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minyue@webrtc.org 6fd9308420 Suppressing warnings in GetRTT() in VoE. 
GetRTT() was separated from GetRTPStatistics() but the warnings were not updated.

Now GetRTT() is only only used by GetRTPStatistics() and the warning pops up pointlessly and too often.

This CL is to suppress these warnings and maintain a proper warning for GetRTPStatistics().

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7899 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 14:56:44 +00:00
..
base
Revert r7885.
2014-12-15 08:04:50 +00:00
build
Move isolate path into webrtc/build/android/test_runner.py
2014-12-11 11:59:46 +00:00
common_audio
Add NEON intrinsics version for min_max_operations_neon.c
2014-12-15 06:07:47 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
Revert r7886:7887.
2014-12-15 07:03:04 +00:00
modules
Use webrtc_root instead of DEPTH for iSAC.
2014-12-15 10:56:50 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
Revert r7885.
2014-12-15 08:04:50 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Add video send bitrates to histogram stats:
2014-12-09 09:47:53 +00:00
test
Removing manual test pages because they have been moved to github.
2014-12-12 09:30:41 +00:00
tools
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video
Wire up RTT statistics to webrtc::Call.
2014-12-11 13:26:09 +00:00
video_engine
Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval.
2014-12-10 11:04:13 +00:00
voice_engine
Suppressing warnings in GetRTT() in VoE.
2014-12-15 14:56:44 +00:00
.gitignore
…
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Wire up RTT statistics to webrtc::Call.
2014-12-11 13:26:09 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Add video send bitrates to histogram stats:
2014-12-09 09:47:53 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
…
config.cc
Log formatting fix for VideoEncoderConfig.
2014-11-06 09:35:08 +00:00
config.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Put send-side bwe probing under finch experiment.
2014-11-10 13:55:16 +00:00
webrtc.gyp
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.