66152, minreq=9596 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=88200, base=4, prebuf=66152, minreq=9596 maxrewind=0 I: protocol-native.c: Final latency 554.42 ms = 391.20 ms + 2*54.40 ms + 54.42 ms D: sink-input.c: SetVolumeWithRamping: Virtual Volume From 1646=0.16 to 41160=0.247734 D: sink-input.c: Sink input's soft volume is 41160= 0.247734 D: sink-input.c: Volume Ramping: Point 1 is 1=0.15, Point 2 is 16236=0.247742 I: module-palm-policy.c: parse_message: ramp command received, sink is 4, volumetoset:70, headphones:0 ALSA - PulseAudio PCM I/O Plugin Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat: STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min: 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 Max peak (11024 samples): 0x2dd0 35% Max peak (11024 samples): 0x332c 39% Max peak (11024 samples): 0x3741 #43% D: protocol-native.c: Requesting rewind due to end of underrun. D: protocol-native.c: Requesting rewind due to end of underrun. Max peak (11024 samples): 0x353f #41% D: protocol-native.c: Requesting rewind due to end of underrun. D: protocol-native.c: Requesting rewind due to end of underrun. Just waits here forever. Sink doesnt get resume call, and alsasink doesn't get resumed either Any ideas? -- -baeksanchang ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
