[Bug 411688]

[Tanu Kaskinen]

Sending a wav file means a TCP connection between two computers. TCP
connections are the most common type of traffic, and the hardware is
optimized accordingly. UDP multicast is different, and apparently
network hardware is not necessarily as efficient with processing that
protocol. Comment #9 seems to suggest that the wifi specification itself
forces multicast to be inefficient, so it's not necessarily just the
hardware's fault.

Starting from PA version 5.0, module-rtp-send should not generate
traffic when nothing is playing (it can still be configured to play
silence, but paprefs doesn't configure the module that way). If you
observe different behaviour, please file a new bug and attach the output
of "pactl list".

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Title:
  pulseaudio floods network with multicast packets

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[Bug 411688]

[Tanu Kaskinen]

(In reply to freedesktop from comment #17)
> Lowering the rate is possibly not a good idea as it results in worse
> quality. The format is by default "s16be" about which I have no idea what a
> different format would help. I do not know if the data can somehow be
> compress further, because this would definitely be an idea, but I dont know
> the procols internal.

The data can certainly be compressed. I don't know if you've ever
compared the sizes of mp3 and wav files. Typical compressed audio, when
using a lossy codec, requires perhaps one tenth of the bandwidth that
uncompressed audio requires, without too much effect on the audio
quality.

> So the only left option is to solve the routing. For my personal setup there
> is no way to fix this on the router side, so the sender pulse module needs
> to take care of it. Using an ip as destination works perfectly and would be
> an option to use (which should be added to paprefs). However I tried that
> the last days and due to dhcp my receiver always gets a new IP and using
> hostnames would be a better idea.
> 
> But the module does not seem to support hostnames. So I am wondering if this
> could be fixed quite simple or needs a whole more work to do.

Changing module-rtp-send to support hostnames doesn't sound terribly
complicated.

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Title:
  pulseaudio floods network with multicast packets

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[Bug 411688]

[Tanu Kaskinen]

To my knowledge RTP can carry pretty much any format. You said on the
mailing list that you want to use Kodi to receive the stream, and I
don't know what Kodi supports, but I would guess that it supports many
formats.

Don't get too excited about compressed audio, though. I'm not
volunteering to implement it, and it's probably pretty complicated to
integrate to PulseAudio.

In case someone is willing to take this task, the current "plan" (if you
can call it a plan - it's all very fuzzy at the moment) is to use
gstreamer to deal with the codecs, but in any case, please bring this
topic up on the mailing list before spending too much time implementing
it, so that there's an agreement about the design.

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  pulseaudio floods network with multicast packets

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[Bug 411688]

[Tanu Kaskinen]

*** Bug 98649 has been marked as a duplicate of this bug. ***

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Title:
  pulseaudio floods network with multicast packets

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[Bug 411688]

[Tanu Kaskinen]

I read through the comments, and to me it looks like module-rtp-send
doesn't have any flooding bug. The data rate of PCM audio is what it is,
and the network hardware isn't always able to handle it. Multicast RTP
is not enabled by default. If the user enables it, I don't know what we
could possibly do to not "flood" the network.

Since this is a common problem for people, there is certainly room for
improvement in the UI and documentation so that users don't get so
surprised when things don't work.

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  pulseaudio floods network with multicast packets

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[Bug 411688]

[Tanu Kaskinen]

Adding to my previous comment: it could be argued that the bug is that
we transmit PCM audio instead of compressed audio. Adding support for
compressed audio should help with the network stress. Does anyone
volunteer to implement this?

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  pulseaudio floods network with multicast packets

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[Bug 1390474]

[Tanu Kaskinen]

The log shows that you're not using module-udev-detect. I don't know if
that's why things stopped working, but that would anyway be the first
thing to fix. Debian changed the packaging so that "pulseaudio-module-
udev" is now a separate package. You have probably disabled automatic
installation of recommended packages, because by default that package
should get installed automatically.

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Title:
  "Volume element PCM has 6 channels. That's too much!" - PA doesn't
  support elements with more than two channels

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[Bug 1521987]

[Tanu Kaskinen]

Created attachment 129226
0001-card-log-the-reason-for-profile-changes.patch (for 10.0)

Here's the updated patch.

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

(In reply to flat from comment #40)
> Same issue on xfce4, so it's not budgie.

It certainly was Budgie according to the log. Maybe xfce4 has a similar
bug in its volume control thingy.

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

Sorry for the long delay, I have trouble keeping up with all email...

(In reply to flat from comment #39)
> Created attachment 129322 [details]
> pulse audio verbose log with
> 0001-card-log-the-reason-for-profile-changes.patch 10.0
> 
> Sorry it took a couple days for it to happen again. Guess it helps when you
> compile to actually apply the patch... haha. Here's a log from it switching
> between digital and normal, even after I set it to the correct
> configurations it seems to switch. Looking through the log I think it might
> have to do with budgie-panel, but I'm not sure.

Yes, "Budgie Volume Control" changes the profile to digital.

> Also got another crash so
> that's the first log. Second log starts on line 4424

It's not a crash, it's a failure to start the daemon. module-esound-
protocol-unix can't setup itself, because the socket it needs is
reserved by some other process (probably another pulseaudio instance).
I'm not sure why this happens, but you can safely remove module-esound-
protocol-unix from /etc/pulse/default.pa. It's extremely unlikely that
you have any applications that still use esound.

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

The new log doesn't seem to have the patch applied. Can you try again?
You can check that pulseaudio is running with the patch by searching the
log for messages that contain "XXX" that are printed whenever the card
profile changes.

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  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

(In reply to flat from comment #34)
> Created attachment 129184 [details]
> pulse verbose log with 0001-card-log-the-reason-for-profile-changes.patch
> 
> So interestingly enough I can't reproduce the issue using git master.

The first log with the crash shows a switch to the digital profile.
Unfortunately, the crash happens before the log message I added is
printed.

The first log shows four "audio-volume-change" sounds being played in
quick succession during the crash. The volume changes are initiated by
gnome-settings-daemon. Did you change the volume manually while
disconnecting the headphones, or is this something that gnome-settings-
daemon does automatically?

> Attached are two logs as one attachment. The second log starts at line 5308
> (Didn't want to spam attachments). The first log shows pulse crashing upon
> disconnecting the headphones. The second log shows pulse switching between
> 5.1-stereo and Analog Stereo working fine. If you rebase the patch on 10.0 I
> can try it again to see what was going on. Version 10.0 is the current
> version from Arch Linux's repos. I assume a commit since then must have
> fixed something either intentionally or unintentionally.

I don't think it's fixed in master, but I can anyway make a patch for
10.0. I'll change it so that the profile change reason is printed in an
earlier phase. I'll also add some logging in an attempt to make it
easier to track down the reason for the crash if it happens again.

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

Thanks, the log is interesting. At line 5943 the headphones are plugged
in. Pulseaudio then decides to switch from the 5.1 profile to the analog
stereo profile, as expected. After switching the profile, however,
pulseaudio decides to immediately switch to digital output instead.
Unfortunately the log doesn't show why this happens.

If I write a patch that adds better logging, will you be able to apply
and test it? Here are some instructions for building and installing
pulseaudio from source:
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer/PulseAudioFromGit/

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  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

Created attachment 129175
0001-card-log-the-reason-for-profile-changes.patch

Here's the patch. It's written against the git master branch. If you
need it for some other PulseAudio version, let me know and I'll rebase
it.

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

The log doesn't match your problem description. The digital profile is
not activated at all. The log shows switching between headphones and
lineout in 2.1 mode. You wanted to use 5.1, so is your problem actually
that you get 2.1 audio instead of 5.1?

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  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

I believe PulseAudio 9.0 will handle your use case well. Please report
back if it doesn't.

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

(In reply to Eugene Crosser from comment #25)
> It happens when audio control application running and not running likewise,
> but desktop's audio widget in the panel is indeed running (Cinnamon and
> Unity, same manifestations). It is possible that this audio widget is indeed
> the culprit.

It's possible, but I don't think it's likely.

Can you attach a log file that shows what happens when you start with
analog output selected and headphones plugged in, and then unplug them
and plug them back in? Instructions for getting the log:
https://wiki.ubuntu.com/PulseAudio/Log

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  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

*** Bug 96237 has been marked as a duplicate of this bug. ***

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1521987]

[Tanu Kaskinen]

Do you happen to have the audio settings application open? If you do, it
can mess up the device selection:
https://bugzilla.gnome.org/show_bug.cgi?id=762932

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Title:
  Sound is not automatically switched back from headphones to speakers

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[Bug 1773167]

[Tanu Kaskinen]

Here's information about submitting alsa bugs:
http://alsa-project.org/main/index.php/Bug_Tracking

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Title:
  [Intel HDMI LPE driver] Audio is not working; pulseaudio consumes 100%
  of a single CPU core

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[Bug 1513384] Re: [pulseaudio-discuss] [PATCH] alsa-mixer: Add "Front Headphone" to headset mic path

[Tanu Kaskinen]

On Thu, 2015-11-05 at 09:59 +0100, David Henningsson wrote:
> The combination "Front Headphone" + "Headset Mic Phantom"
> was found on one the machines we enable. Without this patch,
> the headset mic appeared plugged in when nothing was plugged
> into the jack.
> 
> BugLink: https://bugs.launchpad.net/bugs/1513384
> Signed-off-by: David Henningsson 
> ---
>  src/modules/alsa/mixer/paths/analog-input-headset-mic.conf | 3 +++
>  1 file changed, 3 insertions(+)
> 
> diff --git a/src/modules/alsa/mixer/paths/analog-input-headset-
> mic.conf b/src/modules/alsa/mixer/paths/analog-input-headset-mic.conf
> index a67e1db..b669bc4 100644
> --- a/src/modules/alsa/mixer/paths/analog-input-headset-mic.conf
> +++ b/src/modules/alsa/mixer/paths/analog-input-headset-mic.conf
> @@ -32,6 +32,9 @@ required-any = any
>  [Jack Headphone]
>  state.plugged = unknown
>  
> +[Jack Front Headphone]
> +state.plugged = unknown
> +
>  [Jack Headphone Mic]
>  state.plugged = unknown

Looks good to me.

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Title:
  Missing "Front Headphone" in headset mic path

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[Bug 1458947]

[Tanu Kaskinen]

Updated the bug title.

The error is harmless, as you found out.

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Title:
  QinHeng Electronics CH345 MIDI adapter triggers error

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[Bug 1445358] Re: [pulseaudio-discuss] [PATCH] core-util, pactl: Make one localised and one non-localised version of pa_yes_no

[Tanu Kaskinen]

On Mon, 2015-04-20 at 17:41 +0200, David Henningsson wrote:
 We currently use pa_yes_no to write module arguments, so they can not be
 localised. Instead add a new pa_yes_no_localised function and use it in pactl
 (and thus, revert all other places to use the non-localised version).
 
 BugLink: https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1445358
 Signed-off-by: David Henningsson david.hennings...@canonical.com
 ---
  src/pulsecore/core-util.h |  4 
  src/utils/pactl.c | 20 ++--
  2 files changed, 14 insertions(+), 10 deletions(-)

Looks good to me.

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Title:
  Pulseaudio fails to run when system's language is in certain non-
  English locales

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[Bug 1390474] Re: Volume element PCM has 6 channels. That's too much! with Terratec PHASE26

[Tanu Kaskinen]

Kindly forwarding this bug, please express your thoughts on Launchpad.

I believe that was a direct request for me. My thoughts: it's
unfortunate that PulseAudio doesn't support mixer elements with more
than 2 volume channels. It would be nice if someone would fix that. Due
to lack of time, I won't be that someone, at least in the short (or even
medium) term.

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Title:
  Volume element PCM has 6 channels. That's too much! with Terratec
  PHASE26

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[Bug 421621]

[Tanu Kaskinen]

Changing bug state to fixed.

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Title:
  pavucontrol's window width depends on stream names, may get very wide

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[Bug 1289515] Re: Sound volume resets after making a call

[Tanu Kaskinen]

On Fri, 2014-03-21 at 10:27 +0100, David Henningsson wrote:
 In case a port has not yet been saved, which is e g often the case
 if a sink/source has only one port, reading volume/mute will be done
 without port, whereas writing volume/mute will be done with port.
 
 Work around this by setting a default port before the fixate hook,
 so module-device-restore can read volume/mute for the correct port.
 
 BugLink: https://bugs.launchpad.net/bugs/1289515
 Signed-off-by: David Henningsson david.hennings...@canonical.com

Thanks, this also fixes this bug:
https://bugs.freedesktop.org/show_bug.cgi?id=55262

A couple of comments below.

 ---
  src/pulsecore/sink.c   |   11 +++
  src/pulsecore/source.c |   11 +++
  2 files changed, 22 insertions(+)
 
 diff --git a/src/pulsecore/sink.c b/src/pulsecore/sink.c
 index 08143e9..9c4b0c3 100644
 --- a/src/pulsecore/sink.c
 +++ b/src/pulsecore/sink.c
 @@ -235,6 +235,17 @@ pa_sink* pa_sink_new(
  pa_device_init_icon(data-proplist, true);
  pa_device_init_intended_roles(data-proplist);
  
 +if (!data-active_port  !data-save_port) {

How is data-save_port relevant here? If active_port is NULL, save_port
should always be false (true wouldn't have any meaning).

 +void *state;
 +pa_device_port *p, *p2 = NULL;
 +
 +PA_HASHMAP_FOREACH(p, data-ports, state)
 +if (!p2 || p-priority  p2-priority) {
 +p2 = p;
 +pa_sink_new_data_set_port(data, p2-name);

I'd prefer calling pa_sink_new_data_set_port() only once, i.e. outside
the loop.

 +}
 +}

This partially duplicates the code that is run later:

if (!s-active_port) {
void *state;
pa_device_port *p;

PA_HASHMAP_FOREACH(p, s-ports, state) {
if (p-available == PA_AVAILABLE_NO)
continue;

if (!s-active_port || p-priority  s-active_port-priority)
s-active_port = p;
}
if (!s-active_port) {
PA_HASHMAP_FOREACH(p, s-ports, state)
if (!s-active_port || p-priority  s-active_port-priority)
s-active_port = p;
}
}

I think we should do the fallback port initialization only once, in the
location where you added the new code. We should use the full logic of
the old fallback initialization code.

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** Bug watch added: freedesktop.org Bugzilla #55262
   https://bugs.freedesktop.org/show_bug.cgi?id=55262

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  Sound volume resets after making a call

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[Bug 1256511]

[Tanu Kaskinen]

Ok, so module-card-restore claims to be selecting the analog profile
when the card is created, but actually the hdmi profile gets selected.
It might be that module-switch-on-port-available is overriding the
profile choice made by module-card-restore. I believe that's not
supposed to happen, so that's the first bug.

Another very weird bug is that when the analog profile is activated via
pactl, the hdmi sink doesn't get removed. This was actually visible
already in your initial bug description, but I didn't notice that.

I really would like to get these issues fixed by the time 6.0 is
released, so I'll make this a release blocker, but I'm afraid it will
take a while before I have time to properly start working on this bug.

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Title:
  Sounds settings lost after reboot

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[Bug 1256511]

[Tanu Kaskinen]

module-switch-on-port-available should not automatically change the card
profile from analog to hdmi, and when you switch the profile with
pavucontrol, that choice should be remembered... I'd like to see
PulseAudio logs.

Could you do this:

1) Disable autospawning:
echo autospawn = no  ~/.config/pulse/client.conf

2) Stop pulseaudio:
killall pulseaudio

3) Start pulseaudio in a terminal:
pulseaudio -vvv

4) Set the card profile:
pactl set-card-profile alsa_card.pci-_00_1b.0 output:analog-stereo

5) Stop pulseaudio with Ctrl-C and save the log output.

6) Start pulseaudio again:
pulseaudio -vvv

7) Check the active card profile:
pactl list cards

8) Stop pulseaudio with Ctrl-C and save the log output.

9) Attach the logs from steps 5 and 8 and the pactl output from step 7
to this bug.

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  Sounds settings lost after reboot

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[Bug 1236965] Re: [pulseaudio-discuss] [PATCH] alsa-mixer: Add internal surround speaker elements

[Tanu Kaskinen]

On Mon, 2013-10-14 at 16:16 +0200, David Henningsson wrote:
 These kcontrol names have started to show up lately, especially in
 combination with surround internal speakers.
 
 BugLink: https://bugs.launchpad.net/bugs/1236965
 Signed-off-by: David Henningsson david.hennings...@canonical.com
 ---
  .../alsa/mixer/paths/analog-output-speaker.conf|   31 
 
  1 file changed, 31 insertions(+)
 
 diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker.conf 
 b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
 index ccd438f..d79fad1 100644
 --- a/src/modules/alsa/mixer/paths/analog-output-speaker.conf
 +++ b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
 @@ -38,6 +38,11 @@ required-any = any
  state.plugged = unknown
  state.unplugged = unknown
  
 +[Jack Speaker Front Phantom]
 +required-any = any
 +state.plugged = unknown
 +state.unplugged = unknown
 +
  [Element Hardware Master]
  switch = mute
  volume = merge
 @@ -88,6 +93,13 @@ override-map.1 = all-front
  override-map.2 = front-left,front-right
  required-any = any
  
 +[Element Speaker Front]
 +switch = mute
 +volume = merge
 +override-map.1 = all-front
 +override-map.2 = front-left,front-right
 +required-any = any
 +
  [Element Rear]
  switch = mute
  volume = merge
 @@ -107,12 +119,25 @@ override-map.1 = all-rear
  override-map.2 = rear-left,rear-right
  required-any = any
  
 +[Element Speaker Surround]
 +switch = mute
 +volume = merge
 +override-map.1 = all-rear
 +override-map.2 = rear-left,rear-right
 +required-any = any
 +
  [Element Side]
  switch = mute
  volume = merge
  override-map.1 = all-side
  override-map.2 = side-left,side-right
  
 +[Element Speaker Side]
 +switch = mute
 +volume = merge
 +override-map.1 = all-side
 +override-map.2 = side-left,side-right
 +
  [Element Center]
  switch = mute
  volume = merge
 @@ -152,4 +177,10 @@ volume = merge
  override-map.1 = all-center
  override-map.2 = all-center,lfe
  
 +[Element Speaker CLFE]
 +switch = mute
 +volume = merge
 +override-map.1 = all-center
 +override-map.2 = all-center,lfe
 +
  .include analog-output.conf.common

Should these elements be muted in other paths?

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Title:
  pulseaudio does not recognize internal speakers [P180HMx, Realtek
  ALC892, Speaker, Internal] volume slider problem

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[Bug 1159687] Re: [pulseaudio-discuss] [PATCH] alsa-mixer: Add Headset Playback|Capture element

[Tanu Kaskinen]

On Mon, 2013-03-25 at 15:00 +0100, David Henningsson wrote:
 Found on Logitech B530 USB Headset / kernel 3.8. Because we don't
 have different path for headset and headphone today, just add
 Headset to the existing headphone path.

Wouldn't it be better to have a headset path for output too, at least so
that the UI can show a more accurate description of the device? This
might be useful for automatic routing too, if the path configuration
would set the device.form_factor property.

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Title:
  [USB-Audio - Logitech B530 USB Headset, recording] Recording problem

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[Bug 1159687] Re: [pulseaudio-discuss] [PATCH] alsa-mixer: Add Headset Playback|Capture element

[Tanu Kaskinen]

On Wed, 2013-03-27 at 11:03 +0100, David Henningsson wrote:
 On 03/27/2013 10:52 AM, Tanu Kaskinen wrote:
  On Mon, 2013-03-25 at 15:00 +0100, David Henningsson wrote:
  Found on Logitech B530 USB Headset / kernel 3.8. Because we don't
  have different path for headset and headphone today, just add
  Headset to the existing headphone path.
 
  Wouldn't it be better to have a headset path for output too, at least so
  that the UI can show a more accurate description of the device? This
  might be useful for automatic routing too, if the path configuration
  would set the device.form_factor property.
 
 In 99% of all cases, a headset has a Headphone control. A Headset 
 control is really the exception. And given the HDA specification that is 
 unlikely to change in the near future.
 
 As such, I think the extra path is not worth the cost.

OK, you have my ack for the patch then.

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Title:
  [USB-Audio - Logitech B530 USB Headset, recording] Recording problem

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[Bug 1109298] Re: [alsa-devel] [PATCH] alsa-lib: Add sys/types.h to include list

[Tanu Kaskinen]

On Tue, 2013-02-12 at 10:06 +0100, David Henningsson wrote:
 This is needed by snd_pcm_format_silence* functions which
 return u_int*_t. It was discovered while trying to compile ALSA
 programs with eglibc 2.17.
 
 Credits to Richard Shaw, Gary Buhrmaster, Matthieu Baerts and
  Adam Conrad for this fix.
 
 BugLink: https://bugs.launchpad.net/bugs/1109298
 BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=885306
 Signed-off-by: David Henningsson david.hennings...@canonical.com
 ---
  include/asoundlib-head.h |1 +
  1 file changed, 1 insertion(+)
 
 diff --git a/include/asoundlib-head.h b/include/asoundlib-head.h
 index 20c8a68..6edbab0 100644
 --- a/include/asoundlib-head.h
 +++ b/include/asoundlib-head.h
 @@ -31,6 +31,7 @@
  #include unistd.h
  #include stdio.h
  #include stdlib.h
 +#include sys/types.h

int*_t and uint*_t (not u_int*_t) are standard in C99, and they are
available in stdint.h or inttypes.h (both work, inttypes.h contains also
the PRI* constants for the corresponding printf() format specifiers).
Perhaps it would be better to use one of those headers instead? That
would require converting u_int*_t usage to uint*_t, though, so it would
take some extra work...

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Title:
  [Regression] error when trying to compile a program which uses
  alsa/asoundlib.h with EGLIBC 2.17

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[Bug 1069232]

[Tanu Kaskinen]

(In reply to comment #7)
 Same here.  Really annoying for VLC user.
 
 I'm not familar with pa so I cannot say whether this patch is has something
 wrong or not.  !!!But this commit broken the userspace!!!   Should
 it be reverted?
 
 It's 7 month after this bug has been reported.  Should the developer say
 something to show that this bug is not totally ignored?

Sure, I can say something. The problematic commit implements a useful
feature, so it probably won't be simply reverted. A better fix is needed
(patches very welcome). There have been fixes related to the sample rate
updating since pulseaudio 2.0, so it's worth retrying with pulseaudio
3.0, but I don't remember that those fixes would have been timing
related, so don't expect too much.

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Title:
  Sound lags when opening video files

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[Bug 1069232]

[Tanu Kaskinen]

(In reply to comment #8)
 (In reply to comment #7)
  Same here.  Really annoying for VLC user.
  
  I'm not familar with pa so I cannot say whether this patch is has something
  wrong or not.  !!!But this commit broken the userspace!!!   Should
  it be reverted?
  
  It's 7 month after this bug has been reported.  Should the developer say
  something to show that this bug is not totally ignored?
 
 Sure, I can say something. The problematic commit implements a useful
 feature, so it probably won't be simply reverted. A better fix is needed
 (patches very welcome). There have been fixes related to the sample rate
 updating since pulseaudio 2.0, so it's worth retrying with pulseaudio 3.0,
 but I don't remember that those fixes would have been timing related, so
 don't expect too much.

Thinking a bit more, reverting the patch (or maybe commenting out the
code would be better) is maybe the right approach after all, if nobody
submits a better fix. Removing the code that was added that patch
shouldn't really break anything, the only effect should be that there's
sometimes unnecessary resampling, which is better than breaking the
timing of applications.

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Title:
  Sound lags when opening video files

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[Bug 1069232]

[Tanu Kaskinen]

Before removing any functionality, this must be reproducible by a
pulseaudio developer, though. Otherwise there's no way to know when it's
safe to add the functionality back. I can't reproduce this with vlc
version 2.0.3 and pulseaudio's current development version. I'll try
installing vlc 2.0.1 and pulseaudio 2.0 later today.

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Title:
  Sound lags when opening video files

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[Bug 1069232]

[Tanu Kaskinen]

Any further hints for reproducing? I have pulseaudio 2.0 installed, and
since vlc version shouldn't matter, I still have vlc 2.0.3, not 2.0.1
that the original reporter used. A/V sync is just fine for me, with both
pulseaudio and alsa (through pulseaudio) backends of vlc.

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  Sound lags when opening video files

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[Bug 1069232]

[Tanu Kaskinen]

I was now able to reproduce this. For me this doesn't seem to happen
100% of the time (I test by running vlc ~/misc/sync_test.mp4,
observing the A/V sync, closing vlc, waiting for the sink to suspend,
and trying again). The small hardware buffer size that I have (371 ms)
might make noticing the delay harder, if the delay is variable, but if
it's always one full hw buffer size in length, then I should be able to
reliably notice it... I'll build a custom kernel next with a bigger
audio buffer, and see if that changes anything.

Btw, Remi, you said that this affects all VLC versions [...] with
PulseAudio 2.x. Do you mean that this doesn't affect 3.0?

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  Sound lags when opening video files

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[Bug 1069232]

[Tanu Kaskinen]

With bigger hw buffer I was able to reproduce the problem 100% of time
on pulseaudio 2.0. With the current development version I can't
reproduce the problem. Thus, resolving the bug as fixed.

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[Bug 1058200] Re: [pulseaudio-discuss] [PATCH] stream: Return error in case a client peeks to early

[Tanu Kaskinen]

On Tue, 2012-11-06 at 08:36 +0100, David Henningsson wrote:
 On 11/05/2012 07:13 PM, Tanu Kaskinen wrote:
  On Sun, 2012-11-04 at 23:09 +0100, David Henningsson wrote:
  On 11/04/2012 02:22 PM, Tanu Kaskinen wrote:
  On Sat, 2012-11-03 at 19:51 +0100, David Henningsson wrote:
  Returning NULL seems to be the right thing to do here, even if
  gnome-control-center does not handle that very well IIRC. So we might
  need an additional patch in g-c-c.
  So assuming I commit a patch doing that. If somebody else wants to add
  logic to figure out how large the hole is, that could be discussed
  separately.
  Any objections?
 
  It's not clear what you meant by add logic to figure out how large the
  hole is. Add to where? pa_stream_peek() or gnome-control-center?
 
  I was referring to your earlier comment An addition: the client
  probably wants to know how large the hole is., i e pa_stream_peek.
 
  To me, reporting the hole length in the nbytes parameter of
  pa_stream_peek() seems like the right thing to do, so I hope your patch
  will do this.
 
  It does not: I was just suggesting to discuss that separately.
 
  Ok. Well, I'm suggesting to not make these issues separate, since it's a
  small thing to decide and implement, and doing a separate fix would be
  fixing a fix.
 
 I don't have an issue with step-by-step improvements. I see this as such 
 (if the second step is to be seen as an improvement).
 
  Does someone have an issue with returning the hole size
  in the nbytes argument?
 
 Not if you're volunteering to do the work. All I care about is that PA 
 clients can crash for this reason, and I'd like that to stop happening.

Cool, I certainly volunteer to do the work. It should be trivial, so
I'll probably send a patch today.

 Note: gnome-control-center needs a corresponding fix either way, as it 
 currently checks the length rather than the NULL pointer, IIRC.

Mmm, I guess I should take this task too.

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Title:
  [sound]: gnome-control-center crashed with SIGABRT in
  pa_memblock_acquire()

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[Bug 1058200] Re: [pulseaudio-discuss] [PATCH] stream: Return error in case a client peeks to early

[Tanu Kaskinen]

On Sun, 2012-11-04 at 23:09 +0100, David Henningsson wrote:
 On 11/04/2012 02:22 PM, Tanu Kaskinen wrote:
  On Sat, 2012-11-03 at 19:51 +0100, David Henningsson wrote:
  Returning NULL seems to be the right thing to do here, even if
  gnome-control-center does not handle that very well IIRC. So we might
  need an additional patch in g-c-c.
  So assuming I commit a patch doing that. If somebody else wants to add
  logic to figure out how large the hole is, that could be discussed
  separately.
  Any objections?
 
  It's not clear what you meant by add logic to figure out how large the
  hole is. Add to where? pa_stream_peek() or gnome-control-center?
 
 I was referring to your earlier comment An addition: the client 
 probably wants to know how large the hole is., i e pa_stream_peek.
 
  To me, reporting the hole length in the nbytes parameter of
  pa_stream_peek() seems like the right thing to do, so I hope your patch
  will do this.
 
 It does not: I was just suggesting to discuss that separately.

Ok. Well, I'm suggesting to not make these issues separate, since it's a
small thing to decide and implement, and doing a separate fix would be
fixing a fix. Does someone have an issue with returning the hole size
in the nbytes argument?

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Title:
  [sound]: gnome-control-center crashed with SIGABRT in
  pa_memblock_acquire()

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[Bug 1058200] Re: [pulseaudio-discuss] [PATCH] stream: Return error in case a client peeks to early

[Tanu Kaskinen]

On Sat, 2012-11-03 at 19:51 +0100, David Henningsson wrote:
 2012-11-03 17:19, Colin Guthrie skrev:
  'Twas brillig, and Tanu Kaskinen at 05/10/12 13:58 did gyre and gimble:
  On Wed, 2012-10-03 at 08:50 +0200, David Henningsson wrote:
  On 10/02/2012 10:38 PM, Tanu Kaskinen wrote:
  On Mon, 2012-10-01 at 17:06 +0200, David Henningsson wrote:
  If there is no silence memblock and no data, pa_memblockq_peek can
  return NULL. In this case, do not crash on an assertion in
  pa_memblock_acquire, but instead return a proper error to the client.
  If there is no data in the buffer, pa_stream_peek() is supposed to
  return NULL according to the documentation. And it does that: if there's
  no data, pa_memblock_peek() will return a negative value, causing
  pa_stream_peek() to return NULL.
 
  The problem is the case where the buffer does contain data, but not at
  the read index. That is, there is a hole in the buffer. The client
  documentation doesn't have any warnings about holes, so the only safe
  way to handle holes is to return silence. Fixing this should be a simple
  matter of giving a silence memchunk when creating record_memblockq.
  I'm not so sure. Silence, as in all zeroes, might work for S16 audio
  data, but what about other formats? Compressed audio? Peak audio (which
  I think is the case here)? Etc.
  Good point. Regarding PCM, if pa_memchunk_silence() is used, the
  function will take care of filling the memory with appropriate content.
  But that doesn't work with compressed audio.
 
  Also maybe it could also be valuable for the client to distinguish
  between no data available, and valid zero data.
 
  How about returning NULL and adding to the documentation something like:
 
  -If no data is available this will return a NULL pointer.
  +If no data is available (at the current read position), this will
  return a NULL pointer.
  An addition: the client probably wants to know how large the hole is. It
  might be possible to figure that out somehow from the read index, but I
  think it would make sense to return the hole size in the length
  parameter.
  This discussion seemed to stagnate. Is this worth fixing/documenting for
  the 3.0 release?
 
  Col
 
 
 Returning NULL seems to be the right thing to do here, even if 
 gnome-control-center does not handle that very well IIRC. So we might 
 need an additional patch in g-c-c.
 So assuming I commit a patch doing that. If somebody else wants to add 
 logic to figure out how large the hole is, that could be discussed 
 separately.
 Any objections?

It's not clear what you meant by add logic to figure out how large the
hole is. Add to where? pa_stream_peek() or gnome-control-center?

To me, reporting the hole length in the nbytes parameter of
pa_stream_peek() seems like the right thing to do, so I hope your patch
will do this.

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  [sound]: gnome-control-center crashed with SIGABRT in
  pa_memblock_acquire()

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[Bug 689915] Re: [pulseaudio-discuss] [RFC PATCH] bluetooth: Fix assertion failure (decoded == a2dp-frame_length)

[Tanu Kaskinen]

On Wed, 2012-10-10 at 09:17 +0200, David Henningsson wrote:
 According to the patch author, Karl Hegbloom:
 The assertion is incorrectly testing for '==' when logically it
 should test for '='.
 
 BugLink: https://bugs.launchpad.net/bugs/689915
 Signed-off-by: David Henningsson david.hennings...@canonical.com
 ---
 
 This is a patch I found attached to a bug in Ubuntu. I don't know
 much about bluetooth, and the error was reported against something
 around 0.9.22. But I'm sending it here, hoping that one of our
 bluetooth people can review and say whether this is correct or not.
 
  src/modules/bluetooth/module-bluetooth-device.c |4 ++--
  1 file changed, 2 insertions(+), 2 deletions(-)
 
 diff --git a/src/modules/bluetooth/module-bluetooth-device.c 
 b/src/modules/bluetooth/module-bluetooth-device.c
 index e3ec6ae..06c783d 100644
 --- a/src/modules/bluetooth/module-bluetooth-device.c
 +++ b/src/modules/bluetooth/module-bluetooth-device.c
 @@ -999,9 +999,9 @@ static int a2dp_process_push(struct userdata *u) {
  a2dp-frame_length = sbc_get_frame_length(a2dp-sbc);
  
  pa_assert_fp((size_t) decoded = to_decode);
 -pa_assert_fp((size_t) decoded == a2dp-frame_length);
 +pa_assert_fp((size_t) decoded = a2dp-frame_length);
  
 -pa_assert_fp((size_t) written == a2dp-codesize);
 +pa_assert_fp((size_t) written = a2dp-codesize);
  
  p = (const uint8_t*) p + decoded;
  to_decode -= decoded;

I'm no SBC expert either, but I had a look at sbc_decode()
implementation, and to me it looks like at least decoded ==
a2dp-frame_length should hold. Extrapolating from that, written ==
a2dp-codesize probably should hold too. And it makes sense:
sbc_decode() is supposed to decode one frame, and I think one frame will
always have encoded size of a2dp-frame_length and decoded size of
a2dp-codesize.

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Title:
  pulseaudio aborts with assertion decoded == a2dp-frame_length in
  function a2dp_process_push()

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[Bug 1058200] Re: [pulseaudio-discuss] [PATCH] stream: Return error in case a client peeks to early

[Tanu Kaskinen]

On Wed, 2012-10-03 at 08:50 +0200, David Henningsson wrote:
 On 10/02/2012 10:38 PM, Tanu Kaskinen wrote:
  On Mon, 2012-10-01 at 17:06 +0200, David Henningsson wrote:
  If there is no silence memblock and no data, pa_memblockq_peek can
  return NULL. In this case, do not crash on an assertion in
  pa_memblock_acquire, but instead return a proper error to the client.
 
  If there is no data in the buffer, pa_stream_peek() is supposed to
  return NULL according to the documentation. And it does that: if there's
  no data, pa_memblock_peek() will return a negative value, causing
  pa_stream_peek() to return NULL.
 
  The problem is the case where the buffer does contain data, but not at
  the read index. That is, there is a hole in the buffer. The client
  documentation doesn't have any warnings about holes, so the only safe
  way to handle holes is to return silence. Fixing this should be a simple
  matter of giving a silence memchunk when creating record_memblockq.
 
 I'm not so sure. Silence, as in all zeroes, might work for S16 audio 
 data, but what about other formats? Compressed audio? Peak audio (which 
 I think is the case here)? Etc.

Good point. Regarding PCM, if pa_memchunk_silence() is used, the
function will take care of filling the memory with appropriate content.
But that doesn't work with compressed audio.

 Also maybe it could also be valuable for the client to distinguish 
 between no data available, and valid zero data.
 
 How about returning NULL and adding to the documentation something like:
 
 -If no data is available this will return a NULL pointer.
 +If no data is available (at the current read position), this will 
 return a NULL pointer.

An addition: the client probably wants to know how large the hole is. It
might be possible to figure that out somehow from the read index, but I
think it would make sense to return the hole size in the length
parameter.

-- 
Tanu

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Title:
  [sound]: gnome-control-center crashed with SIGABRT in
  pa_memblock_acquire()

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[Bug 1058200] Re: [pulseaudio-discuss] [PATCH] stream: Return error in case a client peeks to early

[Tanu Kaskinen]

On Mon, 2012-10-01 at 17:06 +0200, David Henningsson wrote:
 If there is no silence memblock and no data, pa_memblockq_peek can
 return NULL. In this case, do not crash on an assertion in
 pa_memblock_acquire, but instead return a proper error to the client.

If there is no data in the buffer, pa_stream_peek() is supposed to
return NULL according to the documentation. And it does that: if there's
no data, pa_memblock_peek() will return a negative value, causing
pa_stream_peek() to return NULL.

The problem is the case where the buffer does contain data, but not at
the read index. That is, there is a hole in the buffer. The client
documentation doesn't have any warnings about holes, so the only safe
way to handle holes is to return silence. Fixing this should be a simple
matter of giving a silence memchunk when creating record_memblockq.

-- 
Tanu

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Title:
  [sound]: gnome-control-center crashed with SIGABRT in
  pa_memblock_acquire()

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[Bug 939161] Re: [alsa-devel] [PATCH] ALSA: hda - Remove ignore_misc_bit

[Tanu Kaskinen]

On Fri, 2012-09-07 at 16:32 +0200, Takashi Iwai wrote:
  Hmm, I'm afraid it's not so easy to turn off PulseAudio's jack detection 
  without being root. Maybe it's possible using some .asoundrc magic?
 
 That's bad.  asoundrc is equally cryptic like udev rules :)
 
 Honestly, can't the whole /usr/share/pulseaudio/* files be specified
 via an environment variable, or automatically looking at
 ~/.pulse/alsa-mixer/paths/* or such?

FWIW, I'm in favor of looking at the home directory...

-- 
Tanu

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Title:
  Hardware not showed in Gnome-Control-Center. - [M31EI Series, Realtek
  ALC861, Green Headphone Out, Rear]

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[Bug 724383] Re: AMD64 Netboot Documentation: typo in instructions (suggested replacement text attached)

[Tanu Kaskinen]

I believe the command should be

# zcat boot.img.gz  /dev/sdX

instead of

# zcat boot.img.gz  /dev/sdX1

So just remove the 1 from the end. This is what solved my problems
with booting from the USB stick, and this is also what the corresponding
Debian documentation says:
http://www.debian.org/releases/stable/amd64/ch04s03.html.en

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Title:
  AMD64 Netboot Documentation: typo in instructions (suggested
  replacement text attached)

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[Bug 724383] Re: AMD64 Netboot Documentation: typo in instructions (suggested replacement text attached)

[Tanu Kaskinen]

To be clear, this bug affects also the 12.04 installation instructions,
in addition to 10.10 that the original reporter was using/installing.

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Title:
  AMD64 Netboot Documentation: typo in instructions (suggested
  replacement text attached)

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