ution from the previous lines was double-counted, unless I am
missing something.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
ime = (snapshot->sink_now + snapshot->sink_latency -
buffer_latency) -
--
2.2.1
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulsea
), this rate is
invalid. Which place should be changed? I.e., should PulseAudio treat
this rate as valid, or should pa_alsa_get_supported_rates() not try this
rate?
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.
3 ms or lower the buffer latency for fixed latency devices.
Some of my USB devices run fine at a buffer latency of fragment size + 4 ms
instead of the dfault fragment size + 20 ms.
I tested it all with Intel HDA, USB and bluetooth sound devices. I would like to
see some test results fro
Same bug as in module-loopback, pointed out by Georg Chini in a private
email.
Signed-off-by: Alexander E. Patrakov
---
src/modules/echo-cancel/module-echo-cancel.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/src/modules/echo-cancel/module-echo-cancel.c
b/src/modules
Extracted from the yet-unsplit patch by Georg Chini with the subject
"[PATCH v4] Make module loopback honor requested latency".
---
src/modules/module-loopback.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
No Signed-off-by line, because the original patch did not have one.
diff --git a
ncy response of each laptop speaker
for a laptop with a built-in subwoofer. It would also be interesting to
see what Windows does here.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
, when someone implements
this properly, per-device equalizer settings and/or an impulse response
for digital room correction.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
g your note - see also
https://bugs.freedesktop.org/show_bug.cgi?id=50113 .
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
Didn't we recently apply a patch that removes the FSF address?
This review is incomplete, I intend to do one or more additional passes
later, or after resubmissions.
--
A
, short *src,
short *dest)
+{
...
+ dest[i] = z;
...
+}
The clamping is misplaced.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo
n
do 5.1 audio, what does the following command produce?
aplay -vv -f dat -d 1 -D surround51:9 /dev/zero
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
16.02.2015 16:35, Alexander E. Patrakov пишет:
16.02.2015 15:52, David Henningsson wrote:
Well, the big issue I guess is that in current state,
PA_SOURCE_OUTPUT_MESSAGE_GET_LATENCY messages always return 0.
Maybe there should be a sentence about that, or I could just push it as
it is. What do
n not
intended, the default PA_SOURCE_OUTPUT_MESSAGE_GET_LATENCY handler was
not called at all, and the latency was thus evaluated incorrectly.
Signed-off-by: Alexander E. Patrakov
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulse
application, if it exposes this
setting.
But in my opinion - it's just a question of additional infrastructure
complexity only for the marketing hype that exists for compressed formats.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing
always use PCM).
And they are staffed better than us.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
revert to ALSA instead because this is
not supported in Pulse audio.
No, that's about PCM.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
uses the system-wide copy. The difference is that
the system-wide copy hopefully contains explicit support for PulseAudio.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedeskto
11.02.2015 21:50, Alexander E. Patrakov wrote:
Signed-off-by: Alexander E. Patrakov
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 of th
Signed-off-by: Alexander E. Patrakov
---
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/modules/module
efault-fragments = 4
default-fragment-size-msec = 5
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
a card, with an additional
complication that 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.
--
Alexander E. Patrakov
_
fe_filter_update_rate(f, f->ss.rate);
+}
+
+pa_memchunk * pa_lfe_filter_process(pa_lfe_filter_t *f, pa_memchunk
*buf) {
Maybe a comment is needed that this function expects an average of all
non-LFE channels in the would-be-LFE channel? Not sure how this wou
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 u
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
han
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?
--
Alexander
yield at most 100 (with adjust_time of 10
seconds), and thus would be of the same order as the fudge factor. So -
the whole deadband, according to your own testing, works fine almost
without this term, maybe it is a good idea to delete it?
--
Alexander E. Patrakov
>From 3fc2d409dcee3d1e9a7ad5df
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
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
osition accurately enough. 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.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@l
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. These are not
n
alternative latency-snapshotting implementation, just to compare.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
allow using headphones at all on that PC if a
non-stereo profile is initially selected, so the question is pointless.
It currently can't switch from the stereo profile to 5.1, because it
never switched from 5.1 to stereo. As a user, I would indeed expect it
to switch
-working idea that we currently have on the topic,
and I have not reviewed all the fine details.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
user moves the audio stream from
the movie player from 5.1 HDMI sink of his video card to the built-in
stereo audio. So - no reason to worry at all.
No comments about the questions in the remainder of your email, because
I am not sure I understand them correctly.
--
Alexander E. Patra
s not reach
headphones.
Sorry, I 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.
--
Alexander E. Patrakov
___
pulseau
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
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.
--
Alexander E. Patrakov
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 and
d latency" a PI controller in disguise?
How was the lowpass filter tuned?
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
isable the effect if the user
plugs headphones in? Or is it yet to be written?
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
ing samples.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
t; idea. I
will review the actual implementation later.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
ned)(pa_rtclock_now() -
ts));
i.memblock = pa_memblock_new(pool, pa_usec_to_bytes(1*PA_USEC_PER_SEC,
&a));
@@ -450,8 +451,8 @@ int main(int argc, char *argv[]) {
pa_sample_format_to_string(b.format),
pa_sample_format_to_string(a.format));
-pa_assert_se(forth = pa_resampler_new(pool, &a, NULL, &b, NULL,
method, 0));
-pa_assert_se(back = pa_resampler_new(pool, &b, NULL, &a, NULL,
method, 0));
+pa_assert_se(forth = pa_resampler_new(pool, &a, NULL, &b, NULL,
crossover_freq, method, 0));
+pa_assert_se(back = pa_resampler_new(pool, &b, NULL, &a, NULL,
crossover_freq, method, 0));
i.memblock = generate_block(pool, &a);
i.length = pa_memblock_get_length(i.memblock);
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
t;lfe_filter)
+buf = pa_lfe_filter_process(r->lfe_filter, buf);
+
if (buf->length) {
buf = convert_from_work_format(r, buf);
*out = *buf;
diff --git a/src/pulsecore/resampler.h b/src/pulsecore/resampler.h
index 0580f85..f60f3b1 100644
--- a/src/pulsecore/resampler.h
++
| \-- hp1 --/
+ * |
+ * \-- hp0 --+-- lp2 --> MID (1)
+ * |
+ * \-- hp2 --> HIGH (2)
+ *
+ *[f0] [f1]
+ *
+ * Each lp or hp is an LR4 filter, which consists of two second-order
+ * lowpass or highpas
01.02.2015 01:29, Felipe Sateler wrote:
On Sat, Jan 31, 2015 at 4:56 PM, David Henningsson
wrote:
On 2015-01-31 08:19, Felipe Sateler wrote:
On Sat, Jan 31, 2015 at 4:52 AM, Alexander E. Patrakov
wrote:
31.01.2015 04:49, David Henningsson wrote:
2)
dpkg-shlibdeps: warning: couldn&#
same Debian
package" statement. At least for the previous version, they are not.
https://packages.debian.org/experimental/amd64/pulseaudio-module-raop/filelist
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.fr
100644 src/pulsecore/filter/lfe-filter.h
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
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 sho
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.
k up spurious jack-detection events.
All of that is with today's git master, and linux-3.19.0-rc4
--
Alexander E. Patrakov
Card #0
Name: alsa_card.pci-_00_03.0
Driver: module-alsa-card.c
Owner Module: 5
Properties:
alsa.card = &q
the during the next weekend (Fri-Sun),
though, 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.
--
Alexander E. Patrakov
__
19.01.2015 18:10, Felipe Sateler пишет:
On Mon, Jan 19, 2015 at 2:01 AM, Alexander E. Patrakov
wrote:
Here is the additional output when I play music with mpv to the sink. But
note that some (but not all!) of that may be due to OpenSSL misfeatures, it
may be a good idea to retest with LibreSSL
If you are using KDE, 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.
--
Alexander E. Patrakov
___
pulsea
19.01.2015 10:24, Arun Raghavan wrote:
On 19 January 2015 at 10:31, Alexander E. Patrakov 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 ./src/.libs/pulse
pper knows best 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.
--
A
it message included a link to
https://www.gnu.org/licenses/gpl-howto.en.html instead of just the words
"the GPL how-to". I have verified that the word "Boston" now only
appears in GPL and LGPL texts or in contexts not involving the FSF addre
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
pa_mutex_unlock(p->mutex);
return e;
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
orate? 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.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pul
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
07.01.2015 02:36, Andrey Semashev wrote:
On Tue, Jan 6, 2015 at 10:55 PM, Alexander E. Patrakov
wrote:
06.01.2015 19:17, Andrey Semashev wrote:
As far as I understand the code, the loop is already there in
pa_sink_input_peek() (see sink-input.c:924, "while (tchunk.length > 0)"
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
es" section, it
may be a good idea to note that 7.1 audio is now supported on HDMI outputs.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
;],
+AS_IF([test "x$enable_samplerate" == "xyes"],
"==" is a bashism. Use "=". Otherwise, the patch is good.
[PKG_CHECK_MODULES(LIBSAMPLERATE, [ samplerate >= 0.1.0 ],
HAVE_LIBSAMPLERATE=1, HAVE_LIBSAMPLERATE
d them in the attached file.
If your PC is not fast enough by default to valgrind pulseaudio (i.e. if
pulseaudio gets killed on starting a new stream due to exhausting the
realtime budget), this default.pa tweak can help:
load-module module-udev-detect tsched_buffer_size=5
--
Alexande
else
Does it mean it won't sleep for a period time but polling continuosly
when timer scheduling is disabled
No. sleep_usec gets eventually copied to rtpoll_sleep:
rtpoll_sleep = PA_MIN(sleep_usec, cusec);
...and then used as follows:
if (rtpoll_sleep > 0) {
pa
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
---
src/pulsecore/socket-server.c | 5 +
src/pulsecore/socket-server.h | 1 -
2 files changed, 1 insertion(+), 5 deletions(-)
diff --git
vice = pa_xstrdup(tcpwrap_service);
-}
+pa_assert_se(ss = pa_socket_server_new(m, fd));
+
+ss->type = SOCKET_SERVER_IPV6;
+ss->tcpwrap_service = pa_xstrdup(tcpwrap_service);
return ss;
--
Alexander E. Patrakov
___
pulseaudio-
eral/22373
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/22375
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
he reasoning must be different. But I don't know
the answer.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
tch with the subject "WIP: Trying to allow being on
unavailable profiles" which may help. After all, you have an unavailable
"headset_head_unit" profile.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
ES += daemon/pulseaudio.desktop
DISTCLEANFILES = esdcompat client.conf default.pa system.pa daemon.conf
start-pulseaudio-x11 pulseaudio.service
+if !CLIENT_LIBS_ONLY
+
if OS_IS_WIN32
SYMLINK_PROGRAM=cd $(DESTDIR)$(bindir) && cp
else
@@ -2202,6 +2225,8 @@ uninstall-hook:
rm
lay + n_bytes > sleep_bytes) {
+left_to_play += n_bytes;
+goto done;
+} else
+break;
+}
n_bytes -= written;
}
}
+done:
input_underrun = pa
nd of /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf
that could be a potential reason to apply this patch.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.o
A_LLIST_HEAD_INIT(pa_memimport, p->imports);
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
55==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==
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freed
pa_assert(chunk->memblock);
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;
--
Alexander E. Patrakov
_
cquire(sr->memblock);
pa_zero(*srh);
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
tunnel_manager_remote_server_set_failed(server, true);
+}
+}
Here one can reduce the need for "else" statements and thus increase
readability by handling the error cases first. I.e.: if
(!server->context) { ... ; return; }
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
te (mainloop.c:931)
==23564==by 0x50EF5BF: pa_mainloop_run (mainloop.c:946)
...and "Connection failure: Protocol error" on the pactl side.
--
Alexander E. Patrakov
>From a160d6e1513aea0efb71ad798d409f0a16399430 Mon Sep 17 00:00:00 2001
From: "Alexander E. Patrakov"
Date:
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
---
src/modules/module-equalizer-sink.c | 4
1 file changed, 4 insertions(+)
diff --git a/src/modules/module-equalizer-sink.c
b
For module-equalizer-sink, I have used the agreed-upon wording.
For module-dbus-protocol, I have tried to write something similar.
Alexander E. Patrakov (2):
Warn on loading module-equalizer-sink
Warn on loading module-dbus-protocol
src/modules/dbus/module-dbus-protocol.c | 5 +
src
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
---
src/modules
pa_hashmap_free(u->loaded_device_paths);
+if (u->discovery)
+pa_bluetooth_discovery_unref(u->discovery);
+
pa_xfree(u);
}
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-
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
->loaded_device_paths);
+if (u->discovery)
+pa_bluetooth_discovery_unref(u->discovery);
+
pa_xfree(u);
}
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists
.
Ok with everyone?
OK for me.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
rding that you proposed in the other
email. Would this phrase be OK for you?
module-equalizer-sink is currently unsupported, and can sometimes cause
PulseAudio to crash, produce excessive audio delays or hearable
artifacts. Use at your own risk.
--
Alexander E. Pat
02.12.2014 14:02, David Henningsson wrote:
On 2014-12-01 17:14, Alexander E. Patrakov wrote:
+AS_HELP_STRING([--with-fftw],[Build FFTW-using module
(equalizer). Note: the module has known-wrong DSP logic]))
+AS_IF([test "x$HAVE_FFTW" = "x1"], ENABLE_FFTW="yes (B
Also, don't build it by default.
Signed-off-by: Alexander E. Patrakov
---
configure.ac| 24 +---
src/modules/module-equalizer-sink.c | 4
2 files changed, 25 insertions(+), 3 deletions(-)
diff --git a/configure.ac b/configure.ac
index fe
practice we never know with analog outputs whether
there's a separate volume control or not.
Right.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
Signed-off-by: Alexander E. Patrakov
---
po/id.po | 2 +-
po/it.po | 4 ++--
src/daemon/server-lookup.c| 2 +-
src/daemon/server-lookup.h| 4 ++--
src/modules/dbus/iface-card-profile.h | 4 ++--
src/modules/dbus
ulting patch is attached, just in case, in order to avoid
damaging non-ASCII characters.
--
Alexander E. Patrakov
>From 7dcd197571840e467d688f0f7354253730bbcc15 Mon Sep 17 00:00:00 2001
From: "Alexander E. Patrakov"
Date: Sat, 29 Nov 2014 20:56:27 +0500
Subject: [PATCH] Fix the What
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.
--
Alexander E. Patrakov
_free(y->devices);
+
if (y->transports) {
pa_assert(pa_hashmap_isempty(y->transports));
pa_hashmap_free(y->transports);
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesk
calls
snd_pcm_start(), and test all available plugins using that program, so
that we know whether this is limited to the multi plugin.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
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.
--
Alexander E. Patrakov
___
pulseaudio-discuss mailing list
pul
60B48: 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.
--
Alexander E. Pat
201 - 300 of 668 matches
Mail list logo
