1df9dc3957
Isolate register post encode callback in video coding module to simplify code and critical sections.
...
R=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5357 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-09 08:01:57 +00:00
b08a12d6e8
Isolate debug recording from video sender into a thread safe small class.
...
R=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5353 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-08 12:38:22 +00:00
bccd53de57
Delay Estimator: Converts a constant into a configurable parameter.
...
The parameter is used in the robust validation scheme, which will be turned on in a separate CL.
* Setter and getter for allowed delay offset.
* Updated unittests.
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5351 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-08 08:18:15 +00:00
d335094852
Init to 16 kHz in the fixed-point profile.
...
Fixes modules_unittests for fixed-point builds (Android).
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/6709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5349 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 18:57:10 +00:00
b6541ca3a1
Ensure capture_levels_ is sized correctly at init time.
...
Fixes failing voe_auto_test and audioproc_perf.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/6699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5348 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 18:36:10 +00:00
60730cfe3c
Remove the requirement to call set_sample_rate_hz and friends.
...
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org , bjornv@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 17:45:09 +00:00
a89d17d5b7
Delay Estimator: robust_validation should be stored over a reset
...
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-02 07:07:04 +00:00
2fb72cfeec
Add include guards to forward_error_correction_internal.h
...
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5789005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5335 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-24 05:06:12 +00:00
000dde99c8
Android build: make it quiet on success and not overly noisy on failure.
...
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 22:49:35 +00:00
f6acf98a46
Fix the android clang bot for compiling with thread annotations.
...
TBR=niklas.enbom@webrtc.org
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6279005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5330 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 21:54:26 +00:00
7fb75ecbd4
Add thread_annotations for clang targets.
...
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.
R=niklas.enbom@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 20:20:50 +00:00
54ae4ffb9e
Add callbacks for receive channel RTCP statistics.
...
This allows a listener to receive new statistics 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=henrika@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19 13:26:02 +00:00
e682aa5077
Refactoring MediaOptimization so it can easily be turned into a thread-safe class.
...
BUG=2732
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5322 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19 10:59:48 +00:00
8ae72560dd
Make MouseCursor mutable
...
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.
R=dcaiafa@chromium.org
Committed: https://code.google.com/p/webrtc/source/detail?r=5310
Review URL: https://webrtc-codereview.appspot.com/6059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5314 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 02:18:01 +00:00
f8be8df33a
audio_processing_unittest: unbreak clang compilation.
...
BUG=2735
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5313 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 23:46:39 +00:00
179908c81c
JNI Audio: remove dead members.
...
BUG=2735
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 23:46:14 +00:00
e4c927208b
Revert "Make MouseCursor mutable"
...
This reverts commit a6db8ab8bc4b569a26633b0ca3665297f1a5349b.
TBR=dcaiafa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5311 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 22:48:50 +00:00
8fd1d26536
Make MouseCursor mutable
...
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.
R=dcaiafa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5310 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 22:19:12 +00:00
e6b871bb29
Added method for getting default module state and protect agains a
...
read/write race for child_modules_.
BUG=2731
TEST=tsan
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5306 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 08:30:40 +00:00
eb7b7bce3d
Modify video_render/ to allow a single old frame.
...
This stabilizes tests as a single frame reaches end-to-end, as well as
allowing slow or heavily-loaded systems to see any video updates even if
the frame takes more than 500ms in the pipeline.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=2724
Review URL: https://webrtc-codereview.appspot.com/5949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5303 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 18:24:37 +00:00
e7b1e11283
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."
...
> Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
>
> > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> >
> > R=holmer@google.com
> >
> > Review URL: https://webrtc-codereview.appspot.com/5049004
>
> TBR=asapersson@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5799004
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:40:36 +00:00
1e7d61270c
Simplification of histogram normalization in delay estimator.
...
- Replaces a for loop with a single element update to save complexity. No regression in performance seen on set of recordings.
- Removes UpdatesMadeUponChange() and put code straight into ProcessBinarySpectrum().
BUG=None
TESTED=module_unittest, trybots, verified manually on set of recordings.
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5298 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 13:37:28 +00:00
5ab756703e
Revert r5294 to re-roll r5293.
...
To fix races in test each stream now owns its own encoder/decoder.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 12:24:44 +00:00
5c64508b03
Adds robust validation functionality to the delay estimator
...
Evaluated over a 51 recordings:
False positives went from 4.4% to 0.7%
Missed detections unchanged at 0.8%
No increase in complexity, but need to re-evaluate that.
TESTED=trybots, unittests, verified against Matlab implementation
BUG=None
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5296 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 10:57:53 +00:00
87ad57bc75
Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver
...
The iterator is incremented both in loop header and loop body. Should
only be incremented in header.
BUG=2727
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5295 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 07:43:51 +00:00
41e2615e02
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
...
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
>
> BUG=
> R=mflodman@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5409004
TBR=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15 18:42:32 +00:00
341e91441a
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
...
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 23:57:54 +00:00
dd393e7b9d
Measure pacer queue size based on when packets are inserted rather than captured.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5291 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 22:03:27 +00:00
24301a67c6
Update talk to 58174641 together with http://review.webrtc.org/4319005/ .
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 19:17:43 +00:00
86bb56a7f5
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
...
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
>
> R=holmer@google.com
>
> Review URL: https://webrtc-codereview.appspot.com/5049004
TBR=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 16:16:45 +00:00
6811b6e308
Callback for send bitrate estimates - new roll
...
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.
Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.
The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:
webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock
webrtc::Bitrate::Process() // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...
webrtc::Bitrate::Update() // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...
This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.
BUG=2235
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 09:46:59 +00:00
e9abd591d7
Making RemoteRateControl::min_configured_bit_rate_ configurable
...
The minimum bitrate can now be configured from WrappingBitrateEstimator.
BUG=2698
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 08:42:42 +00:00
a92baead39
ACM 2 compatibility with ACM 1.
...
Removing an unregisterd codec from ACM 1 does not result in an error, so should be for ACM 2. Also ACM 1 has post-decode VAD on and AMC 2 needs to have it on by default.
BUG=
Test=trybits
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5276 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:10:44 +00:00
9ee75e9c77
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
...
BUG=N/A
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:42:44 +00:00
724947b8ef
Add SwapFrame() to VideoSendStreamInput.
...
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:26:16 +00:00
096e8d9f94
Revert 5259 "Callback for send bitrate estimates"
...
CL is causing flakiness in RampUpTest.WithoutPacing.
> Callback for send bitrate estimates
>
> BUG=2235
> R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4459004
R=mflodman@webrtc.org , pbos@webrtc.org
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/5579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 14:07:33 +00:00
f9bdbe3619
Roll chromium_revision 232627:238260
...
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003
TEST=trybots passing
BUG=none
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
2656cf9f4c
Callback for send bitrate estimates
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 12:53:03 +00:00
7ae8495779
Removed unnecessary Pulse init from VoE startup.
...
Saves 10% (~260ms) of the total PeerConnectionTest wallclock time.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5254 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 21:01:34 +00:00
917306d3fd
Change uses of the obsolete armv7 setting to arm_version==7.
...
BUG=http://crbug.com/234135
R=andrew@webrtc.org , fischman@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5369004
Patch from Mostyn Bramley-Moore <mostynb@opera.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 09:26:07 +00:00
eb7def234e
Fix compilation errors on Fedora 20.
...
peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.
BUG=2700
R=andrew@webrtc.org , fischman@webrtc.org , henrik.lundin@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5019004
Patch from Victor Costan <costan@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 21:34:30 +00:00
de7c9e8884
Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.
...
Move the logic to common.gypi to reduce the chance of the define being
unprotected in the future.
BUG=b/12018143
TESTED=git try, and local Linux build with -Denable_video=0
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5244 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 16:23:00 +00:00
5e13ac967b
Add shape in DesktopFrame.
...
The shape will be used for Me2App mode in chromoting.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5243 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-07 01:03:28 +00:00
8d0ca7f5d2
Add new method to MockAudioProcessing.
...
TBR=henrikg
Review URL: https://webrtc-codereview.appspot.com/5279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5241 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 17:52:27 +00:00
863b536100
Allow opening an AEC dump from an existing file handle.
...
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.
This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.
BUG=2567
R=andrew@webrtc.org , henrika@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 16:05:17 +00:00
88615f0659
Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks
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BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5236 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 13:16:44 +00:00
96a9b2dcdc
Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
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R=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/5049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 15:06:56 +00:00
ebad765ee0
Add callbacks for send channel rtp statistics
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BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:29:02 +00:00
0a3c1471b8
Add API to query video engine for the send-side delay.
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R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:05:07 +00:00
a6ad6e5b58
Add callbacks for send channel rtcp statistics
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BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 09:48:44 +00:00
