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hbos 41c5cd02fc Making ECDSA the default certificate regardless of WEBRTC_CHROMIUM_BUILD. 
Due to the experiment in chromium relying on KT_DEFAULT = KT_RSA (bug:
crbug.com/611698) a conditional was introduced. Now that the experiment is
ending and the experiment flag has been removed we can make KT_DEFAULT=KT_ECDSA
unconditionally.

BUG=chromium:611698

Review-Url: https://codereview.webrtc.org/2009533003
Cr-Commit-Position: refs/heads/master@{#12935}
2016-05-26 22:23:21 +00:00
..
api
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Making ECDSA the default certificate regardless of WEBRTC_CHROMIUM_BUILD.
2016-05-26 22:23:21 +00:00
build
iOS: Update MB config for moved iOS GN bots.
2016-05-26 09:49:25 +00:00
call
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
common_audio
Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ )
2016-05-26 21:48:27 +00:00
common_video
Delete IsMutable and IsExclusive methods.
2016-05-26 13:50:00 +00:00
examples
Improve IP_PATTERN in DirectRTCClient in Android AppRTC Demo
2016-05-25 09:26:29 +00:00
libjingle
Remove unused libjingle_xmpphelp target.
2016-05-24 18:33:39 +00:00
media
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
modules
Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ )
2016-05-26 21:48:27 +00:00
p2p
Increasing "converge" time for remote candidate pair selection in test.
2016-05-26 22:21:56 +00:00
pc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
sdk
Fix iOS GN build and cleanup system_wrappers
2016-05-25 18:37:17 +00:00
system_wrappers
Fix iOS GN build and cleanup system_wrappers
2016-05-25 18:37:17 +00:00
test
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
voice_engine
Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ )
2016-05-26 20:03:13 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
…
BUILD.gn
If building a fuzzer, define WEBRTC_UNSAFE_FUZZER_MODE.
2016-05-26 08:03:10 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +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
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
config.h
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +00:00
transport.h
…
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
…
video_frame.h
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
UI for AppRTC Android demo that doesn't require Design Support Library
2016-05-24 07:18:18 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Split TaskQueue out into a separate build target.
2016-05-24 18:40:44 +00:00
webrtc.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

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.