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pbos@webrtc.org ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 13:50:16 +00:00
..
base
Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port
2014-12-18 04:45:05 +00:00
build
Move isolate path into webrtc/build/android/test_runner.py
2014-12-11 11:59:46 +00:00
common_audio
Make safe_conversions suitable for rtc_base_approved.
2014-12-17 22:56:09 +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 "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
2014-12-18 02:28:25 +00:00
modules
Refactor some receive-side stats.
2014-12-18 13:50:16 +00:00
overrides
Remove or rename typedefs with _t prefixes.
2014-12-17 13:43:55 +00:00
p2p
Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port
2014-12-18 04:45:05 +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
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
tools
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +00:00
video
Refactor some receive-side stats.
2014-12-18 13:50:16 +00:00
video_engine
Refactor some receive-side stats.
2014-12-18 13:50:16 +00:00
voice_engine
Refactor some receive-side stats.
2014-12-18 13:50:16 +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
Refactor some receive-side stats.
2014-12-18 13:50:16 +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
Refactor some receive-side stats.
2014-12-18 13:50:16 +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
…
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
…
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
…
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
Add AGC manager tests.
2014-12-16 14:48:47 +00:00
webrtc.gyp
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +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.