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honghaiz a58ea7806a 1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any. 
This will paves the way for continuous nomination lite and multi-networking.
2. Combined checking and pinging to remove some redundant checking and to make it switch to more frequent ping mode earlier.

Review URL: https://codereview.webrtc.org/1311433009

Cr-Commit-Position: refs/heads/master@{#10057}
2015-09-24 15:13:45 +00:00
..
base
H264 bitstream parser.
2015-09-24 13:07:04 +00:00
build
Don't link with audio codecs that we don't use
2015-09-24 10:18:48 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Enable SurfaceViewRenderer for AppRTCDemo
2015-09-24 14:02:15 +00:00
libjingle
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
modules
Stability improvement for audio recording on Android
2015-09-24 14:45:14 +00:00
overrides
Remove overridden basictypes.h.
2015-09-18 08:48:39 +00:00
p2p
1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any.
2015-09-24 15:13:45 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
test
This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e.
2015-09-18 13:41:18 +00:00
tools
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
video
Reland "Wire up send-side bandwidth estimation."
2015-09-24 13:07:17 +00:00
video_engine
Reland "Wire up send-side bandwidth estimation."
2015-09-24 13:07:17 +00:00
voice_engine
Fix the maximum native sample rate in AudioProcessing
2015-09-23 19:49:21 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Control combined_audio_video_bwe with config bool.
2015-07-24 14:10:25 +00:00
audio_send_stream.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
BUILD.gn
Revert "Android: Enable C99 mode instead of C89 (default)."
2015-09-21 14:50:56 +00:00
call.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove channel ids from various interfaces.
2015-09-17 21:06:02 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
config.h
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-09-24 10:18:48 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
libjingle_examples.gyp
Fix build when using Xcode 7 which contains .tbd files instead of .dylib
2015-09-05 03:01:08 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
video_receive_stream.h
Wire up currently-received video codec to stats.
2015-08-28 14:35:40 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
2015-09-08 12:13:25 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Fix BWE bug where audio has timestamps in us.
2015-09-23 13:54:04 +00:00
webrtc.gyp
This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e.
2015-09-18 13:41:18 +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.