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Philipp Hancke 491fa44ed9 openssl_stream_adapter: improve ssl handshake error logging 
BUG=webrtc:11817

Change-Id: Ia8a04779c028bd8071012211e4ac1cb1c424e759
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180621
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31861}
2020-08-05 21:17:26 +00:00
api Reland "Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader" 2020-08-04 12:17:06 +00:00
audio Remove unused critical section includes. 2020-07-16 13:52:28 +00:00
build_overrides set perfetto flag to default value of false 2020-07-22 10:14:53 +00:00
call Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 2020-08-03 15:45:41 +00:00
common_audio Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
common_video Ignore RTPFragmentationHeader when rewriting H264 SPS 2020-07-27 09:42:25 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Search and replace gendered terms according to style guide: 2020-06-12 14:12:54 +00:00
examples Implemented Android Demo Application for VoIP API 2020-07-21 16:34:22 +00:00
logging Search and replace gendered terms according to style guide: 2020-06-12 14:12:54 +00:00
media Reducing threshold for usrsctp "buffer low" callback. 2020-08-04 20:08:06 +00:00
modules Delete deprecated RTPSenderVideo::SendVideo function 2020-08-05 13:10:36 +00:00
p2p Reland "Implement packets_(sent | received) for RTCTransportStats" 2020-07-10 11:50:59 +00:00
pc sdp: reject sdp with malformed b= lines 2020-08-03 18:21:14 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base openssl_stream_adapter: improve ssl handshake error logging 2020-08-05 21:17:26 +00:00
rtc_tools Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 2020-08-03 15:45:41 +00:00
sdk Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder 2020-08-05 09:41:49 +00:00
stats Reland "Implement packets_(sent | received) for RTCTransportStats" 2020-07-10 11:50:59 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
test Add networks stats collector from PeerConnection GetStats API 2020-08-04 09:05:14 +00:00
tools_webrtc Fix "Assignment had no effect." error during chromium roll. 2020-07-30 12:57:20 +00:00
video Reland "Only enable conference mode simulcast allocations with flag enabled" 2020-08-04 10:30:08 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Reenable libaom decoder by default 2020-03-18 18:04:41 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Use absl_deps in order to preapre to the Abseil component build release. 2020-06-08 12:59:40 +00:00
AUTHORS Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder 2020-08-05 09:41:49 +00:00
BUILD.gn Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
common_types.h Replaces OverheadObserver with simple getter. 2020-05-07 17:33:45 +00:00
DEPS Roll chromium_revision ada5164b5b..ae72e95193 (794977:795090) 2020-08-05 18:32:51 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +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
native-api.md Make transient suppression optionally excludable via defines 2020-04-02 11:44:07 +00:00
OWNERS Remove phoglund as root owner. 2020-03-30 12:15:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md Fix link in documentation. (take 2) 2020-04-16 11:08:43 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Trigger CI bots. 2020-07-15 17:50:55 +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 here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info