2016-04-15 21:36 GMT+05:00 Arun Raghavan :
>
> On 15-Apr-2016 10:04 PM, "Alexander E. Patrakov" wrote:
>>
>> 2016-04-15 18:01 GMT+05:00 Arun Raghavan :
>> > Hi folks,
>> > Tanu rightly pointed out that we're almost upon the freeze date for
>&g
'll get to them.
>
> If you think something is crucial for this release (critical bug fix,
> regression, etc.), please holler now.
We still don't have
https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-0.2.tar.xz
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inphone preferences.
So - no real (consistency-based) opinion here. If you can find anything
in the HIG that explains that the border should not exist in this case,
please add a link in the commit message, and/or quote directly, and push it.
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features externally to PulseAudio if it is easier. And an
external module SDK would also be a step forward.
I also think I should try porting the existing RAOP2 patch set to make
it an ALSA plugin instead, to set an example of the "externally where it
is easier" rule.
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22.03.2016 11:24, Alexander E. Patrakov пишет:
[readding the list]
22.03.2016 05:56, Ahmed S. Darwish пишет:
On Mon, Mar 21, 2016 at 10:01:14AM +0100, David Henningsson wrote:
Alexander mentioned the SteamOS case, where "they don't link
statically, but have a 'known-go
ames that bundle libpulse are rare there, if they exist there at
all (0 copies of libpulse found in 12 randomly chosen games), which is a
good news.
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ht
he users how to upgrade it.
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Not a Steam user
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t looks like something is sometimes creating an
off-by-one-sample error when copying the data. Which is exactly the job
of the trivial resampler, that's why the question.
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pulseaudio-discu
s.
What do you think?
I am a bit concerned with assigning numbers dynamically for
HDMI/DisplayPort audio pcms. So, given the semi-dynamic proposal that
you made a year ago, I'd say the key should be "sink+port+ELD if dynamic
numbering is not in use, sink+ELD if it is in use".
30.01.2016 18:53, Tanu Kaskinen wrote:
On Sat, 2016-01-30 at 12:07 +0500, Alexander E. Patrakov wrote:
30.01.2016 00:33, David Henningsson wrote:
If you have headphones plugged in and plug in HDMI; you want sound
to stay on headphones.
If you have HDMI plugged in and you plug in headphones
ld be,
and you are in a better position for such statements.
Alternatively, if nobody among developers tests profiles with two output
paths on a regular basis, it may be a good idea to drop them, in which
case the patch gets an ACK from me.
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18.12.2015 13:04, Georg Chini wrote:
On 18.12.2015 08:45, Alexander E. Patrakov wrote:
18.12.2015 11:47, Georg Chini wrote:
On 18.12.2015 06:49, Tanu Kaskinen wrote:
Making pa_sample_rate_valid() accept values above PA_RATE_MAX isn't
very nice, but I can see how it's better than t
27;s the simplest
possible one. But I'd say that it's pointless to look at the code
without understanding the math behind all of that, so please look at
these links, too:
https://en.wikipedia.org/wiki/Sinc_filter
https://ccrma.stanford.edu/~jos/resample/resample.pdf
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et, if I understand the situation
correctly, will still contain similar logic, in order to switch between
the "correct this slowly and carefully" and "correct this as quickly as
possible" modes.
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_
18.11.2015 07:58, a...@accosted.net wrote:
From: Arun Raghavan
This forces the canceller engine to be invoked even if playback is not
currently active. We need to do this for cases where the engine provides
additional processing that is independent of playback, such as noise
suppression and AGC
nfidence that
the latency is within the bounds that we log.
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es not even
look at /etc/pulse/daemon.conf
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rough over
HDMI.
Reported-by: Xamindar
Signed-off-by: Alexander E. Patrakov
---
src/modules/alsa/alsa-sink.c | 8 +++-
1 file changed, 7 insertions(+), 1 deletion(-)
diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c
index c5a72c3..2fdebe0 100644
--- a/src/modules/alsa
.
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22.11.2015 21:57, Georg Chini wrote:
On 22.11.2015 16:05, Alexander E. Patrakov wrote:
22.11.2015 18:44, Georg Chini wrote:
On 22.11.2015 14:26, Alexander E. Patrakov wrote:
22.11.2015 17:21, Georg Chini wrote:
The other big problem is that you cannot determine the number
of cycles you will
want, including instrumentation to
retrieve the latency between the last submitted and the last produced
sample. Then you can force the loopback module to use the ffmpeg
resampler, and then get the information that you need.
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22.11.2015 18:44, Georg Chini wrote:
On 22.11.2015 14:26, Alexander E. Patrakov wrote:
22.11.2015 17:21, Georg Chini wrote:
The other big problem is that you cannot determine the number
of cycles you will need to correct the initial latency error because
this error is unknown before the first
ler with min_cycles = 1,
instead of my controller. That's, without paying attention to artifacts.
I will post this controller (and, as requested, perform measurements
with the trivial resampler) in the near future.
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original
pa_cvolume_merge is clearly wrong if s->reference_volume and
root_source->real_volume have channel maps where a shorter one is not
the beginning of the other.
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---
src/pulsecore/source.c | 18 ++
1 file changed, 14 insertions(+), 4 del
19.11.2015 21:47, Alexander E. Patrakov wrote:
19.11.2015 11:48, Georg Chini wrote:
On 19.11.2015 05:08, Alexander E. Patrakov wrote:
19.11.2015 00:43, Georg Chini wrote:
On 15.11.2015 22:08, Alexander E. Patrakov wrote:
However, I'd argue that this phase metric can be improved withou
19.11.2015 11:48, Georg Chini wrote:
On 19.11.2015 05:08, Alexander E. Patrakov wrote:
19.11.2015 00:43, Georg Chini wrote:
On 15.11.2015 22:08, Alexander E. Patrakov wrote:
The second result (https://imgur.com/a/eVahQ ) is with the old module.
You see that, with it, the rate oscillates
19.11.2015 00:43, Georg Chini wrote:
On 15.11.2015 22:08, Alexander E. Patrakov wrote:
The second result (https://imgur.com/a/eVahQ ) is with the old module.
You see that, with it, the rate oscillates wildly and then snaps to
44100 Hz. However, the final latency is not correct, and it
17.11.2015 12:18, Alexander E. Patrakov пишет:
17.11.2015 12:13, a...@accosted.net wrote:
From: Arun Raghavan
The drain reporting improvements that were added to alsa-sink were only
being applied to directly connected sink inputs. This patch makes the
same logic also recurse down the filter
/21582
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---
src/pulsecore/sink.c | 38 +++---
1 file changed, 31 insertions(+), 7 deletions(-)
diff --git a/src/pulsecore/sink.c b/src/pulsecore/sink.c
index 9ddb527..0b44fc7 100644
--- a/src/pulsecore/sink.c
+++ b/src/pulsecore/sink.c
7, would recommend merging up to PATCH 08/13. I
would recommend waiting two weeks before merging further patches, as the
deadband stuff is likely to get undone and redone if the rate controller
gets replaced. And I am going to experiment further with different rate
controllers on the next w
12.11.2015 01:24, Georg Chini wrote:
On 11.11.2015 20:30, Alexander E. Patrakov wrote:
Note: I did not say that following this method is good for our
purposes. The PID controller recommended in these papers (and used in
Jack) is not optimal in the sense of Ziegler-Nichols method:
http
12.11.2015 00:30, Alexander E. Patrakov wrote:
I.e. here Kp = 1 / adjust_time, that's all.
Correction: Kp = base_rate / adjust_time here. And below, when
discussing oscillations, I should have talked about 2 * base_rate /
adjust_time.
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r-Nichols method prescribes.
Note: I did not say that following this method is good for our purposes.
The PID controller recommended in these papers (and used in Jack) is not
optimal in the sense of Ziegler-Nichols method:
http://kokkinizita.linuxaudio.org/papers/usingdll.pdf
http://kokkinizita
, the update becomes non-atomic. This is a valid objection.
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olumes".
Yes, I understand that this proposal looks at odds with my previous
"ack" to Arun's idea. That "ack" is still in force.
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force this upon all
sandboxed applications (xdg-app).
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software volume, make
"software balance".
As this counterproposal comes without a patch, and especially since the
"counterprpopsed" improvement can be done later, this should not block
the "going to push in one week" statement.
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src/
on:
union {
pa_mem mem;
union {
pa_shm shm;
pa_memfd memfd;
pa_privatemem privatemem;
} per_type;
};
Hopefully these patterns will be OK from your side :-)
Regards,
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esent a natural way to declare data types that
can contain either one set of fields or the other. So, a priori, I have
no objections.
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Originally pointed out by Georg Chini.
Calculating buffer = buffer + (send_counter - recv_counter)
in one branch and buffer = 2 * buffer - (recv_counter - send_counter)
looks very obviously wrong. In other words, before the patch, the
contribution from the previous lines was double-counted.
---
de is practical, and it
does not have to be based on the ideas from BruteFIR :)
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multichannel wav
file with the contributions of each input channel to the left output,
and provide it to module-virtual-surround-sink. The limitation to 64
samples can be easily patched out from module-virtual-surround-sink.
And finally: this is the third request to convolve PulseAudio output
, and no page updates.
Is it just that nothing important has changed since the last update, or ...?
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e. room noise), privately to me for further
debugging. In the worst case, there is always module-jack-source,
jack-rack and various jack-based equalizers.
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h
t; but anything else?
Please run "pavucontrol" and go to the last tab. On that tab, set the
profile for the relevant cards to "Off".
You can do the same with "pactl set-card-profile" from the command line.
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DMI instead of SPDIF when there are two Digital
Outs
According to the manual, there is no such setting in the BIOS.
(original reporter BCCed just in case)
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htt
hing here that would need special treatment by ALSA or
PulseAudio? Or, is this just a misnumbered/misnamed codec?
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upload=true&script=true&cardinfo=
!!
!!ALSA Information Script v 0.4.64
!!
!!Script
03.07.2015 22:26, Tanu Kaskinen wrote:
On Thu, 2015-07-02 at 22:07 +0500, Alexander E. Patrakov wrote:
02.07.2015 20:13, Tanu Kaskinen wrote:
On Tue, 2015-06-30 at 22:29 +0500, Alexander E. Patrakov wrote:
30.06.2015 20:08, Tanu Kaskinen wrote:
Event: monitor gets unplugged
02.07.2015 20:13, Tanu Kaskinen wrote:
On Tue, 2015-06-30 at 22:29 +0500, Alexander E. Patrakov wrote:
30.06.2015 20:08, Tanu Kaskinen wrote:
Event: monitor gets plugged in
--
The first thing that happens should be that PulseAudio gets a wakeup
from the alsa mixer
he big picture of the available multi-monitor
use cases.
Currently, if one has a card with multiple HDMI outputs, one can use
audio from one monitor at a time, using profiles. I.e. there are no
profiles like "HDMI Digital Stereo Output + HDMI 3 Digital Surround
Output" (with the poss
ibe-module" and "pactl list
short sinks" to figure out the correct module arguments.
The caveat is that I recently got a private email from ValdikSS that
tunnels don't really work with mpv (playback stops after a few minutes),
even with pcaps attached. I have not debugg
ounds, PulseAudio does not offer this.
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ot yet ACK the patches.
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forget to report
SNDRV_PCM_INFO_BATCH?
>
> Why pulseaudio rely on the flag if your program can find out the
granulatity ?
AFAIK, there isn't a way to figure out granularity. Having this would
be nice as we could be more intelligent about our tsched behaviour.
There is not only no
eports SNDRV_PCM_INFO_BATCH,
so that PulseAudio does not try to use this mode.
This mean that your sound card won't work with timer scheduling or
dynamic latency, you can only archieve low latency by decrease period size
Why do pulseaudio enable timer scheduling when most sound card
= PA_SAMPLE_S16NE;
lft.ss = &a;
pa_assert_se(lft.pool = pa_mempool_new(false, 0));
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quot;, i);
ret = -1;
break;
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25.05.2015 09:49, Hui Wang wrote:
On 32bits OS, this test case fails. The reason is when rewinding to
the middle of a block, some of float parameters in the saved_state
are stored in the memory from FPU registers, and those parameters will
be used for next time to process data with lfe. Here if F
hange Vala bindings.
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ofile = NULL;
-
pa_device_init_description(c->proplist, c);
pa_device_init_icon(c->proplist, true);
pa_device_init_intended_roles(c->proplist);
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06.05.2015 01:40, David Henningsson wrote:
On 2015-05-05 17:28, Alexander E. Patrakov wrote:
05.05.2015 19:08, Tanu Kaskinen wrote:
On Tue, 2015-05-05 at 17:31 +0500, Alexander E. Patrakov wrote:
27.04.2015 16:34, Tanu Kaskinen wrote:
Here's the third version of the patch set that ai
05.05.2015 19:08, Tanu Kaskinen wrote:
On Tue, 2015-05-05 at 17:31 +0500, Alexander E. Patrakov wrote:
27.04.2015 16:34, Tanu Kaskinen wrote:
Here's the third version of the patch set that aims to fix the Aureon
volume bug[1].
Changes in v3:
- Use pa_parse_volume
P.S. PulseAudio also finds a bogus SPDIF output on the Rotel amplifier.
Maybe we need a whitelist, not a blacklist, here?
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e?
Because this path may be digital.
I wonder whether, with USB devices, a situation is more common that we
know whether the path is analog or digital, or that we don't know. Or
even a corner case where the distinction doesn't really apply. The wild
question:
er.
So, I think that the original plan with a 30% default on all analog
outputs is better.
P.S. I do not own any of hardware mentioned in this email. It was only
demonstrated to me by the salesman.
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u are trading one bug for another. Sorry, I don't have
any easy and constructive suggestions.
Does the card support 32-bit input samples? [that was the solution to
the same kind of problem in the plug+softvol+dmix era that was
implemented on my request in 20
have not looked at the patches yet, but will do so when my newly
ordered Rotel RA-1570 amplifier arrives. It contains a USB DAC inside,
but also has a hardware knob for volume control.
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at should be reported to the kernel bugzilla, where
there are much more people familiar with interrupt setup details of your
board. The key words to use when filing it are "lost interrupts", and,
of course, feel free to add a link to you
unofficial and slightly-buggy version. This is not
recommendied for noobies, so please wait for the official release of
PulseAudio with this functionality.
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ces, and we are waiting for them to be fixed.
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n the pa_sink_update_rate() function, in src/pulsecore/sink.c.
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27.03.2015 02:09, Georg Chini wrote:
Also the patch
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/22913
is included in the loopback series. If Alexander does not object, maybe
you can remove that as well.
No objections to removing the duplicate.
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and whether this is important.
No leaks this time, no artifacts either, so I think that the series can
be merged.
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oftware volume) ?
Some rewind will happen. It will not generally be aligned to the
memblock boundary. I.e. a thing will happen that this patch tests.
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http
24.03.2015 23:47, Alexander E. Patrakov wrote:
24.03.2015 14:29, David Henningsson wrote:
Changes since v2:
- Hui has written a test case to test the new lfe filter's rewind
functionality
- Several bugs found and fixed in the lfe filter rewinding; the
trickiest being
the fact th
me, though. Oh, and it doesn't do any matrix decoding. (Which
* probably wouldn't make any sense anyway.)
...which is now too pessimistic.
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r it is in samples before or after resampling,
and maybe a comment that there is a doubt whether this signature is
actually suitable in the general case.
What is certain is that this works as a useful temporary solution in the
no-resampling case, where only the LFE filter has to be rewound.
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ether. But, this is just a nitpick.
+void lr4_process_float32(struct lr4 *lr4, int samples, int channels, float
*src, float *dest)
Yes, it was a good idea to remove in-place processing.
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pulsea
y, as a fallback for
outputs that currently have no connected monitors. So I am also fine
with the rename.
E.g. my use case involves the "Digital Stereo (HDMI2) Output" profile.
When my LG TV is on, then we can of course show "LG TV" as the monitor
name. But when it is off,
?
No. PulseAudio only deals with sampled sound, not MIDI.
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then you can pass a
"--disable-lynx" argument to the configure script.
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13.03.2015 18:54, Felipe Sateler wrote:
On 13 March 2015 at 10:46, Alexander E. Patrakov wrote:
Some lynx versions produce links of the form file:///..., others produce
file://localhost/..., so catch both forms.
Reported-by: Peter Mattern
Signed-off-by: Alexander E. Patrakov
---
doc
Some lynx versions produce links of the form file:///..., others produce
file://localhost/..., so catch both forms.
Reported-by: Peter Mattern
Signed-off-by: Alexander E. Patrakov
---
doc/Makefile.am | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/doc/Makefile.am b/doc
that the local lynx configuration in
/etc/lynxrc can affect the output.
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er. For me, it is also acceptable.
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that was used during evaluation? I.e. to this paper:
http://www.mp3-tech.org/programmer/docs/6_Heusdens.pdf
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11.03.2015 21:58, Alexander E. Patrakov wrote:
11.03.2015 21:18, Alexander E. Patrakov wrote:
11.03.2015 00:42, Alexander E. Patrakov wrote:
So it may be a good idea to retest the new module without
module-stream-restore. I will do that tomorrow.
Done. Result: it moves the streams from mpv
11.03.2015 21:18, Alexander E. Patrakov wrote:
11.03.2015 00:42, Alexander E. Patrakov wrote:
So it may be a good idea to retest the new module without
module-stream-restore. I will do that tomorrow.
Done. Result: it moves the streams from mpv (including active ones)
between headphones and
11.03.2015 00:42, Alexander E. Patrakov wrote:
So it may be a good idea to retest the new module without
module-stream-restore. I will do that tomorrow.
Done. Result: it moves the streams from mpv (including active ones)
between headphones and HDMI, approximately as expected. So I suggest
sound card, provides similar quality.
After comparing them, I decided (for convenience reasons) to continue
using the webcam microphone.
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10.03.2015 23:28, Alexander E. Patrakov wrote:
I have read the rest of the patch, and could not find anything
obviously wrong. But I have not tested it, either, so it is not an
"ack" yet. I intend to do some testing in the next hour and also
tomorrow.
OK, first log of meaningl
But I have not tested it, either, so it is not an "ack" yet. I
intend to do some testing in the next hour and also tomorrow.
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09.03.2015 15:11, Thomas Martitz wrote:
Am 08.03.2015 um 12:42 schrieb Alexander E. Patrakov:
But, why are you using the passthrough mode at all? HDMI, unlike
SPDIF, has enough bandwidth to pass the 7.1 PCM stream. For SPDIF,
passthrough was a necessity due to this bandwidth limitation. For
eg -f alsa -acodec ac3 -i spdif:1 -f pulse default
(not tested, because I don't have anything with spdif-input capability)
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rough over
HDMI.
Reported-by: Xamindar
Signed-off-by: Alexander E. Patrakov
---
src/modules/alsa/alsa-sink.c | 8 +++-
1 file changed, 7 insertions(+), 1 deletion(-)
diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c
index fb3c6cc..e7274b6 100644
--- a/src/modules/alsa
imes think that the proper fix is to disallow
passthrough completely for multichannel HDMI profiles, i.e. to
effectively hide the checkboxes that you have demonstrated on the
screenshot.
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Alexander E. Patrakov
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to the echo-cancelled source using pacucontrol.
There is no need to file a bug, because it already exists:
https://bugs.freedesktop.org/show_bug.cgi?id=83557
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Alexander E. Patrakov
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pulseaudio-discus
in default.conf.
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Alexander E. Patrakov
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?
My guess so far is that it never worked, and you broke it by enabling
the 7.1 profile that was not available before (i.e. I guess that you
were using a stereo profile before).
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Alexander E. Patrakov
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pul
26.02.2015 09:29, Arun Raghavan wrote:
On 25 February 2015 at 11:20, Alexander E. Patrakov wrote:
25.02.2015 07:25, Arun Raghavan wrote:
The falce branch would only be called when recv_counter >
send_counter, so you'd never actually have the zero case. AIUI, this
all just accounting
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