Signed-off-by: Alexander E. Patrakov patra...@gmail.com
---
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
, overview and other unexpected places
where even fully covered windows can render through.
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---
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
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 patra...@gmail.com
---
src/modules/alsa/mixer/profile-sets/default.conf | 16
1 file changed, 16 insertions(+)
Tanu Kaskinen wrote
://www.alsa-project.org/alsa-info.sh
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to enable the peak meter. And with that, it's
perfectly reasonable for PulseAudio to eat 6% of your CPU.
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).
The patches are here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/19625
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/daemon.conf.
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that we cannot fix because we don't know
what's wrong. Please help us!
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it to a pastebin.
Once I see all of that, I will ask more questions about the card.
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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't
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.
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#
# Configuration for the ICE1712 (Envy24) chip
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 - CMSG_LEN(0
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 patra...@gmail.com
---
src/modules/rtp/rtp.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/src/modules/rtp/rtp.c b/src/modules
that the kernel doesn't tell us again and again about it.
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
---
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
/memblock.c:596, function
pa_memblock_unref(). Aborting.
And now this is CVE-2014-3970.
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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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.
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!);
+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 like
them?
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---
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
.
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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
(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
external filters.
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);
+}
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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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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
this on the fly, as implied by making it a module?
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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
wakeup), try using a USB headset.
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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.
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floating-point sample range [-1 .. 1]. speex
* without --enable-fixed-point works fine with this range.
* Care has been taken to call speex_resample_float() only
* for speex compiled without --enable-fixed-point.
*/
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-* in that case.
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
Reported-by: Fahad Arslan fahad_ars...@mentor.com
Cc: Damir Jelić poljari...@gmail.com
Cc: Peter Meerwald pme...@pmeerw.net
FIXED_POINT detection is based on code by Peter Meerwald.
---
src/pulsecore/resampler.c | 57
-* in that case.
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
Reported-by: Fahad Arslan fahad_ars...@mentor.com
Cc: Damir Jelić poljari...@gmail.com
Cc: Peter Meerwald pme...@pmeerw.net
FIXED_POINT detection is based on code by Peter Meerwald.
---
src/pulsecore/resampler.c | 55
a comment about the scaling problem somewhere.
If you prefer, I can do it on the next week, but today I am busy with
another patch.
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with the external speex echo canceller.
I defer the choice between the original patch and these alternatives to
someone else.
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uses different input and output scales. This
is not the case. So you need not only to multiply the input by 32768.0,
but also to divide the output.
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.
Signed-off-by: Alexander E. Patrakov patra...@gmail.com
Reported-by: Fahad Arslan fahad_ars...@mentor.com
Original-author: Peter Meerwald pme...@pmeerw.net
Cc: Damir Jelić poljari...@gmail.com
Cc: Peter Meerwald pme...@pmeerw.net
---
src/pulsecore/resampler.c | 25 ++---
1 file
to provide them at all. If
we fix that, then the extra multiplication in an already-very-slow path
would no longer be relevant.
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http
to
speex_resampler_process_float(), convert back to s16.
Variant B: feed directly to speex_resampler_process_int().
I.e. do not compare the two variants of speex, but two ways of using the
optimally-compiled speex.
I am going to extend this thought in another e-mail.
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not matter who converts s16 to float and back - speex or
PulseAudio.
So the proposal is: after a benchmark, kill the speex-float family of
resamplers and make it an alias of speex-fixed on all platforms (not
just on ARM) for compatibility.
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the
same question in
http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-May/020624.html
- and you already have an answer. So, based on your benchmark, if others
confirm it, let's just kill speex-float on all platforms.
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be removed from the list, because Peter
Meerwald added important runtime probing. The second patch seems to be
broken for normal speex. I intend to review Peter's patch tomorrow.
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The warnings were produced because the command-line flag redefined the
value of _FORTIFY_SOURCE coming from the specs on some distributions,
including Gentoo. So, undefine this macro before defining it.
---
v2: Documented the rationale behind the added flags, including compiler
warnngs.
Also,
to int16_t / float for consistency (as suggested
by A. Patrakov)
pa_set_remap_func() still takes the function pointers with void arguments,
this may be changed lateron
With that taken into account, the series looks good.
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no runtime testing.
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26.04.2014 18:19, Alexander E. Patrakov wrote:
24.04.2014 22:09, Peter Meerwald wrote:
From: Peter Meerwald p.meerw...@bct-electronic.com
The generic matrix remapping is rather inefficient; special-case code
improves performance by 3x easily.
I have looked at this and the 10th patch. For 10
yet.
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program
that can manipulate the parameters of LADSPA plugins loaded by
module-ladspa-sink.
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22.04.2014 13:01, David Henningsson wrote:
On 2014-04-21 09:04, Alexander E. Patrakov wrote:
21.04.2014 07:49, David Henningsson wrote:
On 2014-04-20 21:26, Alexander E. Patrakov wrote:
Thus, it is not possible to tell the hardware device (that can use
rewinds) from a properly wrapped
22.04.2014 20:20, David Henningsson wrote:
On 2014-04-22 10:18, Alexander E. Patrakov wrote:
22.04.2014 13:01, David Henningsson wrote:
So you want to always go low-latency (and high CPU/power consumption),
in case rewind is not possible?
Yes, exactly. And high CPU/power consumption
21.04.2014 07:49, David Henningsson wrote:
On 2014-04-20 21:26, Alexander E. Patrakov wrote:
Thus, it is not possible to tell the hardware device (that can use
rewinds) from a properly wrapped software encoder (that can't rewind and
doesn't pretend to be able to rewind), because for both cases
20.04.2014 16:04, Prunk Dump wrote:
2014-04-17 14:09 GMT+02:00 Alexander E. Patrakov patra...@gmail.com:
Well, the problem here is that the CIFS server gives extra unwanted access
rights to the directory. So PulseAudio rightfully complains. However, in
some cases (e.g. on CIFS and other non
20.04.2014 20:44, David Henningsson wrote:
On 2014-04-19 23:21, Alexander E. Patrakov wrote:
Since the dca ALSA plugin is based on extplug, it has a slave and
therefore
a card. Thus, PulseAudio thinks that it looks like hardware and attempts
rewinds, which this plugin cannot handle correctly
Initially (in commit ef422fa4ae626e9638ca70d1c56f27e701dd69c2),
pa_make_secure_dir followed a simple principle: make a directory, or,
if it exists, check that it is suitable. Later this evolved into make
a directory, or, if it exists, ensure that it is suitable. But the
check remained.
The check
...but this is rather risky from the security standpoint, so any
additional audit would be appreciated.
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Also, ignore unexpected fchmod results inside the user's home
directory. They sometimes happen due to home filesystems violating
POSIX requirements, including those specified at
http://pubs.opengroup.org/onlinepubs/009696899/functions/fchown.html
Details:
(-77)
https://bugzilla.redhat.com/show_bug.cgi?id=608936#c19
Point taken. As you seem to suggest that, with start threshold set to
boundary, the multi plugin fails on all cards, I will no longer
whitelist it.
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Since the dca ALSA plugin is based on extplug, it has a slave and therefore
a card. Thus, PulseAudio thinks that it looks like hardware and attempts
rewinds, which this plugin cannot handle correctly, because ALSA never
notifies extplug plugins about rewinds.
The same mishandling of rewinds
.
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() failures that are not even
supposed to be actionable and are not security-relevant, with a warning.
IMHO a good heuristic to decide whether to propagate fchown() failures
would be uid != -1, or, equivalently, a test for system mode.
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17.04.2014 18:51, Tanu Kaskinen wrote:
On Thu, 2014-04-17 at 17:09 +0600, Alexander E. Patrakov wrote:
17.04.2014 15:14, Tanu Kaskinen wrote:
Patch review status updated:
http://www.freedesktop.org/wiki/Software/PulseAudio/PatchStatus/
I cannot find this patch in any section of the wiki page
that is passed as a reference. On failure, the
* implementation is expected to return -1.
*
* You must use the function pa_sink_set_get_mute_callback() to
* set this callback. */
pa_sink_get_mute_cb_t get_mute;
Yes, thanks!
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has unclear semantics). Same
for sources.
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On Haswell hardware, there are multiple HDMI outputs capable of
digital sound output. As they were identically named, KDE's control
center was unable to distinguish them, restored the wrong profile and
thus routed sound to the wrong HDMI monitor.
---
On Haswell hardware, there are multiple HDMI outputs capable of
digital sound output. As they were identically named, KDE's control
center was unable to distinguish them, restored the wrong profile and
thus routed sound to the wrong HDMI monitor.
Also, having identically-named menu items in other
);
+}
What's the point of having this function in this patch? As demonstrated
above, set_reference_volume_internal does the same assertions.
Same for sources.
The patch does look like an equivalent refactoring.
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++)
-u-hrir_data[i * u-hrir_channels + j] /= hrir_max * 1.2;
-}
-}
+normalize_hrir(u);
/* create mapping between hrir and input */
u-mapping_left = (unsigned *) pa_xnew0(unsigned, u-channels);
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the conventional downmixer (in src/pulsecore/resampler.c), in
average for the music material, would be of the same subjective
loudness. I am not sure if that would be sufficient to eliminate the
clipping, but we can always multiply the result by the fudge factor
later if needed.
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is a library. LADSPA is
just a header defining the interface for plugins.
As for OLA vs OLS - I think that, after you mentioned a time-varying
filter, it is no longer a clear-cut question.
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Mark Lee m...@markelee.com wrote:
the front speakers are echoing content to the rear
speakers (with a small delay).
PulseAudio never adds such delayed channel copies. Check hardware settings.
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, then your original bug report warrants more
investigation here.
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wrong.
And I am not only talking about making sure the filter sampling is at
(filter_size + chunk_size + 1) (which is my point 2), but also about
making filter_size reasonable (which is point 3).
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:43 AM, Alexander E. Patrakov
patra...@gmail.com mailto:patra...@gmail.com wrote:
1. The FFT size and the window size for overlap-add are chosen
inconsistently. The window size is always 16000 samples. The FFT size
depends on the sample rate. Thus, with sample rates of 16 kHz
?
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---
src/modules/bluetooth/bluez5-util.c | 2 +-
src/modules/module-device-manager.c | 4 ++--
src/modules/module-stream-restore.c | 4 ++--
src/pulse/proplist.h| 2 +-
src/pulsecore/sink.c| 2 +-
src/pulsecore/source.c | 2 +-
6 files changed, 8
Due to e-mail size limitation, removals of the following files are not
included:
* src/modules/module-equalizer-sink.c
* src/utils/qpaeq
To apply the patch correctly, git am --scissors this e-mail and then
amend the commit by removing these two files.
Yes, I understand that it is generally a
2014-03-08 1:43 GMT+06:00 Alexander E. Patrakov patra...@gmail.com:
as enough time has passed and people only added FIXMEs instead of fixing
this (unfixable except by a full rewrite?) module.
Oops. This phrase was rude, as it misrepresented, among others, some
commits by Peter Meerwald, Arun
2014-03-08 7:34 GMT+06:00 David Henningsson david.hennings...@canonical.com:
On 03/07/2014 08:43 PM, Alexander E. Patrakov wrote:
Yes, I understand that it is generally a bad idea (and rude) to destroy
other people's contributions. But in this case, I think it is justified,
as enough time has
Also print a scary warning if one attempts to build or use the module,
as suggested by David Henningsson.
---
configure.ac| 22 --
src/modules/module-equalizer-sink.c | 7 ++-
src/pulse/proplist.h| 4 ++--
3 files changed, 28
(m-core, pa_rtclock_now() +
SAP_INTERVAL, sap_event_cb, u);
+u-allow_suspending_on_idle = allow_suspending_on_idle;
pa_source_output_put(u-source_output);
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be moved. It did not matter before,
because the policy was the same anyway. Now it matters, so please think
about it. Maybe PA_SOURCE_OUTPUT_DONT_MOVE is needed, too? I am not
sure, so please use your own judgement.
Reviewed-by: Alexander E. Patrakov patra...@gmail.com
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18.02.2014 22:12, Arun Raghavan wrote:
On 17 February 2014 23:04, Alexander E. Patrakov patra...@gmail.com wrote:
[...]
A sensible model has been demonstrated (as a replacement of the current
logic) that doesn't fit into those assumptions. E.g. (here I am deliberately
trying to misinterpret
that you brought up. Sorry if this ends up
misrepresenting the idea.
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anymore
to control the loudness of all applications at once, and thus don't need
any sort of increment-based volume API.
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with module-equalizer-sink too,
but I still recommend that you avoid loading that module.
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.
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of the resampler used in Wine). But I can't
make any statements about whether the new default is good enough
without the mentioned tool./off-topic
Contacts: Alexander E. Patrakov
Necessary background: digital sound processing, access to scientific
papers on the topic, python with numpy and scipy
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sudo, as follows:
alsamixer -c0
and of course without killing pulseaudio. This way, it will remember
the volume change, as it came from user input. How do the sliders in
pavucontrol move during that command?
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2014/1/23 Nikos Chantziaras rea...@gmail.com:
On 23/01/14 17:42, Alexander E. Patrakov wrote:
2014/1/23 Nikos Chantziaras rea...@gmail.com:
It seems that PA refuses to use the maximum hardware volume of my sound
card
(Asus Xonar D1). As a result, the audio is very silent. I have the volume
2014/1/23 Nikos Chantziaras rea...@gmail.com:
On 23/01/14 18:25, Alexander E. Patrakov wrote:
2014/1/23 Nikos Chantziaras rea...@gmail.com:
On 23/01/14 17:42, Alexander E. Patrakov wrote:
2014/1/23 Nikos Chantziaras rea...@gmail.com:
It seems that PA refuses to use the maximum hardware
level.
2. sudo alsactl store
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audio CD. It's another
matter that off-the-shelf hardware uses more sophisticated upmixing methods
(such as Dolby Pro Logic).
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http
David Henningsson wrote:
On 01/14/2014 04:47 PM, Alexander E. Patrakov wrote:
In my opinion (if this counts at all), this needs to be fixed in
PulseAudio, and not only because of your hardware. Indeed, the
SND_PCM_NO_AUTO_CHANNELS_{UP,DOWN} solution proposed by David would
work for you
Gene Heskett wrote:
On Wednesday 15 January 2014 18:02:04 Alexander E. Patrakov did opine:
2014/1/15 Gene Heskett ghesk...@wdtv.com:
Running an older ubuntu 10.04.4 LTS 32 bit build because a specially
patched rtai kernel is mandatory for one app. However, I am booted to
a 32bit
first. This is just to prevent the fallback to U8 or something else
equally strange. The -F option itself, as well as the list of recognized
formats, is described in speaker-test --help output and is indeed missing
from the manpage.
--
Alexander E. Patrakov
2014/1/16 Gene Heskett ghesk...@wdtv.com:
On Thursday 16 January 2014 12:08:37 Alexander E. Patrakov did opine:
Gene Heskett wrote:
Master and PCM, and those are apparently not slaved to the little
indicator applet that pops up with a horizontal slider at the top of
the screen when I hit
volume in
three situations: with only an unloaded 3.5mm jack connected, with
typical 32-ohm headphones connected, and with your speakers connected.
--
Alexander E. Patrakov
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of this suboptimal thing,
too, by implementing the equivalent functionality inside PulseAudio.
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Alexander E. Patrakov
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