date:20110526

Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

2011-05-26 Thread xing wang

Hi baeksan,

Accidently I see your post about Pulseaudio rewind issue, that's quite
same with mine.
The background is, with timer-based feature enabled, i use default
2000ms  of tsched_size, when playback begins
there's rewind operation because of application's request latency very
low,so the final buffer size will be a small value after calculation.
then there will be nearly 3 seconds before hear the real sound. Maybe
alsa is playing silence data or didnot start to playback operations.

From your log i see the buffer_size/period_size is not too big, so do
you have try with a big tsched_size value(such as 2 second). if we
have same result, that's maybe Pulseaudio's bug about the rewind operations.

--xingchao




=
From baeksan at ccrma.stanford.edu  Sun May  1 00:38:34 2011
From: baeksan at ccrma.stanford.edu (Baek Chang)
Date: Sat, 30 Apr 2011 15:38:34 -0700
Subject: [pulseaudio-discuss] rewind and underrun issues on start of playback
Message-ID: banlktikq1u6duzzikgfqgs0ofm2mkc5...@mail.gmail.com

Hi,

I'm seeing some issue with underruns/rewinds occurring on the beginning of
every sink input playback.
I see rewind requests on alsa sink of 9600 bytes.  The alsa driver is
configured with the following buffer sizes

I: sink.c: device.buffering.buffer_size = 9600
I: sink.c: device.buffering.fragment_size = 4800

So it seems like one buffer size is always being rewinded on the beginning
of playback.
Is there a way to prevent these rewinds/underruns when starting playback?
 after the stream has started, there is no issue with audio dropouts or
underruns, just on the beginning.

If i log the data that gets sent to alsa, from pulseaudio or in the alsa
driver, i do see the beginning being dropped as well.

please see attached logs using both tshed=0 and tsched=1.

any help with this?
thanks!

I: client.c: Created 0 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
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 /usr/palm/sounds/notification.wav on
pcm_output with sample spec s16le 2ch 44100Hz and channel map
front-left,front-right
I: sink-input.c: media.format = WAV (Microsoft)
I: sink-input.c: media.name = /usr/palm/sounds/notification.wav
I: sink-input.c: application.name = pacat
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 = 4474
I: sink-input.c: application.process.user = root
I: sink-input.c: application.process.host = PalmDevice
I: sink-input.c: application.process.binary = pacat
I: sink-input.c: application.language = C
I: sink-input.c: application.process.machine_id = PalmDevice
I: protocol-native.c: Requested tlength=2000.00 ms, minreq=20.00 ms
D: protocol-native.c: Traditional mode enabled, modifying sink usec only for
compat with minreq.
D: memblockq.c: memblockq requested: maxlength=4194304, tlength=352800,
base=4, prebuf=349276, minreq=3528 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=352800,
base=4, prebuf=349276, minreq=3528 maxrewind=0
I: protocol-native.c: Final latency 2054.42 ms = 1960.00 ms + 2*20.00 ms +
54.42 ms
D: protocol-native.c: Requesting rewind due to end of underrun.
D: alsa-sink.c: Requested to rewind 9600 bytes.
D: alsa-sink.c: Limited to 9344 bytes.
D: alsa-sink.c: before: 2336
D: alsa-sink.c: after: 2336
D: alsa-sink.c: Rewound 9344 bytes.
D: sink.c: Processing rewind...
D: sink-input.c: Have to rewind 9344 bytes on render memblockq.
D: source.c: Processing rewind...
D: protocol-native.c: Underrun on '/usr/palm/sounds/notification.wav', 0
bytes in queue.
D: alsa-sink.c: Requested to rewind 9600 bytes.
D: alsa-sink.c: Limited to 9344 bytes.
D: alsa-sink.c: before: 2336
D: alsa-sink.c: after: 2336
D: alsa-sink.c: Rewound 9344 bytes.
D: sink.c: Processing rewind...
D: source.c: Processing rewind...
D: core.c: Hmm, no streams around, trying to vacuum.
I: sink-input.c: Freeing input 0 /usr/palm/sounds/notification.wav
I: client.c: Freed 0 pacat
I: protocol-native.c: Connection died.

-- 
-baeksanchang
-- next part --
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URL: 
https://tango.0pointer.de/pipermail/pulseaudio-discuss/attachments/20110430/97c1f9a4/attachment-0001.htm
-- next part --
root at PalmDevice:/media/internal# pulseaudio -
D: core-rtclock.c: Timer slack is set to 50 us.
D: core-util.c: setpriority() worked.
I: core-util.c: Successfully gained nice level -4.
I: main.c: Found user 'pulse' (UID 31) and

Re: [pulseaudio-discuss] ask about the assumption of min volume and max volume got from ALSA

2011-05-26 Thread Colin Guthrie

Hi,

'Twas brillig, and xing wang at 26/05/11 04:52 did gyre and gimble:
 Would you please give some comments about Guanqun's patch?
 We're working on Pulseaudio and meet some bugs, i think we will run
 faster with community's help.

Absolutely!

To be honest tho', I was hoping Tanu or David would give their takes on
this patch (which they may still do) as they have a better grasp of the
audio control side than I do.


One thing I will say is that PA will use the info from alsa in such a
way as to provide a consistent dB range further up the stack.

We take the alsa-provided 0dB point and mark that as our Base volume
this is exposed e.g. in the pavucontrol and gnome-volume-control GUI's

This base volume is meant to indicate the cleanest path through the h/w.

This might not be super relevant to the specific issue in question, but
that's a little background on the dB levels exposed by alsa, but I get
the impression the volumes you're talking about are not dB related?


Looking at your patch, it seems logical enough (although for internal
code, I'd use pa_bool_t + TRUE/FALSE values, but that's just a minor
observation), so if Tanu ACKs it, I'd happily merge it.

Cheers

Col


 thanks
 --xingchao
 
 2011/5/25 Lu Guanqun guanqun...@intel.com:
 Hi,

 This patch needs a little more love. :)
 No one cared?

 On Wed, May 25, 2011 at 02:17:18PM +0800, Lu Guanqun wrote:
 Hi list,

 I may not have the background on this issue, but from the code, it seems
 the code assumes min_volume and max_volume should be non negative. Is
 this the right assumption we should take? What about some drivers
 exposes that min_volume is -126 and max_volume is 0? Should we fix the
 driver or generalize pulseaudio code to tolerate these issues?

 If we allow min_volume could be less than zero, and how about the below
 patch in your point of view? Could you have a review of the below patch?
 If that's OK, I can send a standalone patch for it.

 From 1f5c8fa0e80d9fe2e3d56133afa8fe9a5dbe6693 Mon Sep 17 00:00:00 2001
 From: Lu Guanqun guanqun...@intel.com
 Date: Wed, 25 May 2011 14:12:52 +0800
 Subject: [PATCH] fix the assumption that volume is always positive

 Add a variable to track whether the actual volume is set or not.
 Suppose this:
   min volume: -126max volume: 0
 then when user wants to set some constant volume to -10, it would fail.

 Signed-off-by: Lu Guanqun guanqun...@intel.com
 ---
  src/modules/alsa/alsa-mixer.c |6 +-
  1 files changed, 5 insertions(+), 1 deletions(-)

 diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c
 index 1bd709a..23651b0 100644
 --- a/src/modules/alsa/alsa-mixer.c
 +++ b/src/modules/alsa/alsa-mixer.c
 @@ -1123,6 +1123,7 @@ static int 
 element_set_constant_volume(pa_alsa_element *e, snd_mixer_t *m) {
  snd_mixer_selem_id_t *sid = NULL;
  int r = 0;
  long volume = -1;
 +int volume_set = 0;

  pa_assert(m);
  pa_assert(e);
 @@ -1136,6 +1137,7 @@ static int 
 element_set_constant_volume(pa_alsa_element *e, snd_mixer_t *m) {
  switch (e-volume_use) {
  case PA_ALSA_VOLUME_OFF:
  volume = e-min_volume;
 +volume_set = 1;
  break;

  case PA_ALSA_VOLUME_ZERO:
 @@ -1143,18 +1145,20 @@ static int 
 element_set_constant_volume(pa_alsa_element *e, snd_mixer_t *m) {
  long dB = 0;

  volume = decibel_fix_get_step(e-db_fix, dB, +1);
 +volume_set = 1;
  }
  break;

  case PA_ALSA_VOLUME_CONSTANT:
  volume = e-constant_volume;
 +volume_set = 1;
  break;

  default:
  pa_assert_not_reached();
  }

 -if (volume = 0) {
 +if (volume_set) {
  if (e-direction == PA_ALSA_DIRECTION_OUTPUT)
  r = snd_mixer_selem_set_playback_volume_all(me, volume);
  else
 --
 1.7.2.3

 --
 guanqun
 ___
 pulseaudio-discuss mailing list
 pulseaudio-discuss@mail.0pointer.de
 https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss

 --
 guanqun
 ___
 pulseaudio-discuss mailing list
 pulseaudio-discuss@mail.0pointer.de
 https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss



-- 

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/]
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Re: [pulseaudio-discuss] ask about the assumption of min volume and max volume got from ALSA

2011-05-26 Thread Lu Guanqun

Hi Col,

On Thu, May 26, 2011 at 04:30:03PM +0800, Colin Guthrie wrote:
 Hi,
 
 'Twas brillig, and xing wang at 26/05/11 04:52 did gyre and gimble:
  Would you please give some comments about Guanqun's patch?
  We're working on Pulseaudio and meet some bugs, i think we will run
  faster with community's help.
 
 Absolutely!
 
 To be honest tho', I was hoping Tanu or David would give their takes on
 this patch (which they may still do) as they have a better grasp of the
 audio control side than I do.
 
 
 One thing I will say is that PA will use the info from alsa in such a
 way as to provide a consistent dB range further up the stack.
 
 We take the alsa-provided 0dB point and mark that as our Base volume
 this is exposed e.g. in the pavucontrol and gnome-volume-control GUI's
 
 This base volume is meant to indicate the cleanest path through the h/w.
 
 This might not be super relevant to the specific issue in question, but
 that's a little background on the dB levels exposed by alsa, but I get
 the impression the volumes you're talking about are not dB related?

Yes, you're right. The volume here is not related to dB. It's just plain
volume number I got from ALSA via this API 
(snd_mixer_selem_get_playback_volume_range).
In some ALSA drivers, I got the min_vol to be less than zero. So when I
try to set it to be a constant volume, it fails to do so.

 
 
 Looking at your patch, it seems logical enough (although for internal
 code, I'd use pa_bool_t + TRUE/FALSE values, but that's just a minor
 observation), so if Tanu ACKs it, I'd happily merge it.

I'll change it to bool and send another separate patch for it.

 
 Cheers
 
 Col
 
 
  thanks
  --xingchao
  
  2011/5/25 Lu Guanqun guanqun...@intel.com:
  Hi,
 
  This patch needs a little more love. :)
  No one cared?
 
  On Wed, May 25, 2011 at 02:17:18PM +0800, Lu Guanqun wrote:
  Hi list,
 
  I may not have the background on this issue, but from the code, it seems
  the code assumes min_volume and max_volume should be non negative. Is
  this the right assumption we should take? What about some drivers
  exposes that min_volume is -126 and max_volume is 0? Should we fix the
  driver or generalize pulseaudio code to tolerate these issues?
 
  If we allow min_volume could be less than zero, and how about the below
  patch in your point of view? Could you have a review of the below patch?
  If that's OK, I can send a standalone patch for it.
 
  From 1f5c8fa0e80d9fe2e3d56133afa8fe9a5dbe6693 Mon Sep 17 00:00:00 2001
  From: Lu Guanqun guanqun...@intel.com
  Date: Wed, 25 May 2011 14:12:52 +0800
  Subject: [PATCH] fix the assumption that volume is always positive
 
  Add a variable to track whether the actual volume is set or not.
  Suppose this:
min volume: -126max volume: 0
  then when user wants to set some constant volume to -10, it would fail.
 
  Signed-off-by: Lu Guanqun guanqun...@intel.com
  ---
   src/modules/alsa/alsa-mixer.c |6 +-
   1 files changed, 5 insertions(+), 1 deletions(-)
 
  diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c
  index 1bd709a..23651b0 100644
  --- a/src/modules/alsa/alsa-mixer.c
  +++ b/src/modules/alsa/alsa-mixer.c
  @@ -1123,6 +1123,7 @@ static int 
  element_set_constant_volume(pa_alsa_element *e, snd_mixer_t *m) {
   snd_mixer_selem_id_t *sid = NULL;
   int r = 0;
   long volume = -1;
  +int volume_set = 0;
 
   pa_assert(m);
   pa_assert(e);
  @@ -1136,6 +1137,7 @@ static int 
  element_set_constant_volume(pa_alsa_element *e, snd_mixer_t *m) {
   switch (e-volume_use) {
   case PA_ALSA_VOLUME_OFF:
   volume = e-min_volume;
  +volume_set = 1;
   break;
 
   case PA_ALSA_VOLUME_ZERO:
  @@ -1143,18 +1145,20 @@ static int 
  element_set_constant_volume(pa_alsa_element *e, snd_mixer_t *m) {
   long dB = 0;
 
   volume = decibel_fix_get_step(e-db_fix, dB, +1);
  +volume_set = 1;
   }
   break;
 
   case PA_ALSA_VOLUME_CONSTANT:
   volume = e-constant_volume;
  +volume_set = 1;
   break;
 
   default:
   pa_assert_not_reached();
   }
 
  -if (volume = 0) {
  +if (volume_set) {
   if (e-direction == PA_ALSA_DIRECTION_OUTPUT)
   r = snd_mixer_selem_set_playback_volume_all(me, volume);
   else
  --
  1.7.2.3
 
  --
  guanqun
  ___
  pulseaudio-discuss mailing list
  pulseaudio-discuss@mail.0pointer.de
  https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
 
  --
  guanqun
  ___
  pulseaudio-discuss mailing list
  pulseaudio-discuss@mail.0pointer.de
  https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
 
 
 
 -- 
 
 Colin Guthrie
 gmane(at)colin.guthr.ie
 http://colin.guthr.ie/
 
 Day Job:
   Tribalogic Limited

[pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

2011-05-26 Thread Lu Guanqun

Add a variable to track whether the actual volume is set or not.
Suppose this:
min volume: -126max volume: 0
then when user wants to set some constant volume to -10, it would fail.
---
 src/modules/alsa/alsa-mixer.c |6 +-
 1 files changed, 5 insertions(+), 1 deletions(-)

diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c
index f236da0..03a5312 100644
--- a/src/modules/alsa/alsa-mixer.c
+++ b/src/modules/alsa/alsa-mixer.c
@@ -1092,6 +1092,7 @@ static int element_set_constant_volume(pa_alsa_element 
*e, snd_mixer_t *m) {
 snd_mixer_selem_id_t *sid = NULL;
 int r = 0;
 long volume = -1;
+pa_bool_t volume_set = FALSE;
 
 pa_assert(m);
 pa_assert(e);
@@ -1105,6 +1106,7 @@ static int element_set_constant_volume(pa_alsa_element 
*e, snd_mixer_t *m) {
 switch (e-volume_use) {
 case PA_ALSA_VOLUME_OFF:
 volume = e-min_volume;
+volume_set = TRUE;
 break;
 
 case PA_ALSA_VOLUME_ZERO:
@@ -1112,18 +1114,20 @@ static int element_set_constant_volume(pa_alsa_element 
*e, snd_mixer_t *m) {
 long dB = 0;
 
 volume = decibel_fix_get_step(e-db_fix, dB, +1);
+volume_set = TRUE;
 }
 break;
 
 case PA_ALSA_VOLUME_CONSTANT:
 volume = e-constant_volume;
+volume_set = TRUE;
 break;
 
 default:
 pa_assert_not_reached();
 }
 
-if (volume = 0) {
+if (volume_set) {
 if (e-direction == PA_ALSA_DIRECTION_OUTPUT)
 r = snd_mixer_selem_set_playback_volume_all(me, volume);
 else

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Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

2011-05-26 Thread Colin Guthrie

'Twas brillig, and Lu Guanqun at 26/05/11 09:49 did gyre and gimble:
 Add a variable to track whether the actual volume is set or not.
 Suppose this:
   min volume: -126max volume: 0
 then when user wants to set some constant volume to -10, it would fail.

Tanu, are you OK with this patch?

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/]

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Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

2011-05-26 Thread Colin Guthrie

'Twas brillig, and xing wang at 26/05/11 09:09 did gyre and gimble:
 I: main.c: This is PulseAudio 0.9.22

Just as a very small aside, David did some work on Fighting Rewinds
recently.

Just search the stable-queue git log for Fighting rewinds...

These may already be included in your build, but if not it's perhaps
worth grabbing those patches?

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/]

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Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

2011-05-26 Thread xing wang

2011/5/26 Colin Guthrie gm...@colin.guthr.ie:
 'Twas brillig, and xing wang at 26/05/11 09:09 did gyre and gimble:
 I: main.c: This is PulseAudio 0.9.22

 Just as a very small aside, David did some work on Fighting Rewinds
 recently.

 Just search the stable-queue git log for Fighting rewinds...

 These may already be included in your build, but if not it's perhaps
 worth grabbing those patches?

Thank you Colin, let me have a try.

--xingchao

 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

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Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

2011-05-26 Thread Colin Guthrie

'Twas brillig, and xing wang at 26/05/11 10:42 did gyre and gimble:
 2011/5/26 Colin Guthrie gm...@colin.guthr.ie:
 'Twas brillig, and xing wang at 26/05/11 09:09 did gyre and gimble:
 I: main.c: This is PulseAudio 0.9.22

 Just as a very small aside, David did some work on Fighting Rewinds
 recently.

 Just search the stable-queue git log for Fighting rewinds...

 These may already be included in your build, but if not it's perhaps
 worth grabbing those patches?
 
 Thank you Colin, let me have a try.

Of course it's probably sensible to just run the whole stable-queue
branch. That's generally what I do in my distro packages... just apply
all stable queue patches on top of the official release. A 0.9.23
release will likely go out soon (if I can nail Lennart down!) based on
this branch.

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/]

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Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

2011-05-26 Thread xing wang

2011/5/26 Colin Guthrie gm...@colin.guthr.ie:
'Twas brillig, and xing wang at 26/05/11 10:42 did gyre and gimble:
2011/5/26 Colin Guthrie gm...@colin.guthr.ie:
'Twas brillig, and xing wang at 26/05/11 09:09 did gyre and gimble:
I: main.c: This is PulseAudio 0.9.22

Just as a very small aside, David did some work on Fighting Rewinds
recently.

Just search the stable-queue git log for Fighting rewinds...

These may already be included in your build, but if not it's perhaps
worth grabbing those patches?

Thank you Colin, let me have a try.

Of course it's probably sensible to just run the whole stable-queue
branch. That's generally what I do in my distro packages... just apply
all stable queue patches on top of the official release. A 0.9.23
release will likely go out soon (if I can nail Lennart down!) based on
this branch.

Hi Colin, i think my release had include David's patchset. I see
there're three patches about Fast rewind, here's the link:
http://gstreamer-devel.966125.n4.nabble.com/Fighting-rewinds-pulseaudio-crash-update-td3248107.html.

The delay issue met based on my release build could not be fixed with
David's patch.
But i think it's quite like Ccrma's description in post rewind and
underrun issues on start of playback. If change the buffer size to
smaller value, the delay becomes short. I am not sure whether alsa is
playing silence data or just didnot start playback.

Thanks
--xingchao

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/]

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Re: [pulseaudio-discuss] [PATCH 1/2] alsa: add jack detection support

2011-05-26 Thread David Henningsson


I was asked by Colin to give this a quick review, so...

On 2011-05-19 18:44, Jorge Eduardo Candelaria wrote:

From: Margarita Olaya Cabreram...@slimlogic.co.uk

This patch adds support to module-alsa-card so that we can read jack insertion
and removal events using the input device subsystem. i.e. we can detect
headphone insertions and removals.

This patch adds a new module parameter called jack_id that contains the ID of
the jack input device associated with this sound card. If we receive a valid
jack_id during module init then we start a reader thread that will read the
jack input device and fire a hook on every removal and insertion event.

Jack support development was kindly sponsored by Wolfson Microelectronics PLC

Signed-off-by: Margarita Olaya Cabreram...@slimlogic.co.uk
Signed-off-by: Jorge Eduardo Candelariaj...@slimlogic.co.uk
---
src/modules/alsa/alsa-jack.h|   42 
src/modules/alsa/module-alsa-card.c |  120 +++
src/pulsecore/core.h|2 +
3 files changed, 164 insertions(+), 0 deletions(-)
create mode 100644 src/modules/alsa/alsa-jack.h

diff --git a/src/modules/alsa/alsa-jack.h b/src/modules/alsa/alsa-jack.h
new file mode 100644
index 000..4fc67c6
--- /dev/null
+++ b/src/modules/alsa/alsa-jack.h


Along the lines of Tanu, if this is not alsa-specific, let's take this 
header and move into pulsecore.



@@ -0,0 +1,42 @@
+#ifndef foopulsejackdetecthfoo
+#define foopulsejackdetecthfoo
+
+/***
+  This file is part of PulseAudio.
+
+  Copyright 2011 Wolfson Microelectronics PLC
+  Author Margarita Olayam...@slimlogic.co.uk
+
+  PulseAudio is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser General Public License as published
+  by the Free Software Foundation; either version 2.1 of the License,
+  or (at your option) any later version.
+
+  PulseAudio is distributed in the hope that it will be useful, but
+  WITHOUT ANY WARRANTY; without even the implied warranty of
+  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+  General Public License for more details.
+
+  You should have received a copy of the GNU Lesser General Public License
+  along with PulseAudio; if not, write to the Free Software
+  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+  USA.
+***/
+
+#includeinttypes.h
+
+typedef enum pa_jack_event {
+PA_JACK_HEADPHONES,
+PA_JACK_HEADSET,


I'm a little hesitant to whether we should have one headset or whether 
that should be emulated by one headphone and one microphone, as that 
is what a headset is.



+PA_JACK_MICROPHONE,
+PA_JACK_LINEOUT,


add PA_JACK_VIDEOOUT for HDMI/Displayport (which was just added to ALSA).


+PA_JACK_UNKNOWN,
+PA_JACK_MAX
+} pa_jack_event_t;
+
+typedef struct pa_jack_detect {
+pa_jack_event_t event;
+char *card;


Agree with Tanu about pa_card pointer here.


+} pa_jack_detect_t;
+
+#endif
diff --git a/src/modules/alsa/module-alsa-card.c 
b/src/modules/alsa/module-alsa-card.c
index e60aa5e..75202fb 100644
--- a/src/modules/alsa/module-alsa-card.c
+++ b/src/modules/alsa/module-alsa-card.c
@@ -29,6 +29,9 @@
#includepulsecore/core-util.h
#includepulsecore/modargs.h
#includepulsecore/queue.h
+#includepulsecore/thread.h
+
+#includelinux/input.h

#includemodules/reserve-wrap.h

@@ -39,6 +42,7 @@
#include alsa-util.h
#include alsa-sink.h
#include alsa-source.h
+#include alsa-jack.h
#include module-alsa-card-symdef.h

PA_MODULE_AUTHOR(Lennart Poettering);
@@ -55,6 +59,7 @@ PA_MODULE_USAGE(
 source_properties=properties for the source  
 namereg_fail=pa_namereg_register() fail parameter value  
 device_id=ALSA card index  
+jack_id=Jack device index  


Eh, a card can have more than one jack, and is further rather connected 
to a port rather than a card, so this (and thus the rest of the 
implementation) doesn't seem right to me.


--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
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Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

2011-05-26 Thread Tanu Kaskinen

On Thu, 2011-05-26 at 09:50 +0100, Colin Guthrie wrote:
 'Twas brillig, and Lu Guanqun at 26/05/11 09:49 did gyre and gimble:
  Add a variable to track whether the actual volume is set or not.
  Suppose this:
  min volume: -126max volume: 0
  then when user wants to set some constant volume to -10, it would fail.
 
 Tanu, are you OK with this patch?

Maybe, maybe not. I posted a question to alsa-devel:
http://mailman.alsa-project.org/pipermail/alsa-devel/2011-May/040213.html

If negative steps are allowed, I expect there to be more bugs.

-- 
Tanu

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Re: [pulseaudio-discuss] How to change profile of bluetooth headset for different applications?

2011-05-26 Thread Tanu Kaskinen

On Mon, 2011-05-23 at 18:18 +0100, Colin Guthrie wrote:
 How about this:
 
 Define a new card property similar in concept to
 PA_PROP_DEVICE_INTENDED_ROLES (device.intended_roles) called
 PA_PROP_DEVICE_INTENDED_PROFILE (device.intended_profile).
 
 In bluetooth device, when loading a headset that has both HSP and A2DP,
 populate the device with the following code:
 
 if (supports_hsp  supports_a2dp)
 {
   char *k;
 
   k = pa_sprintf_malloc(%s.%s, PA_PROP_DEVICE_INTENDED_PROFILE, music);
   pa_proplist_sets(card_data.proplist, k, a2dp);
   pa_xfree(k);
 
   k = pa_sprintf_malloc(%s.%s, PA_PROP_DEVICE_INTENDED_PROFILE, video);
   pa_proplist_sets(card_data.proplist, k, a2dp);
   pa_xfree(k);
 
   k = pa_sprintf_malloc(%s.%s, PA_PROP_DEVICE_INTENDED_PROFILE, phone);
   pa_proplist_sets(card_data.proplist, k, hsp);
   pa_xfree(k);
 }
 
 This should result in pacmd list-cards dumping a proplist that includes:
 
  device.intended_profile.music = a2dp
  device.intended_profile.video = a2dp
  device.intended_profile.phone = hsp
 
 
 
 Then a separate module (module-intended-profiles?) could take care of
 listening quite generically for streams landing on sink and do the
 following logic:
 
 
 if (!si-sink || !si-sink-card)
   return PA_HOOK_OK;
 
 char *role = pa_proplist_gets(si-proplist, PA_PROP_MEDIA_ROLE);
 if (!role)
   return PA_HOOK_OK;
 
 char *k = pa_sprintf_malloc(%s.%s, PA_PROP_DEVICE_INTENDED_PROFILE, role);
 pa_card *card = si-sink-card;
 char *profile = pa_proplist_gets(card-proplist, k);
 pa_xfree(k);
 if (!profile)
   return PA_HOOK_OK;
 
 if (!pa_streq(card-active_profile, profile)) {
/* Save the current profile for later restoration */
...
 
if (pa_card_set_profile(card, profile, FALSE)  0)
/* Find the sink of this card as the above call will likely have
 changed it */
sink = pa_idxset_first(card-sinks, NULL);
 
pa_sink_input_move_to(si, sink, FALSE);
 }
 
 
 This code is then completely generic. I'm not 100% sure how well this
 will work in practice as there may be some nasty races etc. but I think
 this general approach could work... Perhaps others (i.e. Tanu :D) have
 better suggestions?

If the priority lists would exist already, the module deciding the
routing could examine the stream properties and pick the sink+port with
the highest priority that can be made available - if the sink+port isn't
immediately available, but can be made available, then the routing
module would change the profile and port as needed.

But the priority lists are not available. I believe the logic that you
propose is otherwise good, except that I don't see how the stream could
land on the bluetooth (a2dp) sink automatically on phone calls.
module-bluetooth-device tags only hsp sinks with the phone tag, not a2dp
sinks, so module-intended-roles can't do the required magic. If the hsp
sink doesn't exist because the a2dp profile is selected, I don't think
there's currently any logic to route the stream to the bluetooth sink.
If, however, by some magic the phone stream would be routed to the a2dp
sink, then I think your proposal would work fine.

For better suggestions, I don't think I have any. Except that use the
policy framework - Lin posted from an intel.com address, so I assume he
or she (sorry, I don't know Chinese names) is working on Meego, and I've
thought that Meego got the policy framework from Maemo. In Maemo the
routing is handled by module-policy-enforcement (which gets the command
to enable the bluetooth-hsp routing mode from the policy framework), and
it works pretty nicely for this kind of use cases.

-- 
Tanu

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Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

2011-05-26 Thread Tanu Kaskinen

On Thu, 2011-05-26 at 14:08 +0300, Tanu Kaskinen wrote:
 On Thu, 2011-05-26 at 09:50 +0100, Colin Guthrie wrote:
  'Twas brillig, and Lu Guanqun at 26/05/11 09:49 did gyre and gimble:
   Add a variable to track whether the actual volume is set or not.
   Suppose this:
 min volume: -126max volume: 0
   then when user wants to set some constant volume to -10, it would fail.
  
  Tanu, are you OK with this patch?
 
 Maybe, maybe not. I posted a question to alsa-devel:
 http://mailman.alsa-project.org/pipermail/alsa-devel/2011-May/040213.html
 
 If negative steps are allowed, I expect there to be more bugs.

I got a confirmation that negative volumes are allowed, so the patch
should be fine. Somebody should review the alsa mixer code for other
cases of the invalid assumption (I plan to do it in the indefinite
future, if nobody else does it).

-- 
Tanu

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Re: [pulseaudio-discuss] [PACKAGERS] New dep

2011-05-26 Thread Maarten Bosmans

2011/5/16 Colin Guthrie gm...@colin.guthr.ie:
 Hi,

 I've just pushed Arun's (mostly, Pierre-Louis also had a hand!)
 passthrough work to master.

 This carries with it a new dependency: json-c

What is the upstream location of that software? Googling for json
implementations gives a lot of small libraries, it is not clear which
one is needed as the pulse dep.

I guess it could be the one at http://oss.metaparadigm.com/json-c/,
but that website is out of order for some time now. The .tar file is
only locatable through the packaging systems of various Linux
distros/BSDs.

Needless to say, such an unclear status of a dependency is not really
helpfull in packaging PulseAudio.

 We may yet remove this and write our own parser for the simple subset of
 JSON formatting we use but there may ultimately be other reasons to keep
 this longer term.

 But in the interests of simplicity, I'd certainly not be against any
 native implementation patches that came along provided they were simple
 and clean.

 Please test these changes. I'm sure Arun will post more details and test
 requests in due course.

 Col

Maarten
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Re: [pulseaudio-discuss] [PACKAGERS] New dep

2011-05-26 Thread Colin Guthrie

'Twas brillig, and Maarten Bosmans at 26/05/11 15:23 did gyre and gimble:
 2011/5/16 Colin Guthrie gm...@colin.guthr.ie:
 Hi,

 I've just pushed Arun's (mostly, Pierre-Louis also had a hand!)
 passthrough work to master.

 This carries with it a new dependency: json-c
 
 What is the upstream location of that software? Googling for json
 implementations gives a lot of small libraries, it is not clear which
 one is needed as the pulse dep.
 
 I guess it could be the one at http://oss.metaparadigm.com/json-c/,
 but that website is out of order for some time now. The .tar file is
 only locatable through the packaging systems of various Linux
 distros/BSDs.
 
 Needless to say, such an unclear status of a dependency is not really
 helpfull in packaging PulseAudio.

I dunno, we already had a package in Mandriva and I just updated it to
the latest version from the Fedora package.

Arun, do you have more definitive info?


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/]

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[pulseaudio-discuss] module-echo-cancel

2011-05-26 Thread Baek Chang

I'm trying to use the echo cancellation module on my embedded platform but
it seems to be taking all the cpu usage and renders audio useless.

I'm continually seeing these messages:
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
D: module-echo-cancel.c: diff -269 (43555 - 88163 + 44155) 0 0 9600 184
E: module-echo-cancel.c: Playback after capture (-269), drop sink 84
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
D: module-echo-cancel.c: diff 475 (42663 - 87687 + 45315) 0 0 9600 184
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
D: module-echo-cancel.c: diff 168 (39254 - 87687 + 48410) 0 0 9600 191
D: module-echo-cancel.c: diff -411 (38089 - 87687 + 49002) 0 0 9600 185
E: module-echo-cancel.c: Playback after capture (-411), drop sink 112
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
D: module-echo-cancel.c: diff 1491 (41407 - 87052 + 46961) 0 0 9600 175
E: module-echo-cancel.c: playback too far ahead (1491), drop source 260
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
warning: The echo canceller started acting funny and got slapped (reset). It
swears it will behave now.
D: module-echo-cancel.c: diff -1166 (44921 - 88526 + 42252) 0 0 9600 187
E: module-echo-cancel.c: Playback after capture (-1166), drop sink 244

cpu is at 90-100% at the highest cpu scaling.

I was loading the module like so in my .pa conf file:

load-module module-echo-cancel source_name=echosource
source_master=pcm_input sink_name=echosink sink_master=pcm_output rate=44100
channels=2

Is there a step I am missing, or a link to documentation on how to set up
the module properly?

Thanks
-- 
-baeksanchang
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Re: [pulseaudio-discuss] module-echo-cancel

2011-05-26 Thread Arun Raghavan

On Thu, 2011-05-26 at 10:07 -0700, Baek Chang wrote:
 I'm trying to use the echo cancellation module on my embedded platform
 but it seems to be taking all the cpu usage and renders audio useless.
 
 
 I'm continually seeing these messages:
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.

This particular message is surprising. Was there a lot of feedback when
you saw this? It's also possible that your clock is drifting a lot and
is thus causing the echo canceller to give up. What hardware are you
trying this on?

 D: module-echo-cancel.c: diff -269 (43555 - 88163 + 44155) 0 0 9600
 184
 E: module-echo-cancel.c: Playback after capture (-269), drop sink 84
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.
 D: module-echo-cancel.c: diff 475 (42663 - 87687 + 45315) 0 0 9600 184
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.
 D: module-echo-cancel.c: diff 168 (39254 - 87687 + 48410) 0 0 9600 191
 D: module-echo-cancel.c: diff -411 (38089 - 87687 + 49002) 0 0 9600
 185
 E: module-echo-cancel.c: Playback after capture (-411), drop sink 112
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.
 D: module-echo-cancel.c: diff 1491 (41407 - 87052 + 46961) 0 0 9600
 175
 E: module-echo-cancel.c: playback too far ahead (1491), drop source
 260
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.
 warning: The echo canceller started acting funny and got slapped
 (reset). It swears it will behave now.
 D: module-echo-cancel.c: diff -1166 (44921 - 88526 + 42252) 0 0 9600
 187
 E: module-echo-cancel.c: Playback after capture (-1166), drop sink 244
 
 
 cpu is at 90-100% at the highest cpu scaling.
 
 
 I was loading the module like so in my .pa conf file:
 
 
 load-module module-echo-cancel source_name=echosource
 source_master=pcm_input sink_name=echosink sink_master=pcm_output
 rate=44100 channels=2

Do you really need to work with 44.1 kHz stereo streams? Your voip stack
is probably working with 8 or 16 kHz audio and running in single channel
mode would probably not make a difference either. It will, however, make
a huge difference with respect to your CPU consumption.

I pushed some more preprocessing code to module-echo-cancel yesterday,
but there's a bit of a bug which I should be pushing a fix for by
tomorrow. Once that's done, you can try it out by passing agc=1 /
denoise=1 / echo_suppress=1. We're still looking at various potential
improvements to get AEC working better, and I'll try to post a status in
the coming weeks as things start to improve.

You may also want to play with aec_method=adrian, which is an alternate
canceller. It seems to learn faster than speex, but leaves a larger echo
residue compared to once speex has had sufficient time to learn.

Cheers,
Arun


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Re: [pulseaudio-discuss] [PACKAGERS] New dep

2011-05-26 Thread Arun Raghavan

On Thu, 2011-05-26 at 16:23 +0200, Maarten Bosmans wrote:
 2011/5/16 Colin Guthrie gm...@colin.guthr.ie:
  Hi,
 
  I've just pushed Arun's (mostly, Pierre-Louis also had a hand!)
  passthrough work to master.
 
  This carries with it a new dependency: json-c
 
 What is the upstream location of that software? Googling for json
 implementations gives a lot of small libraries, it is not clear which
 one is needed as the pulse dep.

Yes, it's unfortunate that it's a buzz word and the library name doesn't
generate unique search results.

 I guess it could be the one at http://oss.metaparadigm.com/json-c/,
 but that website is out of order for some time now. The .tar file is
 only locatable through the packaging systems of various Linux
 distros/BSDs.

The website worked fine as of a month ago. I don't know why it's down
now. The library serves the purpose we need well (light weight, doesn't
invent its own type system, allows you to parse out values individually
instead of mandating key-value pairs), and is available on every major
distribution ...

 Needless to say, such an unclear status of a dependency is not really
 helpfull in packaging PulseAudio.

... so the dependency was not picked arbitrarily and the unclear status
you speak of is a recent development. I'll mail the developers and ask
them what's happening.

-- Arun

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Re: [pulseaudio-discuss] [PACKAGERS] New dep

2011-05-26 Thread Arun Raghavan

On Fri, 2011-05-27 at 00:22 +0530, Arun Raghavan wrote:
[...]
 you speak of is a recent development. I'll mail the developers and ask
 them what's happening.

Should be up in a day or so.

-- Arun

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Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

2011-05-26 Thread Lu Guanqun

On Thu, May 26, 2011 at 09:21:45PM +0800, Colin Guthrie wrote:
 'Twas brillig, and Tanu Kaskinen at 26/05/11 13:48 did gyre and gimble:
  On Thu, 2011-05-26 at 14:08 +0300, Tanu Kaskinen wrote:
  On Thu, 2011-05-26 at 09:50 +0100, Colin Guthrie wrote:
  'Twas brillig, and Lu Guanqun at 26/05/11 09:49 did gyre and gimble:
  Add a variable to track whether the actual volume is set or not.
  Suppose this:
   min volume: -126max volume: 0
  then when user wants to set some constant volume to -10, it would fail.
 
  Tanu, are you OK with this patch?
 
  Maybe, maybe not. I posted a question to alsa-devel:
  http://mailman.alsa-project.org/pipermail/alsa-devel/2011-May/040213.html
 
  If negative steps are allowed, I expect there to be more bugs.
  
  I got a confirmation that negative volumes are allowed, so the patch
  should be fine. Somebody should review the alsa mixer code for other
  cases of the invalid assumption (I plan to do it in the indefinite
  future, if nobody else does it).
 
 OK, sounds good. I will queue it up at my end.

Thanks!

 
 Judging by Takashi's comments on alsa-devel, in an ideal world the
 driver would be updated too to avoid the use of the negative steps
 (which is funky :p), but we still need this fix all the same.

I'll push a patch into the mis-behaved alsa driver to fix the negative
volume.

Thanks Tanu's info as well.

 
 Thanks for the patch.
 
 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/]
 
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-- 
guanqun
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Re: [pulseaudio-discuss] [PACKAGERS] New dep

2011-05-26 Thread Luke Yelavich

On Fri, May 27, 2011 at 04:52:02AM EST, Arun Raghavan wrote:
 The website worked fine as of a month ago. I don't know why it's down
 now. The library serves the purpose we need well (light weight, doesn't
 invent its own type system, allows you to parse out values individually
 instead of mandating key-value pairs), and is available on every major
 distribution ...

It is availab ein Ubuntu yes, however its in our universe component of the 
package archive, universe being unsupported by Canonical. If PulseAudio master 
was used in future versions of Ubuntu, I would feel comfortable knowing that 
the json c library being used will reguaruly receive upstrea maintenance, as 
that in turn makes it easier for Canonical to commit to supporting the package 
in long term releases.

Luke
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Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

Re: [pulseaudio-discuss] ask about the assumption of min volume and max volume got from ALSA

Re: [pulseaudio-discuss] ask about the assumption of min volume and max volume got from ALSA

[pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

Re: [pulseaudio-discuss] rewind and underrun issues on start of playback

Re: [pulseaudio-discuss] [PATCH 1/2] alsa: add jack detection support

Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

Re: [pulseaudio-discuss] How to change profile of bluetooth headset for different applications?

Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

Re: [pulseaudio-discuss] [PACKAGERS] New dep

Re: [pulseaudio-discuss] [PACKAGERS] New dep

[pulseaudio-discuss] module-echo-cancel

Re: [pulseaudio-discuss] module-echo-cancel

Re: [pulseaudio-discuss] [PACKAGERS] New dep

Re: [pulseaudio-discuss] [PACKAGERS] New dep

Re: [pulseaudio-discuss] [PATCH] fix the assumption that volume is always positive

Re: [pulseaudio-discuss] [PACKAGERS] New dep

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