Joerg Anders wrote:
> users which have no hardware MIDI synthesizer und rely
> to TiMidity++
>
> (http://timidity.sourceforge.net/)
>
> http://vsr.informatik.tu-chemnitz.de/staff/jan/nted/doc/ch01s50.html
>
> This text is an own chapter of the NtEd documentation.
> Many users wrote me they can
Hi,
I triage some sound bugs in Ubuntu and for one user I found the
following lines in the "pulseaudio -" log:
D: sink-input.c: Requesting rewind due to corking
D: module-suspend-on-idle.c: Sink alsa_output.default becomes idle,
timeout in 5 seconds.
D: sink-input.c: Requesting rewind due to
Lennart Poettering wrote:
> On Fri, 25.12.09 03:57, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>> Hi,
>>
>> I triage some sound bugs in Ubuntu and for one user I found the
>> following lines in the "pulseaudio -" log:
>>
>
Rustom Mody wrote:
> Well one correction (mine and the wikis :-)
> The wiki talks of changing /etc/default/timidity.cfg
>
> There is no such file.
> The 2 files are /etc/timidity/timidity.cfg and /etc/default/timidity
>
> Evidently the wiki is referring to the second whereas I took it to be the
Bill Cox wrote:
> On Thu, Dec 24, 2009 at 12:27 PM, Lennart Poettering
> wrote:
>> We actually cover that inside of gdm, where you can get access to the
>> boot messages.
>>
>> Lennart
>
> Speakup doesn't stop reading when the user logs into Gnome. When we
> type Ctrl+Alt+F1, we get a console sc
(Answer to both Colin and Bill)
Colin Guthrie wrote:
> 'Twas brillig, and Bill Cox at 02/01/10 15:03 did gyre and gimble:
>> Hi, David.
>>
>> On Sat, Jan 2, 2010 at 1:08 AM, David Henningsson
>> wrote:
>>> I was just thinking, and this idea is perhaps n
Hi,
I'm trying to get RtKit for my application up and running, and in the lack
of an rtkit mailing list I'm using this one as the closest I could find
:-)
I'm using the reference implementation almost unmodified and end up with
an "org.freedesktop.DBus.Error.AccessDenied" - "Operation not permitt
> On Tue, 05.01.10 07:39, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>> Hi,
>>
>> I'm trying to get RtKit for my application up and running, and in the
>> lack
>> of an rtkit mailing list I'm using this one as the closest
Hi,
is there a way, e g via pacmd, to get the current default/fallback
playback (and record) device? There is no get-default-sink command.
List-sinks shows a "priority" item, is this something I can reliably
use, i e take the highest value to get the current fallback device?
// David
Colin Guthrie wrote:
> 'Twas brillig, and David Henningsson at 06/01/10 11:33 did gyre and gimble:
>> is there a way, e g via pacmd, to get the current default/fallback
>> playback (and record) device? There is no get-default-sink command.
>> List-sinks shows a "prio
In the spirit of a recent thread here, I have a similar minor problem. I
have an ESI Juli@ card, which PulseAudio names as "VT1720/24
[Envy24PT/HT] PCI Multi-Channel Audio Controller". I'd prefer "ESI
Juli@" as ALSA suggests, and I assume most other ESI Juli@ users would
also prefer that.
I can pr
Ng Oon-Ee wrote:
> On Thu, 2010-01-07 at 19:51 +0100, David Henningsson wrote:
>> In the spirit of a recent thread here, I have a similar minor problem. I
>> have an ESI Juli@ card, which PulseAudio names as "VT1720/24
>> [Envy24PT/HT] PCI Multi-Channel Audio Controller
elated bug: https://bugs.launchpad.net/bugs/485488
The patch is for the alsa-plugins tree at git.alsa-projects.org.
// David
>From 2e9f8fa4d6f7a0cabe5aea1cbfa98980cdfb6d84 Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Sat, 9 Jan 2010 09:09:14 +0100
Subject: [PATCH] pulse: Fix i
Colin Guthrie wrote:
> 'Twas brillig, and David Henningsson at 09/01/10 09:00 did gyre and gimble:
>> The pulse ALSA plugin has been known, for a while, to not work properly,
>> causing underruns, hangs etc. I sat down yesterday trying to figure it
>> out, and I&#
> 'Twas brillig, and David Henningsson at 09/01/10 09:00 did gyre and
> gimble:
>> The pulse ALSA plugin has been known, for a while, to not work properly,
>> causing underruns, hangs etc. I sat down yesterday trying to figure it
>> out, and I'm pretty certain
>> 'Twas brillig, and David Henningsson at 09/01/10 09:00 did gyre and
>> gimble:
>>> The pulse ALSA plugin has been known, for a while, to not work
>>> properly,
>>> causing underruns, hangs etc. I sat down yesterday trying to figure it
>>>
David Henningsson wrote:
>> 'Twas brillig, and David Henningsson at 09/01/10 09:00 did gyre and
>> gimble:
>>> The pulse ALSA plugin has been known, for a while, to not work properly,
>>> causing underruns, hangs etc. I sat down yesterday trying to figure it
&
Lennart Poettering wrote:
> On Sat, 09.01.10 10:00, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>> The pulse ALSA plugin has been known, for a while, to not work properly,
>> causing underruns, hangs etc. I sat down yesterday trying to figure it
>>
David Henningsson wrote:
> Lennart Poettering wrote:
>> On Sat, 09.01.10 10:00, David Henningsson (launchpad@epost.diwic.se)
>> wrote:
>>
>>> The pulse ALSA plugin has been known, for a while, to not work properly,
>>> causing underruns, hangs etc. I
> On Wed, 06.01.10 05:53, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>> Thanks for the pointers, they were most helpful.
>>
>> First, RtKit version is 0.4-0ubuntu2, OS is Ubuntu 10.04 (the
>> development
>> version). If that matters.
Somewhat due to the lack of feedback and better ideas in this forum, I'm
proposing Ubuntu to remove the underrun handling in alsa-plugins-pulse.
If you think this is a bad idea and/or have a better one, now is the
time to speak up.
The reasoning behind this is:
1) If pulseaudio gets an underrun,
To complete the previous patch that implemented properties in rtkit,
here's the client-side code that tests that the properties work, and make
them more accessible for the casual C programmer.
// David
>From 309659d47396544cd2b8d97ac5dac4460f543723 Mon Sep 17 00:00:00 2001
From: David Hen
Colin Guthrie wrote:
> 'Twas brillig, and Jeremy Nickurak at 09/02/10 19:24 did gyre and gimble:
> This whole thing has been discussed to death, and I really don't feel
> like being drawn into the whole thing again.
>From what I've read here, I'm afraid it's going to keep coming up until
we solve
Colin Guthrie wrote:
> 'Twas brillig, and David Henningsson at 09/02/10 21:52 did gyre and gimble:
>> I wrote down a few use cases here, I'm sure there are more:
>>
>> https://wiki.ubuntu.com/BluePrints/multiuser-soundcards-pulseaudio
>
> For user Foo, the
Lennart Poettering wrote:
> On Tue, 09.02.10 22:52, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>> There are just too many people for where the ordinary PA setup (all
>> soundcards are of exclusive use to the person logged into the current X
>> session) is n
Lennart Poettering wrote:
> On Wed, 10.02.10 07:14, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>> But printers are more of a system-wide resource, and for some use cases,
>> so is the soundcard.
>
> This is nonsense. I am not sure how your ears
Tristin Celestin wrote:
> This application I am writing a plugin for has a mode where it uses a timer
> to drive the delivery of sound. At every tick, if some condition is filled,
> the buffer is filled with data, otherwise the application runs.
>
> I managed to write a plugin that successfully
Tristin Celestin wrote:
> Is there a downside to making a version of pa_stream_writable_size available
> in the
> simple API?
Good question. In your use case, can see the use for a function
returning how many bytes that can be written to pa_simple_write without
pa_simple_write blocking.
> Why w
> well. (I'd use the ALSA->PA plugin, but as reported earlier, this
> doesn't work either, stuttering like crazy and frequently locking up.
Even with this patch?
http://launchpadlibrarian.net/37640450/0001-pulse-Fix-buffer-pointer-issues-improves-recovering-.patch
// David
> Heya,
>
> just wanted to point everybody to this new "canned response" type note
> I just added to the wiki:
>
> http://pulseaudio.org/wiki/BadDecibel
>
> I wrote a little tool that can be used to track invalid dB data
> exposed by ALSA drivers. Incorrect dB data usually means that playing
> a st
Chen: let me know if you want this patch as a merge proposal as
well. There is no bug in Launchpad AFAIK.
>From fa2dcce3201753bff94adba99b6bc36a7eaf57f9 Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Sun, 14 Mar 2010 20:20:12 +0100
Subject: [PATCH] Fix crash on jack server shutdown
Lennart Poettering wrote:
> On Sun, 14.03.10 20:50, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>> On sink unlinking, existing sink inputs are moved, which in turn calls
>> a get latency callback, which references the jack client. Therefore,
>> mak
Lennart Poettering wrote:
> On Tue, 23.03.10 00:36, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>>> What is missing is that the jack loop does not depend on the PA sink to
>>> be around resp. the PA IO loop doesn't call into jack when it i
Brendon Costa wrote:
> Hi again,
>
> Thanks for the info on the default devices. I have it working fine now.
>
> However I have run into another problem. When I run my application,
> the sinks are working fine and audio sounds good. However the sources
> are not working correctly. If i run my app
Colin Guthrie wrote:
> 'Twas brillig, and David Henningsson at 16/01/10 07:07 did gyre and gimble:
>> David Henningsson wrote:
>>> Lennart Poettering wrote:
>>>> On Sat, 09.01.10 10:00, David Henningsson (launchpad@epost.diwic.se)
>>>> wrote:
David Henningsson wrote:
> Colin Guthrie wrote:
>> 'Twas brillig, and David Henningsson at 16/01/10 07:07 did gyre and gimble:
>>> David Henningsson wrote:
>>>> Lennart Poettering wrote:
>>>>> On Sat, 09.01.10 10:00, David Henningsson (launchpad.
Nix wrote:
> On 18 Apr 2010, David Henningsson spake thusly:
>
>> Both of them were tested by Daniel, and AFAIK neither have been applied
>> upstream.
>
> These are both alsa-plugin things, right?
Yes.
> In that case, they can't fix the hangs I reported bac
Maarten Lankhorst wrote:
> Well, the use case would be wine's wineserver. On windows programs
> usually set audio threads to THREAD_PRIORITY_TIME_CRITICAL to indicate
> that they have to have a certain priority. But in windows thread handles
> are global, so doing it inside wine's 'ntdll' library w
I'll just resend my patch after having discussed it on LAC with Lennart.
Compared to the previous patch this patch also adds a comment Lennart
wanted, and also does the same change for jack sources.
// David>From 0dddabf79b51600db45b5205e85b4589a56825df Mon Sep 17 00:00:00 2001
Fro
On 2010-04-25 22:11, Lennart Poettering wrote:
> On Sun, 25.04.10 21:41, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
>>
>>> which handle corresponds to the thread, or even if that handle is local
>>> or not. Wineserver controls this information
On 2010-05-09 13:06, Mihai Sucan wrote:
> The problem is with mplayer. I cannot play *any* sound with it, and worse,
> it causes PulseAudio to go "bonkers". Once I start mplayer, I cannot hear
> any sound, from any application.
>
> I tested mplayer from the default Ubuntu repositories [1] and a mu
Sometimes the input device shows up at device ID 0, and sometimes device
ID 1, so try both.
BugLink: https://bugs.launchpad.net/bugs/569932
>From 02fb6c46c5910646292fa7e63bde2f3c70e3ded0 Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Wed, 19 May 2010 00:49:03 +0200
Subject: [PATCH
> I just upgraded from fc11 to fc12. With fc12 comes PA 0.9.21.
>
> Pulseaudio fails to detect any of my audio cards, I have two cards.
>
> I'm positive the problem is with UDEV. When I boot the system,
> I'm getting a cryptic message from udev about some {ATTR ... uvent}
> file not found. I'm h
On 2010-05-25 01:26, Lennart Poettering wrote:
> On Wed, 19.05.10 01:00, David Henningsson (launchpad@epost.diwic.se)
> wrote:
>
> Heya,
>
>> Sometimes the input device shows up at device ID 0, and sometimes device
>> ID 1, so try both.
>
> Humpf. W
On 2010-06-15 20:19, José Tomás Tocino García wrote:
> Hi, I'm using the Pulseaudio Simple Api to get the microphone input.
> It works ok, because I can hear exactly what I say on the mic if I
> open an output stream, but if I dump the audio stream to a file and
> then plot it (using gnuplot for in
On 2010-06-15 20:49, José Tomás Tocino García wrote:
> I think that's not the problem, because Gnuplot does not expect a
> exact format, it renders whatever it gets.
>
> The thing is, a usual waveform gets positive and negative values all
> the time. Sometimes that's what some DSP algorithms use t
On 2010-07-23 03:04, Chris wrote:
> What is the best way to start PA for debugging and still have all the
> usual clients running?
pacmd set-log-level 4
...if your log target is set to syslog, just watch it grow :-)
--
David Henningsson, Canonical Ltd.
http://launchpad.net/
sitant to whether it
actually does the right thing in all cases though, and Colin's work in
progress is probably a more complete solution, once it is easily
accessible, but I think it does what you want for now.
Michael, will you update the ticket with the changes you did yesterday?
--
is is ready for everyday usage?
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
fragment-size, or something else), but this patch at least fixes the
immediate problem, causing "crackling" output on (at least) one machine.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
>From 87e8a04f029c18cdb5abebb4f3f3c2100d42107a Mon Sep 17 00:00:00 2001
From: Davi
2010-08-17 20:03, Colin Guthrie skrev:
> 'Twas brillig, and David Henningsson at 17/08/10 19:00 did gyre and gimble:
>> The tsched watermark variable was incorrectly used even for sinks
>> with timer scheduling disabled, causing XRUNs on every rewind. This
>> patch sets
2010-08-17 21:10, Colin Guthrie skrev:
> 'Twas brillig, and David Henningsson at 17/08/10 19:20 did gyre and gimble:
>> According to what you say in that bug, you could reproduce it yourself
>> by setting tsched=0, so I'm eager to hear if this fix fixes your issue
>>
correlate with the way the hardware works.
So your idea is to prevent DMA transfers being modified, but I'm
thinking of the maximum duration between the rewind and the point it
gets filled up again by PA - all of that time we risc getting an XRUN
because there is nothing in the buffer. And so I as
2010-08-19 07:32, Tanu Kaskinen skrev:
> On Wed, 2010-08-18 at 22:55 +0200, David Henningsson wrote:
>>> Yes the safeguard is needed in both cases, timer scheduling or good
>>> ol' audio interrupts. This comes from limitations of the
>>> snd_pcm_rewind() routi
ther David's or Pierre's patch for fixing tsched=0
> mode is merged into s-q tho' :D
Yes please :-) Hope we don't get stuck in never-ending discussions on
that one.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
're about to release 0.9.22
(heard something from Lennart yet?), I suggest we go with my version for
0.9.22 as that one is the least invasive (only touches non-tsched
devices), and keep Pierre's version in master.
--
David Henningsso
(untested)? If you then
would like to turn the define of dma_rewind_margin_bytes into a
parameter, that should be fairly simple.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
>From 5ae6c880e20adfa0f372f2caf293e70253651c5a Mon Sep 17 00:00:00 2001
From: David Henningsson
Dat
2010-09-02 10:41, Colin Guthrie skrev:
> 'Twas brillig, and David Henningsson at 02/09/10 07:29 did gyre and gimble:
>> 2010-09-01 20:06, pl bossart skrev:
>>>>> Probably either one will work, but if we're about to release 0.9.22
>>>>> (heard some
our guess is as good as
mine. Colin, feel free to go ahead with Pierre's suggestion - it's
likely to be good enough.
As for the watermark usage, I admit to not knowing enough of CPU
scheduling and wake-up times to either prove Pierre right or wrong.
--
David Henningsson, Canonical
On 2010-09-04 14:10, Colin Guthrie wrote:
'Twas brillig, and David Henningsson at 03/09/10 09:46 did gyre and gimble:
2010-09-02 16:06, pl bossart skrev:
Agreed: You can pick those two patches, and then we add a third patch to
both branches, which brings back the watermark for tsched de
s it is that cause
the persistent crackling?
The
second is that you could experience underflows if you don't do the
update fast enough. By enabling logs you should be able to find out if
there are real underflows.
Exactly.
--
David Henningsson, Canonical Ltd.
On 2010-09-04 14:10, Colin Guthrie wrote:
'Twas brillig, and David Henningsson at 03/09/10 09:46 did gyre and gimble:
2010-09-02 16:06, pl bossart skrev:
Agreed: You can pick those two patches, and then we add a third patch to
both branches, which brings back the watermark for tsched de
On 2010-09-13 13:03, Colin Guthrie wrote:
'Twas brillig, and David Henningsson at 13/09/10 11:14 did gyre and gimble:
On 2010-09-04 14:10, Colin Guthrie wrote:
I'd be interested as to whether anyone else can repeat this experiment
and get similar results. Do you guys get a broken cho
read the readme file for compilation and install instructions.
When it is installed, run "alsamixertest -r" for a small tutorial and
"alsamixertest -h" for command line options help.
Looking forward to your comments about this new little tool! I think it
should be considered &qu
expected
> values differ from the test value.
So when it did the initial testing signal, it set everything to 0 dB and
the input signal received was -3.04 dB. So when setting Master to -18
dB, it expects the input signal to be -18-3.04 = -21.04 dB.
--
David Henningsson, Canonical Ltd.
h
and ARM, this could occur
when the number of channels were 3.
Signed-off-by: David Henningsson
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
>From a7a099463370c1f49e470597fb7a70a6ac72d685 Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Fri, 8 Oct 2010 18:47:00 +0
embly, that should also benefit from this
change.
I'm not exactly sure where and for what PA_VOLUME_MAX is used, but does
it correspond to 0 dB in any way? Thinking assembly, could it be that we
have some e g fixed-point arithmetic that we must compensate?
--
David Henningsson, Canonical Ltd.
We have got three confirmations that the patch is working in the bug
below, so I believe it can be safely applied to both master and
stable-queue.
On 2010-10-08 19:00, David Henningsson wrote:
I would kindly ask for comments for this patch before applying, just
double-check that I thought the
On 2010-10-13 09:53, Colin Guthrie wrote:
'Twas brillig, and David Henningsson at 13/10/10 07:27 did gyre and gimble:
We have got three confirmations that the patch is working in the bug
below, so I believe it can be safely applied to both master and
stable-queue.
Great!
Would you
rying to add explicit delays just for the volume sync.
Either that, or some kind of volume ramping. Just curious if you
considered that solution as well?
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing l
tracks constant high frequency
noise...more git bisect ahead of me...
It would also be good to know is PULSE_NO_SIMD=1 affects the HF noise,
and if switching to stable-queue affects the HF noise?
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
that auto-mute mode, what profile should he set it to? My natural
assumption would be "analog-output", but I'd like to get it confirmed.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing lis
quot; so the user is left wondering what does
happen in such a case. So, fixing the grammar, the ambiguity, and
re-phrasing it less tersely: "pulseaudio reads things from a file X. If
file X exists, pulseaudio will not read anything from file Y. If and
only if file X does not exist, pulseaudio reads things from file Y."
Substitute X="~/.pulse/default.pa", Y="/etc/pulse/default.pa",
things="configuration directives", and repeat with appropriate
substitutions for each of the man pages.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
On 2010-11-03 10:35, pl bossart wrote:
Hi,
As part of the passthrough/compressed data support in PulseAudio, we
need to disable monitoring. The main use of monitoring is to build
peakmeters to show volume information, and that just doesn't make
sense for compressed raw data
I have also been
Just following up on an idea brought to my attention at the
Ubuntu/Linaro Developer Summit, and I'm not sure whether this is the
original idea or if I refined it a little myself. Anyway.
Since rewinds often come in chunks, and sometimes there are no rewinds
for a very long time, it makes sense
This should be a patch without any controversy. Patch is against
stable-queue.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
>From ff068f1dc07e086ba01b4a4db398d9d81f79f60b Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Fri, 19 Nov 2010 10:41:46 +0100
Subject: [PA
ble master unstable for the time being, with no
release plan?
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
while ago.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
n
the same file as we have today. Any thoughts?
3)
It'd be great if we could present something else than "Microphone 1",
"Microphone 2" and so on when we have more than one mic input. Any idea
of where this name actually comes from, and how we can make it better?
Thank
On 2010-12-01 16:09, Colin Guthrie wrote:
'Twas brillig, and David Henningsson at 01/12/10 13:55 did gyre and gimble:
Hi folks,
The way we control mic input is quite broken. I've tested here with both
IDT codecs and Realtek codecs (the two most common HDA Codecs AFAIK) and
as fa
iming for a just-works
experience (connect=true), which might upset a few people who don't want
it to work that way...I've gone with connect=true as the default to
mimic the behaviour of module-jack-sink and module-jack-source.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/
entioned this in the
past, and thus makes me thing that speaking to jack directly (as you do)
is the only way to make this work correctly)
'Twas brillig, and David Henningsson at 03/12/10 08:29 did gyre and gimble:
* A question is about the default; patch 2 here adds it to default.pa.
Should &q
On 2010-12-04 19:09, Colin Guthrie wrote:
'Twas brillig, and David Henningsson at 02/12/10 10:38 did gyre and gimble:
On 2010-12-01 16:09, Colin Guthrie wrote:
'Twas brillig, and David Henningsson at 01/12/10 13:55 did gyre and
gimble:
Hi folks,
The way we control mic input is qu
ewind.
Hopefully this will improve the situation for at least a few users. The
idea is to let you do initial comment and review, then make a package
and ask some Ubuntu users to test it. After that I'll report back and we
can consider inclusion into stable-queue.
--
David Henningsson, Can
Allow a message in the queue to perform both a seek and a post data.
For clients that do not use PA_SEEK_RELATIVE (e g gstreamer), this
cuts the message count - and sometimes even the rewinds - in half.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
>F
If many small blocks are in queue, handle_seek is being called
for every one of them, sometimes causing a rewind. Delay the
call until all blocks are handled, then call handle_seek only
once.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
>F
If the amount of data in the implementor buffer is very tiny,
i e even less than what we will likely be asked for, don't ask
for a rewind as that would lead to another underrun.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
>From 3c0bf348c3395b3cff0d77fd52a2e1e725c6
thing big
for people not requesting anything in particular?
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
far from tlength (or tlength/2,
tlength/4, or whatever is an optimal number of segments).
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
On 2010-12-09 14:54, David Henningsson wrote:
As some of you have seen on IRC, I spent the some of the last week
fighting rewinds.
An never-ending stream of rewinds seems to be one of the most common
reasons PulseAudio crashes or produces crackling/stuttering output, so
there is a strong
On 2010-12-17 17:39, Marius Bjørnstad wrote:
On 15/12/10 08:41, David Henningsson wrote:
For users running Ubuntu Maverick, there is a ppa for easy testing here:
https://launchpad.net/~diwic/+archive/fighting-rewinds
It also includes two fixes on the gstreamer side.
Hi, I installed the PPA
and why toggling the switch crashes alsamixer...
:-( I was hoping to be able to fix some of that by the time of merge
window for 2.6.38, but I'm not sure I have the time/priority to do so.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
_
one spoken of above, the other
allows for sending larger packets than one gstreamer period-size.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tang
to add
quirks in the driver to correct the range.
http://www.pulseaudio.org/wiki/BadDecibel
http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025213.html
David Henningsson from Canonical also wrote a similar tool:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/7542
I'm not sure
sure you don't mean "s[0] - s[1]"?
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
far down on my TODO list - it'll go faster
if someone else helps out.)
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
nusual hardware.
// David
>From 729223d4c2c8e56e1e4bf838ee9aa11d36f2d90a Mon Sep 17 00:00:00 2001
From: David Henningsson
Date: Mon, 20 Dec 2010 11:13:37 +0100
Subject: [PATCH 1/6] alsa-mixer: Add a few well-known descriptions
Add front mic, rear mic, and docking line
On 2011-01-19 16:59, David Henningsson wrote:
Yeah, I guess it's time. Let me know if you want six emails instead of one.
Over the past few weeks - although there always seems to be something
else you have to fix first - I've been working on fixing the
long-standing issues with the
On 2011-01-25 15:19, Kurt Taylor wrote:
Any other thoughts on buffer size growth? Is there a way to increase it
during playback that I may have missed?
Well, my thought has been to have a huge buffer from the start, just not
fill it up entirely after a rewind.
--
David Henningsson
the time being.
For reference I resend the three patches (one in GStreamer, two in
pulseaudio) that I would like to see applied.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
[1] Well, Wim Taymans has some credit for the helping out with the first
one.
>From f07d6d2f692a
1 - 100 of 147 matches
Mail list logo
