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Danil Chapovalov e09a174746 Fix ssl_certificate_fuzzer 
Bug: webrtc:10395
Change-Id: Iba79f257c427545c36052e74296d3c07a166ee7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225540
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34446}
2021-07-09 13:50:29 +00:00
api
Implement nack_count metric for outbound audio rtp streams.
2021-07-09 13:29:10 +00:00
audio
Implement nack_count metric for outbound audio rtp streams.
2021-07-09 13:29:10 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Implement nack_count metric for outbound audio rtp streams.
2021-07-09 13:29:10 +00:00
common_audio
Fix -Wunreachable-code-aggressive.
2021-06-30 11:14:37 +00:00
common_video
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
g3doc
Move supported_platforms.md under g3doc/.
2021-06-28 12:07:45 +00:00
logging
Replace PacketView by vector of pointers in a wrapper class.
2021-07-02 08:17:32 +00:00
media
Implement nack_count metric for outbound audio rtp streams.
2021-07-09 13:29:10 +00:00
modules
Correctly set number of reference buffers in H264 encoder
2021-07-09 13:49:41 +00:00
net/dcsctp
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
p2p
Reland "Port: migrate to TaskQueue."
2021-06-21 22:21:04 +00:00
pc
Implement nack_count metric for outbound audio rtp streams.
2021-07-09 13:29:10 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
rtc_tools
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
sdk
Remove backwards compatible build targets.
2021-07-08 20:26:49 +00:00
stats
Move WebRTC to non deprecated jsoncpp APIs.
2021-06-28 12:09:14 +00:00
system_wrappers
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
test
Fix ssl_certificate_fuzzer
2021-07-09 13:50:29 +00:00
tools_webrtc
Add ability to build XCFramework for iOS
2021-07-06 11:23:00 +00:00
video
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Increase iOS deployment target from 10 to 12.
2021-07-02 17:02:27 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
AUTHORS
Added PeerConnectionObserverJni::OnRemoveTrack()
2021-06-03 19:24:55 +00:00
BUILD.gn
Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering.
2021-06-23 09:56:20 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
2018-10-25 08:19:53 +00:00
DEPS
Roll chromium_revision 17c6abea92..a5d70b42f2 (899632:899910)
2021-07-09 09:02:25 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
PATENTS
…
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
Run pylint on presubmit only for modified python files.
2021-07-02 15:06:56 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
iOS: Get WebRTC building for Mac Catalyst
2021-07-06 21:07:59 +00:00
whitespace.txt
Revert "Reland "Trigger postsubmit tests.""
2021-06-28 19:44:42 +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

Description
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%