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zhihuang d82eee0675 Log how often DTLS negotiation failed because of incompatible ciphersuites. 
Log the DTLS handshake error code in OpenSSLStreamAdapter.
Forward the error code to WebRTCSession with the Signals.
This part is only for the WebRTC native code.
To make it work, need another CL for Chromium.

BUG=webrtc:5959

Review-Url: https://codereview.webrtc.org/2167363002
Cr-Commit-Position: refs/heads/master@{#13940}
2016-08-26 18:25:09 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
audio
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
base
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
build
Deactivated the intelligibility enhancement functionality by default
2016-08-26 14:16:13 +00:00
call
Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video.
2016-08-23 20:00:21 +00:00
common_audio
Replace calls to assert() with RTC_DCHECK_*() in .c code
2016-08-26 11:33:41 +00:00
common_video
GN: Synchronize resources between Android and iOS.
2016-08-24 14:48:48 +00:00
examples
Adding AecDump functionality to AppRTCDemo for iOS
2016-08-26 05:15:22 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Deactivated the intelligibility enhancement functionality by default
2016-08-26 14:16:13 +00:00
modules
Added logging of the level controller metrics.
2016-08-26 18:15:52 +00:00
p2p
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
pc
Renaming BaseChannel methods and adding comments for added clarity.
2016-08-25 20:31:24 +00:00
sdk
Adding AecDump functionality to AppRTCDemo for iOS
2016-08-26 05:15:22 +00:00
stats
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +00:00
system_wrappers
Replace calls to assert() with RTC_DCHECK_*() in .c code
2016-08-26 11:33:41 +00:00
test
GN: Fix Windows Clang errors
2016-08-25 11:15:46 +00:00
tools
Add missing "//build/config/sanitizers:deps" to executable targets.
2016-08-25 09:44:11 +00:00
video
Fixing TSan data race warning in video end-to-end tests.
2016-08-23 01:14:21 +00:00
voice_engine
Added a level indicator to new mixer.
2016-08-24 08:17:20 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
GN: Synchronize resources between Android and iOS.
2016-08-24 14:48:48 +00:00
call.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
codereview.settings
…
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
GN: Fix test_support_unittests and MIPS compile issue.
2016-08-23 11:52:19 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.