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Steve Anton d5585ca956 Move almost all references from WebRtcSession to PeerConnection 
WebRtcSession is being merged into PeerConnection, and to make the
code review easier this is the first step towards achieving that.

Bug: webrtc:8323
Change-Id: I33778e46f20cb14089dff4328947868e207476bd
Reviewed-on: https://webrtc-review.googlesource.com/8760
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20413}
2017-10-24 17:59:20 +00:00
api Remove QUIC transport/data channel 2017-10-24 16:14:18 +00:00
audio New classes RefCounter and RefCountedBase. 2017-10-23 11:46:47 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Revert "Add fine grained dropped video frames counters on sending side" 2017-10-21 09:23:54 +00:00
common_audio Stop using std::tr1 2017-10-23 22:11:58 +00:00
common_video Use $rtc_libyuv_dir in common_video/BUILD.gn, not hard-encoded "libyuv" 2017-10-16 16:05:37 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Fixing crash in Mac client when no cameras are available. 2017-10-23 13:56:08 +00:00
infra Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
logging Prevent unbounded memory consumption through RtcEventLogImpl::config_history_ 2017-10-13 10:47:26 +00:00
media Remove QUIC transport/data channel 2017-10-24 16:14:18 +00:00
modules Add application extension field to RtpPacketReceived. 2017-10-24 14:22:18 +00:00
ortc Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers. 2017-09-27 09:14:28 +00:00
p2p Remove QUIC transport/data channel 2017-10-24 16:14:18 +00:00
pc Move almost all references from WebRtcSession to PeerConnection 2017-10-24 17:59:20 +00:00
resources Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process 2017-10-23 14:25:37 +00:00
rtc_base Revert "Enable the clang style plugin in rtc_base/" 2017-10-24 00:43:59 +00:00
rtc_tools Print state of AcknowledgedBitrateEstimator in event_log_visualizer. 2017-10-24 11:19:48 +00:00
sdk Support RGB frames in RTCCVPixelBuffer 2017-10-23 15:34:28 +00:00
stats Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
system_wrappers Implement Atomic32 using C++11's std::atomic 2017-10-20 23:25:04 +00:00
test Adding libFuzzer target for UlpFEC receiver. 2017-10-23 11:37:07 +00:00
tools_webrtc MB: Add Android Perf (swarming) 2017-10-23 13:02:07 +00:00
video Only treat H.264 frames containing SPS, PPS, and IDR as key frames. 2017-10-24 11:51:18 +00:00
voice_engine Let ChanneOwner use scoped_refptr and RefCountedBase. 2017-10-23 14:22:17 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. 2017-10-02 16:57:09 +00:00
.gn Remove remaining mentions of gflags 2017-09-25 15:34:41 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) 2017-09-25 13:37:12 +00:00
BUILD.gn Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." 2017-09-29 13:48:29 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
common_types.h Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. 2017-10-06 13:41:14 +00:00
DEPS Roll chromium_revision 65e1b24a78..f4ecd4bed3 (508708:508787) 2017-10-16 11:00:27 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
LICENSE_THIRD_PARTY Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
OWNERS Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
pylintrc Removing invalid-name from disabled pylint checks. 2017-10-11 08:06:49 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Update README.md and codereview.settings for new source location 2017-09-13 19:54:59 +00:00
style-guide.md Style guide: Attempt to make the L2 and L3 headings more visually distinct 2017-09-09 03:52:23 +00:00
typedefs.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
WATCHLISTS Add myself to the watchlist for webrtc/api/ and webrtc/base/ 2017-05-04 13:22:46 +00:00
webrtc.gni Remove QUIC transport/data channel 2017-10-24 16:14:18 +00:00
whitespace.txt Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info