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1023 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
fcfc6a990e Small refactoring of NetEq unittest for CNG with clock drift 
Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.

This change is only made for NetEq4.

R=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/8599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 11:42:28 +00:00
andrew@webrtc.org
17342e5092 Add a method to inform AudioProcessing that its output will be muted. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 22:28:31 +00:00
jiayl@webrtc.org
de782180b0 Change the type of propagation delta from int64 to int. 
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.

BUG=2910
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 19:19:23 +00:00
andrew@webrtc.org
07b5950c12 Initialize key_pressed_. 
Was resulting in an error on Mac Asan:
[ RUN      ] ApmTest.DebugDump
[libprotobuf FATAL ../../third_party/protobuf/src/google/protobuf/message_lite.cc:224] CHECK failed: !coded_out.HadError():

TBR=aluebs

Review URL: https://webrtc-codereview.appspot.com/8539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 16:41:13 +00:00
andrew@webrtc.org
ce8e077cf0 Add a keypress field to the audioproc debug proto. 
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.

TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.

R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 15:28:30 +00:00
andrew@webrtc.org
75dd2885c5 Add an interface for accepting keypress signals to AudioProcessing. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5529 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 20:52:30 +00:00
fischman@webrtc.org
8685af7ea0 Remove "Too long processing time of Incoming frame" logspam. 
This isn't indicative of anything actionable and spams android logcat with times
in the 10-30ms range several times per second.

BUG=2732
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 17:48:11 +00:00
turaj@webrtc.org
a80be4b23c Add boundary checking to supress gcc 4.8.3 warning. 
BUG=2888
Test=try, voe_cmd_test

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 16:38:45 +00:00
michaelbai@google.com
82ebb463fd Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. 
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.

It make the simliar feature's implementation consistent.

R=andrew@webrtc.org, fischman@webrtc.org, fischman@chromium.org
BUG=334447

Committed: https://code.google.com/p/webrtc/source/detail?r=5517

Review URL: https://webrtc-codereview.appspot.com/7769006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 04:48:27 +00:00
michaelbai@google.com
a65abf9d3a Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." 
This reverts commit 7686f0ddda717a9e776be0e219f039f68a10f9ed.

BUG=

TBR=andrew@webrtc.org, fischman@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/8369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 19:26:26 +00:00
jiayl@webrtc.org
1f64f06784 Add stats of incoming frame delays for debugging bandwidth estimation. 
BUG=crbug/338380
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 19:12:14 +00:00
michaelbai@google.com
7686f0ddda Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. 
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.

It make the simliar feature's implementation consistent.

R=andrew@webrtc.org, fischman@webrtc.org, fischman@chromium.org
BUG=334447

Review URL: https://webrtc-codereview.appspot.com/7769006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 17:42:34 +00:00
sergeyu@chromium.org
ad3035fc9e Fix WindowCapturerWin to unselect bitmap before destroying DC. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=2901
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 21:24:04 +00:00
stefan@webrtc.org
77c917a6ee Plot the capacity of a trace-based delivery filter. 
Breaks out the instantaneous rate counters to its own class.

R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 16:34:47 +00:00
stefan@webrtc.org
c88d3368d5 Add delay and send/receive throughput plots to BWE simulation. 
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 15:57:14 +00:00
henrik.lundin@webrtc.org
75642fcd9a Implementing replacement audio support in neteq_rtpplay 
This CL makes it possible to replace the payload in an RTP stream
with audio from another (PCM) file. The new payload will be encoded as
PCM16b. The RTP headers will be updated to reflect this change of
payload type.

BUG=2834
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 08:49:13 +00:00
henrik.lundin@webrtc.org
e6ab21b9ca Fixing a bug in DummyRTPpacket 
This bug caused writing outside allocated memory when RTP header
extensions were used.

BUG=2834
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8009005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 08:46:46 +00:00
andrew@webrtc.org
54744918ef Update AudioProcessing::Create docs. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/8039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 06:30:29 +00:00
jiayl@webrtc.org
20a60ea39d Fix a cursor capturing issue on Windows. 
The input position to WindowFromPoint should be relative to the desktop, not
relative to the window; if the result from WindowFromPoint is a child window
of the shared top window, it should be captured.

BUG=
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5487 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 17:49:12 +00:00
stefan@webrtc.org
0e5a2b5de6 Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered. 
This can happen when switching between multiple streams and a single while getting feedback from the receiver.

BUG=2881
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 14:38:25 +00:00
andrew@webrtc.org
f6a638e001 Trivial rename of non-compile time consts. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7669006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 01:31:28 +00:00
stefan@webrtc.org
422fdbf502 Wire up feedback to VideoSender. 
BUG=
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 16:33:50 +00:00
aluebs@webrtc.org
c9ee412070 Re-enabling audio processing tests 
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 14:41:57 +00:00
xians@webrtc.org
c1e28038ba Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-02 15:30:20 +00:00
jiayl@webrtc.org
1af5ea0538 Implement single monitor capture on Mac. 
BUG=2787, 2824
TESTED=MacBook Pro Retina with an external monitor; verified changing display configuration while capturing; add/remove monitor while capturing; verified cursor position.
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-01 02:03:24 +00:00
henrik.lundin@webrtc.org
83aee8f450 Fixing test name for NetEqPerformanceTest 
The naming did not follow conventions.

BUG=2859
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-31 11:46:34 +00:00
stefan@webrtc.org
1dd9b4d98e Add BWE tools for parsing RTP files. 
bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates.
bwe_rtp_to_text parses an RTP file and outputs the result to a text file.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7689006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-31 09:15:48 +00:00
jiayl@webrtc.org
bda5fa77af Fix the mouse cursor offset issue on Mac. 
The problem is that MouseCursorMonitor returns coordinates in DIPs, while DisplayAndMouseComposer assumes that they are in physical pixels. The fix is to convert the position to physical pixels in MouseCursorMonitorMac.

R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7739006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 23:27:35 +00:00
henrikg@webrtc.org
c693704cc2 Move out typing detection to its own class. 
This will allow an embedder to use it directly.

Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)

R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 09:50:46 +00:00
jiayl@webrtc.org
cf1b51b6fb Moves the display reconfiguration callback into a separate class, 
so that it can be shared with the cursor monitor when single monitor capturing
is added (https://webrtc-codereview.appspot.com/4679005/).
This Cl should have no functionality change.

BUG=2253
R=henrike@webrtc.org, sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 21:59:12 +00:00
stefan@webrtc.org
f7c6e743b3 Fix deadlock in video_receiver.cc. 
In webrtc::vcm::VideoReceiver::ResetDecoder(), the lock order is:
1. take _receiveCritSect,
2. take process_crit_sect_

This conflicts with the follow code path:
1. webrtc::vcm::VideoReceiver::Process(), take process_crit_sect_
  call -> webrtc::vcm::VideoReceiver::NackList(),
2.  with nackStats=kNackKeyFrameRequest, take _receiveCritSect

BUG=2861
TEST=trybots
R=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 10:27:51 +00:00
andrew@webrtc.org
c7c7a531f3 Add Config struct for experimental AGC. 
Disable in the audio mixer.

TBR=aluebs,bjornv
BUG=2844

Review URL: https://webrtc-codereview.appspot.com/7769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 04:57:25 +00:00
mallinath@webrtc.org
7433a088d2 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." 
We reverted the r5421 to allow us roll webrtc to chrome without any modifications
to libjingle. Since webrtc is rolled with r5444, we can add back the original CL
and changes to libjingle will be upstreamed in the next roll.

TBR=andresp@webrtc.org

> Revert 5421 "Fix deadlock on register/unregister observer while ..."
> 
> Failure to compile on Chromium Internal bots, because of API changes.
> 
> http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
> 
> You need to follow the steps mentioned in 
> https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
> 
> Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
> as mentioned in the doc.
> 
> > Fix deadlock on register/unregister observer while there is a an going callback.
> > 
> > BUG=2835
> > R=mallinath@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/7119005
> 
> TBR=andresp@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7679004

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 00:56:02 +00:00
henrik.lundin@webrtc.org
84eb0e952e Add clean test to NetEq perf test 
Add another test to NetEqPerformanceTest with no packet losses or
clock drift. The purpose of this test would be to focus on the
"clean" code path, i.e., the path taken when there are no network
problems. The reason is that this code path is presumably much
lighter in complexity, and regressions could easily drown in the
heavier code involved when combating losses and drift.

BUG=2859
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 21:50:35 +00:00
fischman@webrtc.org
932b0193e7 VideoCaptureAndroid: stop preview in opposite order of starting. 
While the SDK documentation doesn't prescribe a required shutdown order, good
hygiene suggests stopping should happen in reverse order of starting.  It also
seems to relieve a crash in the system capturer on at least the Galaxy Note 10.

BUG=2793
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 22:32:05 +00:00
mallinath@webrtc.org
18586d38bc Revert 5421 "Fix deadlock on register/unregister observer while ..." 
Failure to compile on Chromium Internal bots, because of API changes.

http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio

You need to follow the steps mentioned in 
https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.

Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
as mentioned in the doc.

> Fix deadlock on register/unregister observer while there is a an going callback.
> 
> BUG=2835
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7119005

TBR=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 22:00:57 +00:00
sprang@webrtc.org
a45cac0fb7 Avoid potential dead lock in StreamStatisticianImpl 
Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to
ReceiveStatisticsImpl, into separate methods with guards agains having
incorrect lock order.

BUG=2856
R=andresp@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 16:22:08 +00:00
sprang@webrtc.org
5314e85926 Race condition in RTPSender::UpdateRtpStats 
The ssrc should not be access directly from the ssrc_ field, without
holding the send_critsect_ lock. A better way is to just use the SSRC()
getter method.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7539006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 13:20:36 +00:00
andresp@webrtc.org
2397a17c6b Fix bug introduced during replace of list wrapper with std equivalents in r5378. 
R=henrika@webrtc.org, pbos@webrtc.org, henrike@webrtc.org
TBR=henrike@webrtc.org
BUG=2164

Review URL: https://webrtc-codereview.appspot.com/7639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 12:33:30 +00:00
sprang@webrtc.org
c00adbed73 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket 
StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should
  be cached while holding lock to avoid race condition.

  Also, rtp_callback_ do not need to be called in GetStatistics() at all

BUG=2853
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 10:42:48 +00:00
pbos@webrtc.org
99eab02fb1 Fix "field '_testNo' is uninitialized" warnings. 
BUG=2849
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 09:30:35 +00:00
andrew@webrtc.org
e84978f3d8 Add a Config parameter to AudioProcessing::Create(). 
Also add a parameter-less version; the (int) version is deprecated and
should be removed.

TBR=aluebs,bjornv
BUG=2844

Review URL: https://webrtc-codereview.appspot.com/7609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-25 02:09:06 +00:00
asapersson@webrtc.org
871d949299 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 13:23:49 +00:00
stefan@webrtc.org
99a8c7e039 Add trace-based delivery filter to BWE test framework. 
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 10:00:27 +00:00
andresp@webrtc.org
8d375c95b7 Fix deadlock on register/unregister observer while there is a an going callback. 
BUG=2835
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 23:09:25 +00:00
andrew@webrtc.org
754de528b7 Fix array declarations in aec_rdft.h. 
Was causing warnings in Chromium such as:
warning C4742: 'rdft_wk2i' has different alignment in
'webrtc\modules\audio_processing\aec\aec_rdft_sse2.c' and
'webrtc\modules\audio_processing\aec\aec_rdft.c': 4 and 16

BUG=chromium:336620
R=cduvivier@google.com

Review URL: https://webrtc-codereview.appspot.com/7489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 20:55:14 +00:00
sprang@webrtc.org
0e93257cee Add callbacks for receive channel RTP statistics 
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.

TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 10:00:39 +00:00
sprang@webrtc.org
7dba27c740 Potential dead lock in receive statistics 
A dead lock could occur if the following to code paths are called
concurrently:

ReceiveStatisticsImpl::IncomingPacket() ->
  StreamStatisticianImpl::IncomingPacket()

StreamStatisticianImpl::GetStatistics() ->
  ReceiveStatisticsImpl::StatisticsUpdated()

Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().

BUG=2818
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 16:33:37 +00:00
elham@webrtc.org
32c3247418 Fix for libtalkmobile build error 
bug=b/12549061

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 16:16:58 +00:00
asapersson@webrtc.org
efaeda0c76 Add configuration and test for extended RTCP reference time reports to new video api. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 08:34:49 +00:00