e not tested. I will write instructions on the next week, so that
you can test yourself every situation that you want to.
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[sent only to Tanu by mistake, resending to the list]
07.09.2014 15:48, Tanu Kaskinen wrote:
On Sun, 2014-08-24 at 13:35 +0600, Alexander E. Patrakov wrote:
04.08.2014 19:29, I wrote:
Anyway, I think that the task of objectively testing the resampler
speed and quality also needs to be done
03.09.2014 18:28, Alexander E. Patrakov wrote:
03.09.2014 16:47, David Henningsson wrote:
2. Should enable-lfe-remixing be a global setting? Or should it be
configured per port? Per profile?
Definitely not global, not sure about the other options.
And now sure.
Since we allow the users to
.net/repo/valib/file/49e0393398e6/valib/iir
We can discuss this further or hack a prototype in Dusseldorf.
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25.08.2014 00:53, Alexander E. Patrakov wrote:
Unfortunately, there is a bug on win81 plots, because Windows Media
Player by default attenuates the file by 6 dB, and my scripts compensate
for that, but also amplify the quantization noise. I am too lazy to fix
this today. Please shift the whole
02.09.2014 14:16, David Henningsson wrote:
On 2014-08-24 20:53, Alexander E. Patrakov wrote:
I have finished the first stage of my work on resampler quality
evaluation.
The scripts are here: https://gitorious.org/psy-eval/psy-eval/
The results are here: https://imgur.com/a/jtIEj
Note: they
X220T, both for
"docked" and "undocked" cases? (/proc/asound/card0/codec\#0 or something
similar)
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e-jack-source.c
@@ -51,7 +51,7 @@
PA_MODULE_AUTHOR("Lennart Poettering");
PA_MODULE_DESCRIPTION("JACK Source");
PA_MODULE_VERSION(PACKAGE_VERSION);
-PA_MODULE_LOAD_ONCE(true);
+PA_MODULE_LOAD_ONCE(false);
PA_MODULE_USAGE(
"source_name= "
-specific patch be welcome that attempts to log which
process keeps the device open? If so - should I walk /proc myself, or
defer to fuser?
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http
ightly audible
distortions on pure tones above 2 kHz, try speex-fixed-1.
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am to use PulseAudio.
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oduces more
latency than traditional resamplers - 20ms for the HQ variant vs less
than 1ms for a traditional resampler. Should PulseAudio be aware of it?
How can we make it aware?
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25.08.2014 01:02, Weedy wrote:
On 24 Aug 2014 14:53, "Alexander E. Patrakov" mailto:patra...@gmail.com>> wrote:
> P.S. The following resamplers are not on the plots:
>
> src-zero-order-hold: exactly the same as trivial.
> speex-float-4: very very similar t
into typical music and speech for my talk at the
audio mini conference.
P.S. The following resamplers are not on the plots:
src-zero-order-hold: exactly the same as trivial.
speex-float-4: very very similar to speex-float-3. Not perfect.
speex-float-2: worse than speex-float-1.
Please ignore th
+
5 files changed, 162 insertions(+), 1 deletion(-)
I'd rather not accept this resampler. It is worse than speex-float-1,
and, for 44100 -> 48000 Hz resampling, produces audible (according to
the model) distortions for every full-scale sine wave above 3 kHz.
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implementation error. This resampler behaves incorrectly for the 44100
-> 48000 Hz when given a sine wave with frequency higher than 11025 Hz.
A proper resampler would output a sine wave with the same frequency as
input. This one creates a sine wave of a different frequency.
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tolerable size for the
list). I think that it makes sense to expose other quality settings
provided in soxr.h.
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good enough". The (useless) answer is almost always the same:
"no, here is a sine wave frequency that it either attenuates audibly or
distorts audibly", even though nobody listens to e.g 18 kHz sine waves.
I will test the two new resamplers among the others today.
--
A
tabase);
> +}
>
> if (u->database)
> pa_database_close(u->database);
ACK.
This saves the database on module unload if there is already an
intention to save it "later", which is the correct criterion.
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_
patch is indeed minimal and
sufficient. So this patch only gets a "possibly-incomplete ACK".
Maybe it is a good idea to write a test that verifies that the
analog-output path is indeed a subset of all mixer paths that should
kill it?
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_
In commit 72103e1e, there are things that I want to clean up.
No functional changes.
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There was no code that included files from other directories using
the #include "..." style before.
Signed-off-by: Alexander E. Patrakov
---
src/pulsecore/resampler/ffmpeg.c| 2 +-
src/pulsecore/resampler/libsamplerate.c | 2 +-
src/pulsecore/resampler/peaks.c |
IMHO code that calls into speex belongs in speex.c, not in resampler.c.
Signed-off-by: Alexander E. Patrakov
---
src/pulsecore/resampler.c | 43 -
src/pulsecore/resampler.h | 3 +++
src/pulsecore/resampler/speex.c | 31
04.08.2014 18:40, Peter Meerwald wrote:
+#include
+
+#include "pulsecore/resampler.h"
Hm. Inconsistent style of <> vs "" includes. And in fact, this adds the
first instance of including a file from a different directory via the ""
style of quoting the
quality evaluation, but
don't want them to be upstreamed yet, until we have tools to judge them.
Hopefully this clears the "what's next" question.
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17.08.2014 10:50, Tanu Kaskinen wrote:
On Fri, 2014-08-08 at 16:33 +0600, Alexander E. Patrakov wrote:
Hello.
[Note: the whole e-mail is purely theoretical. I still don't have any
surround-sound system in my room. A salesman told me that I should buy
5.1 speakers, not 7.1, because the
http://www.razersupport.com/gaming-audio/razer-tiamat-71/ - in
particular, the answer to the "What is the difference between 5.1 and
7.1 surround sound?" question.
How should PulseAudio deal with this nomenclature mess?
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things that I work on occasionally in fifo
order).
May I look at the other items in this list?
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= no
Try uncommenting and flipping it.
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06.08.2014 17:39, I wrote:
06.08.2014 17:11, Tanu Kaskinen wrote:
On Wed, 2014-08-06 at 16:35 +0600, Alexander E. Patrakov wrote:
Tanu proposed:
3) Add a second volume control to streams, one which represents the
stream's own volume only, i.e. never flat volume. Applications that
wa
06.08.2014 17:11, Tanu Kaskinen wrote:
On Wed, 2014-08-06 at 16:35 +0600, Alexander E. Patrakov wrote:
Tanu proposed:
3) Add a second volume control to streams, one which represents the
stream's own volume only, i.e. never flat volume. Applications that want
to avoid flat volume can use
be used for volume groups, one can
be used for ducking when it is in effect, and one for client-specific needs.
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easible, then I would rank it above (1), but below (2) and
(3). And of course, if (3) is implemented, then (4) becomes unneeded.
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06.08.2014 12:15, Arun Raghavan wrote:
On 6 August 2014 10:28, Alexander E. Patrakov wrote:
06.08.2014 10:20, Arun Raghavan wrote:
On 6 August 2014 09:41, Alexander E. Patrakov wrote:
06.08.2014 09:49, Arun Raghavan wrote:
[...]
Not entirely true - with the patch I've posted, fo
it on login. See
/etc/xdg/autostart/pulseaudio.desktop , so even the "pulseaudio --start"
command is not actually needed to be issued.
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06.08.2014 10:20, Arun Raghavan wrote:
On 6 August 2014 09:41, Alexander E. Patrakov wrote:
06.08.2014 09:49, Arun Raghavan wrote:
You didn't address my actual concern here - making such a change would
either require the user to use their desktop volume control to change
browser volum
n the web side.
Well, no comments here.
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06.08.2014 06:31, Arun Raghavan wrote:
On 6 August 2014 01:12, Alexander E. Patrakov wrote:
I think that the real problem that needs to be solved is not only that flat
volumes are inapplicable to the web, but also that PulseAudio doesn't
provide any API for intra-application mixing. Pleas
t inside the browser should be able to set,
invisibly for the user, tab-relative (and thus non-flat) volumes for
each substream.
Here is a test page for you:
http://www.kibagames.com/Game/Kiba_Kumba_Jungle_Chaos
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exist (as a
stub) even without speex. Maybe it indeed should not be moved, maybe it
should be moved and a stub has to be created then in resampler.h.
Other than that, the patch looks good.
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d
and quality also needs to be done, in order to provide such
justifications. Please see
http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-February/019968.html
for the formulation.
Feel free to come to the Plumbers conference and discuss it there if we
don't reach any concl
/resampler/speex.c
create mode 100644 src/pulsecore/resampler/trivial.c
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ut frequencies or enumerations.
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Remove extra-hdmi.conf, as the performance reasons behind it are invalid
Add 7.1 profiles
Add extra HDMI devices, for a total of 8
Add DTS-encoded profiles (they need dcaenc from git)
Signed-off-by: Alexander E. Patrakov
---
v2: actually ship the extra HDMI paths using Makefile.am
src
2014-08-01 13:42 GMT+06:00 David Henningsson :
> I went to test and then commit your patch, but I noticed that our test case
> alsa-mixer-path-test started failing. Did you forget to ship the new
> hdmi-output-*.conf files?
Thanks for catching my mistake.
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2014-07-31 14:46 GMT+06:00 David Henningsson :
>
>
> On 2014-07-30 20:36, Alexander E. Patrakov wrote:
>>
>> Add DTS-encoded profiles (they need dcaenc from git).
>>
>> The use cases for DTS over HDMI are:
>>
>> 1. Radeon driver on old kerne
HDMI
audio.
2. A TV that acts as a converter between an HDMI-connected computer and
an SPDIF-connected receiver, with no other way to connect the components.
Signed-off-by: Alexander E. Patrakov
---
This supersedes the old patch with the title "Add HDMI Surround 7.1 profiles"
2014-07-30 16:48 GMT+06:00 David Henningsson :
>
>
> On 2014-07-30 12:34, Alexander E. Patrakov wrote:
>>
>> 2014-07-30 16:31 GMT+06:00 David Henningsson
>> :
>>>
>>>
>>>
>>> On 2014-07-30 11:59, Alexander E. Patrakov wrote:
>
2014-07-30 16:31 GMT+06:00 David Henningsson :
>
>
> On 2014-07-30 11:59, Alexander E. Patrakov wrote:
>>
>> 2014-07-27 17:24 GMT+06:00 Tanu Kaskinen :
>>>
>>> On Tue, 2014-07-22 at 15:11 +0200, David Henningsson wrote:
>>>>
>>>>
nel
>> and the 3.4 LTS kernel. We could consider removing that workaround at
>> some time...
>
> Oh, ok, I didn't know they're unnecessary on newer kernels. Yes, we
> should drop them at some point, but I guess now is not the time.
Due to these complications, I
c" to "Multichannel"
* Removed unnecessary !!
I have looked at the patches, and have no questions and no objections.
However, I didn't test them either.
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2014-07-27 23:48 GMT+06:00 David Henningsson :
>
>
> On 2014-07-26 06:47, Alexander E. Patrakov wrote:
>>
>> 25.07.2014 19:31, David Henningsson wrote:
>>>
>>> In case all other profiles fail, try this fallback mapping as well.
>>> It allows the d
valid when applied to PulseAudio.
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25.07.2014 19:31, David Henningsson wrote:
In case all other profiles fail, try this fallback mapping as well.
It allows the device to specify the channel count, so it can be used
for devices that only supports being opened in multichannel mode.
Signed-off-by: David Henningsson
---
src/module
phenomenon (might be a buggy patch due to the
noise here).
The new timing patch is attached, as well as the gzipped
very-very-verbose log from that PC.
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>From 3f0f6c2505be1617875a1fac811bece20d47c8cf Mon Sep 17 00:00:00 2001
From: "Alexander E. Patrakov"
bash script (thus
having access to common programming concepts such as variables and
loops), but you may have a better idea.
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15.07.2014 22:28, Alexander E. Patrakov wrote:
13.07.2014 18:12, Tanu Kaskinen wrote:
Someone (not me, at least any time soon) could write a simple
patch that measures and logs (at error level - measurements shouldn't be
done at debug log level) the time that the probing takes. Then test
ice 1 took
11445 usec in total
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sec
E: [pulseaudio] alsa-mixer.c: Checking for profile
output:hdmi-surround-extra3 took 192 usec
E: [pulseaudio] alsa-mixer.c: Checking for all profiles on device 1 took
10929 usec in total
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pulse
igure that out is to test
PulseAudio from git. It's non-destructive, you don't have to install it,
it runs fine from the build directory as long as --prefix, --sysconfdir
and --localstatedir are the same as in the distribution-provided versi
Pi.
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>From 5d20303b677bad7dfcc12d548edc7748d821c0d7 Mon Sep 17 00:00:00 2001
From: "Alexander E. Patrakov"
Date: Tue, 15 Jul 2014 22:19:13 +0600
Subject: [PATCH] alsa: Report profile-probing timings
---
src/modules/alsa/alsa-mixer.c | 13 +
1 fil
o eliminate the unknown, which version of skype do you use?
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still be
correct unless the compiler documentation says otherwise).
http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-April/020533.html
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re/sink.c,
function pa_sink_update_rate().
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t that X server does just a memcpy in the worst case (and exactly
nothing in the best case), and PulseAudio does some arithmetics, and
you'll understand why the CPU usage is of the similar order.
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://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=fe346cadedd2a9b48fc36886c4a8d48f50302f07
Could you please try and see whether PulseAudio from git behaves better?
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se we now know that there are two problems
(one related to the peak meter of doom and one unrelated), to be
profiled separately. Profiling instructions will be sent later today,
when I return from work.
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; - there is pavucontrol, which
causes PulseAudio to enable the peak meter. And with that, it's
perfectly reasonable for PulseAudio to eat 6% of your CPU.
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and the information collected by
alsa-info.sh script:
http://www.alsa-project.org/alsa-info.sh
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13.07.2014 20:07, Tanu Kaskinen wrote:
On Sun, 2014-07-13 at 19:49 +0600, Alexander E. Patrakov wrote:
Signed-off-by: Alexander E. Patrakov
---
src/modules/alsa/mixer/profile-sets/default.conf | 16
1 file changed, 16 insertions(+)
Tanu Kaskinen wrote:
Somewhat related
Signed-off-by: Alexander E. Patrakov
---
src/modules/alsa/mixer/profile-sets/default.conf | 16
1 file changed, 16 insertions(+)
Tanu Kaskinen wrote:
> Somewhat related, I also wonder why the surround mappings (both 5.1 and
> 7.1) are only in extra-hdmi.conf. My underst
ency, taskbar button hints, overview and other unexpected places
where even fully covered windows can render through.
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Signed-off-by: Alexander E. Patrakov
---
Untested.
P.S. should I also upstream profiles that allow sending software-DTS-encoded
stream to HDMI? The git version of dcaenc supports this, and that's how
I test it. Here is what my README file tells the users about this
(slightly rephrased):
&q
ich will break a lot of
GNOME stuff).
The patches are here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/19625
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ontrolled with the
enable-lfe-remixing setting in /etc/pulse/daemon.conf.
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PulseAudio: killall
pulseaudio). If that succeeds, I will post the formal patch to the
alsa-devel list.
As for the second SPDIF, sorry, I think there is some misunderstanding
here. According to the reviews, the card has one spdif input and one
spdif output, both of which are already supported.
30.06.2014 09:29, Alexander E. Patrakov wrote:
30.06.2014 06:41, Matt Zagrabelny wrote:
% pulseaudio --version
pulseaudio 5.0
The card has:
2 digitial output ports (S/PDIF)
8 analog output ports
8 analog input ports
1 MIDI input port
1 MIDI output port
Yet the list of profiles [1] doesn&#
from here:
http://www.alsa-project.org/alsa-info.sh
It will gather some information from your card and paste it to a pastebin.
Once I see all of that, I will ask more questions about the card.
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Audio that we cannot fix because we don't know
what's wrong. Please help us!
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24.06.2014 14:06, David Henningsson wrote:
On 2014-06-17 11:09, David Henningsson wrote:
On 2014-06-01 20:55, Alexander E. Patrakov wrote:
30.05.2014 17:59, David Henningsson wrote:
+else if (cmh->cmsg_type == SCM_RIGHTS) {
+int nfd = (cmh->cmsg_len - CM
kernel doesn't tell us again and again about it.
Signed-off-by: Alexander E. Patrakov
---
src/modules/rtp/rtp.c | 25 +++--
1 file changed, 23 insertions(+), 2 deletions(-)
diff --git a/src/modules/rtp/rtp.c b/src/modules/rtp/rtp.c
index 570737e..7b75e0e 100644
--- a/src/mo
05.06.2014 17:13, Tanu Kaskinen пишет:
On Sat, 2014-05-31 at 23:48 +0600, Alexander E. Patrakov wrote:
Signed-off-by: Alexander E. Patrakov
---
src/modules/rtp/rtp.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/src/modules/rtp/rtp.c b/src/modules/rtp/rtp.c
index
io-5.0/src/pulsecore/memblock.c:596, function
pa_memblock_unref(). Aborting.
And now this is CVE-2014-3970.
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This fixes assertion failures that manifest themselves with cards that
support only weird rates such as 37286Hz. Tested with snd-pcsp.
Signed-off-by: Alexander E. Patrakov
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=48109
---
src/daemon/daemon-conf.c | 6 ++
src/modules/dbus
only one source combining these two to GStreamer
pipeline through a pulsesrc.
Do you have any idea ?
Load module-null-sink. Then load module-loopback twice, looping both
sources to this sink. Then record from the monitor source of the null sink.
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le descriptors!");
+continue;
+}
+memcpy(ancil->fds, CMSG_DATA(cmh), nfd * sizeof(int));
+ancil->nfd = nfd;
}
Don't we need to close these injected file descriptors if we don't lik
kets, as nobody is going to run the real
esd on the same local machine as PulseAudio.
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Signed-off-by: Alexander E. Patrakov
---
src/modules/rtp/rtp.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/src/modules/rtp/rtp.c b/src/modules/rtp/rtp.c
index 570737e..8451386 100644
--- a/src/modules/rtp/rtp.c
+++ b/src/modules/rtp/rtp.c
@@ -183,7 +183,7 @@ int
002-avahi-daemon-remote-denial-of-service/
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30.05.2014 18:01, Tanu Kaskinen wrote:
On Wed, 2014-05-28 at 14:30 +0600, Alexander E. Patrakov wrote:
28.05.2014 12:08, Jay Sorg wrote:
I don't want a TCP or UDP connection for each session or a confusing
sink or source name.
module-esound-sink works with unix-domain sockets, too
ood idea,
independently from any xrdp-related stuff, to implement a real RTP sink
and source, not needing the null-sink trickery.
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es.
Just as a side note (not as a proposal): a different solution is
employed by ALSA. They also don't expose the full possibly-unstable
plugin API, but have an "external plugin SDK" with a stable API that
allows to build limited-functionality plugins that do external I/O, and
28.05.2014 10:53, Alexander E. Patrakov пишет:
28.05.2014 02:56, Jay Sorg wrote:
Hi Alexander,
One big question up-front, sorry for not asking it earlier. Why do
you need
to invent a custom protocol to communicate with xrdp, instead of
implementing something more standard inside XRDP
rs on the wire, and can wait until June 6, I can help
you add ESD protocol support in XRDP. This way, not a single line of
PulseAudio source code would need to be changed.
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PLAY"));
+
+if (!(u->thread = pa_thread_new("xrdp-sink", thread_func, u))) {
+pa_log("Failed to create thread.");
+goto fail;
+}
+
+pa_sink_put(u->sink);
+
+pa_modargs_free(ma);
+
+return 0;
+
+fail:
+
making it a module?
--
Alexander E. Patrakov
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, please run latencytop and see the cause of the "Scheduling
delay" message there. Also please verify your rtkit setup.
As for "ALSA woke us up", it is clearly a kernel bug. Please fix your
driver.
If you need to separate the two bugs (scheduling latency vs premature
ke sure it uses
the echo-cancelled sink & source.
If the problem does not go away, let's hope someone else has a better
idea what's going on.
--
Alexander E. Patrakov
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pulseaudio-
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-fixed-* in that case.
Signed-off-by: Alexander E. Patrakov
Reported-by: Fahad Arslan
Cc: Damir Jelić
Cc: Peter Meerwald
FIXED_POINT detection is based on code by Peter Meerwald.
---
src/pulsecore/resampler.c | 57 ++-
1 file changed, 56 insertions
401 - 500 of 668 matches
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