to always use PCM).
And they are staffed better than us.
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default-fragment-size-msec = 5
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Signed-off-by: Alexander E. Patrakov patra...@gmail.com
---
src/modules/module-loopback.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/src/modules/module-loopback.c b/src/modules/module-loopback.c
index 7e2b92a..3b0d68d 100644
--- a/src/modules/module-loopback.c
+++ b/src
11.02.2015 21:50, Alexander E. Patrakov wrote:
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
Forgot to say that this has been reported by Georg Chini in
http://lists.freedesktop.org/archives/pulseaudio-discuss/2015-February/023120.html:
Something else seems to
call one
cannot change the profile of a port that is not
currently active. Two versions of the enitiy-relationship diagram (with
and without the module) must thus be maintained in one's mind.
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10.02.2015 01:57, David Henningsson wrote:
Thanks for the review!
On 2015-02-01 15:37, Alexander E. Patrakov wrote:
If adding proper comments to all the math is a too-hard requirement,
then, in v2, I'd suggest to (temporarily?) squash patches 1 and 2 for
easier review, and to remove all unused
08.02.2015 13:21, Alexander E. Patrakov wrote:
08.02.2015 02:14, Georg Chini wrote:
Sorry, but I do not think the smoother is the problem here. I do get
quite reliable latency results.
The problem is really (if there is a problem at all) the execution time
of the code
. For USB devices, the
granularity of position reports is 6 ms (for large period sizes), but
for others, it may be up to one period size.
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without this term, maybe it is a good idea to delete it?
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From 3fc2d409dcee3d1e9a7ad5df5ee06d3e1457f7e0 Mon Sep 17 00:00:00 2001
From: Alexander E. Patrakov patra...@gmail.com
Date: Sun, 8 Feb 2015 23:14:12 +0500
Subject: [PATCH] Use a less jittery sample rate in calculations
08.02.2015 22:43, Georg Chini wrote:
On 08.02.2015 16:52, Alexander E. Patrakov wrote:
OK, then I think there was some misunderstanding on my side. Could you
please post some log lines with two USB devices to completely clear
this up? I want logs without the stop criterion (which is properly
0.75% due to min_cycles */
+new_rate = base_rate * (1.0 + latency_difference / min_cycles /
u-adjust_time) + 0.5;
What's the aim here with min_cycles? Why not just clamp new_rate
post-factum to 0.75% vicinity of base_rate, as this is done in the 2‰ case?
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08.02.2015 18:50, Georg Chini wrote:
On 08.02.2015 14:03, Alexander E. Patrakov wrote:
08.02.2015 17:35, Georg Chini wrote:
I think there is some misunderstanding. Let me repeat in a different
way.
The smoother works perfectly (both for timer-based scheduling and for
the needs of your
09.02.2015 00:35, Georg Chini пишет:
On 08.02.2015 20:30, Georg Chini wrote:
On 08.02.2015 19:54, Alexander E. Patrakov wrote:
01.02.2015 03:43, Georg Chini wrote:
+/* Minimum number of adjust times + 1 needed to adjust at 0.75%
deviation from base rate */
+min_cycles = (double)abs
currently have on the topic,
and I have not reviewed all the fine details.
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headphones.
The real question is where and how should my preference to use the 5.1
output be stored in PulseAudio without it being lost when I plug my
headphones.
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, because
I am not sure I understand them correctly.
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values follow each other
you will have one adjust time that is around 60 ms too long and another
one which is 60 ms too
short. Maybe this also contributes significantly to the (in)stability of
regulation.
Thanks for this observation. I will look into it, too.
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cannot retest this at home without additional jack-retasking
(which could make the result untrustworthy), because my home PC does not
have any audio sockets at the front panel.
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06.02.2015 16:41, David Henningsson wrote:
On 2015-02-04 18:45, Alexander E. Patrakov wrote:
Sorry for a possibly-stupid question, but...
Which part of PulseAudio is supposed to disable the effect if the user
plugs headphones in? Or is it yet to be written?
Hrm, that is actually a good
First of all, thanks for a quick and detailed answer.
06.02.2015 02:02, Georg Chini wrote:
On 05.02.2015 16:59, Alexander E. Patrakov wrote:
01.02.2015 03:43, Georg Chini wrote:
This is the final version of my patch for module-loopback. It is on
top of the
patch I sent about an hour ago
in disguise?
How was the lowpass filter tuned?
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the effect if the user
plugs headphones in? Or is it yet to be written?
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01.02.2015 01:29, Felipe Sateler wrote:
On Sat, Jan 31, 2015 at 4:56 PM, David Henningsson
david.hennings...@canonical.com wrote:
On 2015-01-31 08:19, Felipe Sateler wrote:
On Sat, Jan 31, 2015 at 4:52 AM, Alexander E. Patrakov
patra...@gmail.com wrote:
31.01.2015 04:49, David Henningsson
]
+ *
+ * Each lp or hp is an LR4 filter, which consists of two second-order
+ * lowpass or highpass butterworth filters.
+ */
Do we actually need this three-band filter on any known hardware?
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will review the actual implementation later.
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for
appending, not for rewinding samples.
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for the previous version, they are not.
https://packages.debian.org/experimental/amd64/pulseaudio-module-raop/filelist
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27.01.2015 18:23, David Henningsson wrote:
On 2015-01-27 08:10, Tanu Kaskinen wrote:
On Mon, 2015-01-26 at 05:37 -0500, David Henningsson wrote:
On 2015-01-25 13:52, Alexander E. Patrakov wrote:
Hello.
I have noticed some ports in my pactl list cards output, that I
think should
26.01.2015 15:37, David Henningsson wrote:
On 2015-01-25 13:52, Alexander E. Patrakov wrote:
Hello.
I have noticed some ports in my pactl list cards output, that I
think should not be there.
1. iec958-stereo-input on my webcam
This one is very likely an alsa-lib thing. See
https
of that is with today's git master, and linux-3.19.0-rc4
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Card #0
Name: alsa_card.pci-_00_03.0
Driver: module-alsa-card.c
Owner Module: 5
Properties:
alsa.card = 0
alsa.card_name = HDA Intel HDMI
, but today or next week would be fine.
Well, I also have some limited-scope tasks. My nick is patrakov, and I
usually visit the #pulseaudio channel around 16:00-18:00 GMT.
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, then module-device-manager can also be the cause
of this problem. It should not be unloaded, but you can lower the
priority of the receiver in the KDE control center, in the Multimedia
applet.
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19.01.2015 10:24, Arun Raghavan wrote:
On 19 January 2015 at 10:31, Alexander E. Patrakov patra...@gmail.com wrote:
19.01.2015 05:33, Hajime Fujita wrote:
[...]
Also, here's how I launched valgrind. If more detailed options are
necessary please let me know.
$ valgrind --leak-check=yes
how to do that.
Of course, we can combine our flags, and replace --leak-check=yes with
--leak-check=full.
Detailed output from valgrind can be found here:
https://github.com/hfujita/pulseaudio-raop2/issues/35
That's only the tail of it.
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);
pa_mutex_unlock(p-mutex);
return e;
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08.01.2015 17:44, Andrey Semashev wrote:
On Thursday 08 January 2015 11:29:42 Alexander E. Patrakov wrote:
08.01.2015 01:52, Andrey Semashev wrote:
Also, with PulseAudio forced to 44.1 kHz, FooBar2000 v1.2 (which uses
DirectSound and thus, by default, resamples everything to 48 kHz) just
? Are you
running Windows in a VM here? Which one?
That's in wine. FooBar2000 v1.2 (and not any later version) is good for
testing DirectSound-related code paths.
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07.01.2015 15:14, Alexander E. Patrakov wrote:
I will retest with qemu (via its alsa backend) later today.
Done, both for pa and alsa backends. I must say that soxr-based
resamplers break a configuration that worked fine (except the initial
period of several seconds) with speex-float-5
06.01.2015 19:17, Andrey Semashev wrote:
On Wednesday 19 November 2014 00:58:04 Alexander E. Patrakov wrote:
19.11.2014 00:42, Andrey Semashev wrote:
On Tuesday 18 November 2014 21:32:53 Alexander E. Patrakov wrote:
18.11.2014 19:14, David Henningsson wrote:
On 2014-11-18 14:46, Alexander E
== xyes],
== is a bashism. Use =. Otherwise, the patch is good.
[PKG_CHECK_MODULES(LIBSAMPLERATE, [ samplerate = 0.1.0 ],
HAVE_LIBSAMPLERATE=1, HAVE_LIBSAMPLERATE=0)],
HAVE_LIBSAMPLERATE=0)
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be a good idea to note that 7.1 audio is now supported on HDMI outputs.
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pa_rtpoll_set_timer_disabled(u-rtpoll);
i.e. sleep_usec == 0 means disable the timer. In this case, pulseaudio
will wait on the file descriptors provided by snd_pcm_poll_descriptors().
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While at it, also remove SOCKET_SERVER_GENERIC, because it is always
being overwritten with a specific socket type.
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
---
src/pulsecore/socket-server.c | 5 +
src/pulsecore/socket-server.h | 1 -
2 files changed, 1 insertion(+), 5
(ss = pa_socket_server_new(m, fd));
+
+ss-type = SOCKET_SERVER_IPV6;
+ss-tcpwrap_service = pa_xstrdup(tcpwrap_service);
return ss;
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://thread.gmane.org/gmane.comp.audio.pulseaudio.general/22375
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. But I don't know
the answer.
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=pa_xnew0_internal
--alloc-fn=pa_xnew_internal ./pulseaudio
And running massif does not make sense if only client libs are enabled.
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.
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;
+} else
+break;
+}
n_bytes -= written;
}
}
+done:
input_underrun = pa_sink_process_input_underruns(u-sink, left_to_play);
if (u-use_tsched) {
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example at
the end of /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf
that could be a potential reason to apply this patch.
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(mainloop.c:900)
==555==by 0x50EF519: pa_mainloop_iterate (mainloop.c:931)
==555==by 0x50EF5BF: pa_mainloop_run (mainloop.c:946)
==555==by 0x406DB5: main (main.c:1136)
==555==
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);
PA_LLIST_HEAD_INIT(pa_memimport, p-imports);
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);
pa_memblockq_drop(u-output_q, chunk-length);
-/** FIXME: Uh? you need to unref the chunk here! */
-
//pa_log_debug(gave %ld, chunk-length/fs);
//pa_log_debug(end pop);
return 0;
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See also
https://www.mail-archive.com/ubuntu-audio-dev@lists.launchpad.net/msg00268.html
The warning may be useful for users who carried over the module-loading
statement from default configuration files shipped with old PulseAudio
versions.
Signed-off-by: Alexander E. Patrakov patra
See objections to the code in this email:
http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-March/020174.html
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
---
src/modules/module-equalizer-sink.c | 4
1 file changed, 4 insertions(+)
diff --git a/src/modules/module
(mainloop.c:900)
==23564==by 0x50EF519: pa_mainloop_iterate (mainloop.c:931)
==23564==by 0x50EF5BF: pa_mainloop_run (mainloop.c:946)
...and Connection failure: Protocol error on the pactl side.
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From a160d6e1513aea0efb71ad798d409f0a16399430 Mon Sep 17 00:00:00 2001
)
+goto fail;
+
srh = pa_memblock_acquire(sr-memblock);
pa_zero(*srh);
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03.12.2014 16:55, Alexander E. Patrakov wrote:
03.12.2014 16:37, David Henningsson wrote:
In case there are still devices in the hashmap when the module is
unloaded, we need to free the hashmap before the devices, because
the hashmap key points to the device's name instead of making a copy
-loaded_device_paths)
pa_hashmap_free(u-loaded_device_paths);
+if (u-discovery)
+pa_bluetooth_discovery_unref(u-discovery);
+
pa_xfree(u);
}
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with everyone?
OK for me.
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Also, don't build it by default.
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
---
configure.ac| 24 +---
src/modules/module-equalizer-sink.c | 4
2 files changed, 25 insertions(+), 3 deletions(-)
diff --git a/configure.ac b
positive enough. The patch indeed helps against
valgrind errors if no audio is being streamed when pulseaudio gets a
SIGTERM. The newly posted error is a pre-existing one - i.e. the patch
fixes only one out of two or three problems.
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computer, or making a symlink from extra-hdmi.conf to default.conf (in
/usr/share/pulseaudio/alsa-mixer/profile-sets).
Does a pulseaudio server restart also resolve the problem?
No. udevadm trigger does.
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: pa_module_free (module.c:227)
==16639==by 0x4E61929: pa_module_unload_all (module.c:292)
==16639==by 0x406476: main (main.c:1161)
There were no Bluetooth sinks and sources. Also, this PC does not have
ofono installed.
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19.11.2014 11:08, Alexander E. Patrakov wrote:
19.11.2014 10:49, David Henningsson wrote:
If possible, it would make more sense to improve the library of our
choice (probably speex) to support our needs, instead of rewriting a new
resampler from scratch. That way, other apps using
?
Luke
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pa_resampler_request.
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19.11.2014 00:42, Andrey Semashev wrote:
On Tuesday 18 November 2014 21:32:53 Alexander E. Patrakov wrote:
18.11.2014 19:14, David Henningsson wrote:
On 2014-11-18 14:46, Alexander E. Patrakov wrote:
Add support for libsoxr resampler: David's objection about overriding
pa_resampler_request
19.11.2014 10:49, David Henningsson wrote:
On 2014-11-18 20:58, Alexander E. Patrakov wrote:
19.11.2014 00:42, Andrey Semashev wrote:
On Tuesday 18 November 2014 21:32:53 Alexander E. Patrakov wrote:
18.11.2014 19:14, David Henningsson wrote:
On 2014-11-18 14:46, Alexander E. Patrakov
). There, 100% is the correct default, especially
because volume is expected to be the same by default in the cases of
passthrough and normal playback.
P.S. I have not read enough context, sorry if this code path is already
only ever executed for analog outputs.
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to
unwanted distortions at one target frequency, these contributions are
uncorrelated.
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13.11.2014 02:13, Martin Blanchard wrote:
On Tue, 2014-11-11 at 23:45 +0500, Alexander E. Patrakov wrote:
Hi.
Hello Alexander!
I have cloned the raop2-v2 branch from
https://github.com/colinleroy/pulseaudio
First, thanks for the testings.
The following test was applied.
I have
is not fixable without external help.
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is strange.
I will recheck the quality separately later today, in order to verify
that it is still as good as in the previous tests. Please don't merge
the patches until this is done.
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12.11.2014 15:59, Andrey Semashev wrote:
On Wed, Nov 12, 2014 at 12:26 PM, Alexander E. Patrakov
patra...@gmail.com wrote:
We generally don't need a zoo of resamplers. But you have definitely changed
something important from an earlier submission by Peter Meerwald so that the
CPU figure became
for that. It doesn't have a way to
inject a custom message, but it does log a file and a line.
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and
newly-rendered frames), and write (via mmap) as much as possible.
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13.11.2014 01:16, Andrey Semashev пишет:
On Wednesday 12 November 2014 20:03:48 Alexander E. Patrakov wrote:
12.11.2014 14:26, Alexander E. Patrakov пишет:
I will recheck the quality separately later today, in order to verify
that it is still as good as in the previous tests. Please don't
the little extra would be desired in embedded domain
to conserve battery.
Well, in the embedded domain, there is usually no FPU or a bad FPU, and
a different resampler (hint: speex-fixed-X) is needed anyway :)
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distributors, not users.
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06.11.2014 15:50, Alexander E. Patrakov wrote:
06.11.2014 15:39, David Henningsson wrote:
On 2014-09-30 15:59, Alexander E. Patrakov wrote:
30.09.2014 19:07, David Henningsson wrote:
On 2014-09-26 14:17, Tanu Kaskinen wrote:
A summary of my proposal:
To add LFE low-pass filtering, just
09.11.2014 21:59, Peter Meerwald wrote:
the macro PA_USED... may be used to suppress a warning when a variable
+#define PA_UNUSED
Misleading subject and commit message?
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06.11.2014 15:39, David Henningsson wrote:
On 2014-09-30 15:59, Alexander E. Patrakov wrote:
30.09.2014 19:07, David Henningsson wrote:
On 2014-09-26 14:17, Tanu Kaskinen wrote:
A summary of my proposal:
To add LFE low-pass filtering, just hack the current remixer
implementation, no need
) PA_GCC_PURE;
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+ * keep track of min_latency and reuse it when
+ * this routine is called from IO context */
u-min_latency_ref = u-source-thread_info.min_latency;
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today's meeting at
work :(
Patch 50 needs more testing on a card with a large FIFO, e.g. on a Xonar DX.
No opinion so far on other patches.
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by a bad edit to
/usr/share/alsa/alsa.conf.
The bad edit was:
-pcm.front cards.pcm.default
+##pcm.front cards.pcm.front
+pcm.front cards.pcm.default
As a result, the front:0 device was redirected to dmix, which by default
accepts only 48 kHz.
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commands:
aplay -vv -d 1 -f cd -D hw:0 /dev/zero
aplay -vv -d 1 -f cd -D front:0 /dev/zero
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it) while
PulseAudio is not running?
aplay -vv -d 1 -f cd -D hw:0 /dev/zero
aplay -vv -d 1 -f cd -D front:0 /dev/zero
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29.10.2014 15:37, Alexander E. Patrakov wrote:
29.10.2014 15:14, mailing lists пишет:
Alexander, thank you very much for the detailed explanation.
The theory is very simple. It probes at startup whether the configured
sample rate and the alternate sample rate (both can be seen in
/etc/pulse
29.10.2014 15:37, Alexander E. Patrakov пишет:
29.10.2014 15:14, mailing lists пишет:
Alexander, thank you very much for the detailed explanation.
The theory is very simple. It probes at startup whether the configured
sample rate and the alternate sample rate (both can be seen in
/etc/pulse
kHz if this is supported by the card. Then,
if some music appears before module-echo-cancel corks its stream due to
timeout, it will also get resampled to 48 kHz.
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Alexander E. Patrakov
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, const char *s) {
While you are at it, could you please move the (pre-existing) magic
number 100 into a #define?
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Alexander E. Patrakov
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= pa_packet_new_data(data, length));
pa_tagstruct_free(t);
pa_pstream_send_packet(p, packet, ancil);
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Alexander E. Patrakov
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24.10.2014 01:44, Andrew Eikum wrote:
On Fri, Oct 24, 2014 at 12:38:16AM +0600, Alexander E. Patrakov wrote:
24.10.2014 00:08, Andrew Eikum wrote:
I have a patch in-hand which improves the current driver
significantly, but it isn't enough to fix the USB devices problem. If
someone wants
|2 +-
src/pulsecore/source-output.c |2 +-
src/pulsecore/source.c |2 +-
src/tests/srbchannel-test.c | 18 +-
18 files changed, 38 insertions(+), 41 deletions(-)
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Alexander E. Patrakov
, i.e. in the submitted
patch, and not in libavresample itself. Thus, it should be searched for
and fixed.
As for quality evaluation, I have not performed any testing yet, and
will do it live during today's discussion of resampler quality evaluation.
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Alexander E. Patrakov
make any sense out of it. And
anyway, this would be a different patch.
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Alexander E. Patrakov
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