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philipel 355c47309d Fix VideoRtpDepacketizerVp{8,9} copy assignment signature. 
Bug: none
Change-Id: I4adca8b4cbf4ffa15172fabc1eaba8c2b65c6fb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222650
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34306}
2021-06-16 17:09:05 +00:00
api Sanitize hostname literals when mDNS obfuscation is on. 2021-06-15 14:41:46 +00:00
audio Don't recreate the audio receive stream when updating the local_ssrc. 2021-06-16 10:03:31 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Don't recreate the audio receive stream when updating the local_ssrc. 2021-06-16 10:03:31 +00:00
common_audio Avoid undefined behavior in a division operation. 2021-04-23 07:49:24 +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 Added PeerConnectionObserverJni::OnRemoveTrack() 2021-06-03 19:24:55 +00:00
g3doc doc: document rtp payload type mapping behaviour 2021-06-04 06:23:32 +00:00
logging Add documentation for RTC event log 2021-06-03 09:03:18 +00:00
media Don't recreate the audio receive stream when updating the local_ssrc. 2021-06-16 10:03:31 +00:00
modules Fix VideoRtpDepacketizerVp{8,9} copy assignment signature. 2021-06-16 17:09:05 +00:00
net/dcsctp dcsctp: Do explicit bounds checking in bounded IO 2021-06-16 13:02:32 +00:00
p2p Sanitize hostname literals when mDNS obfuscation is on. 2021-06-15 14:41:46 +00:00
pc Modify Bundle logic to not add & destroy extra transport at add-track 2021-06-15 09:44:36 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base LOG DTLS (failed) handshake retransmission 2021-06-16 13:13:52 +00:00
rtc_tools Reland "Correctly handle retransmissions/padding in early loss detection." 2021-06-16 08:14:27 +00:00
sdk Remove the createDecoder(String) overload 2021-06-16 12:31:52 +00:00
stats Populate qualityLimitationDurations stats for outbound RTP streams 2021-05-31 21:39:37 +00:00
system_wrappers Make Clock::ConvertTimestampToNtpTime pure virtual 2021-05-21 09:55:14 +00:00
test Factor out common receive stream methods to a common interface. 2021-06-14 16:54:07 +00:00
tools_webrtc Add MB configs for M1 bots 2021-06-09 19:02:28 +00:00
video Add rtp_config() accessor to ReceiveStream. 2021-06-14 17:57:57 +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 Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +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 Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +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
DEPS Roll chromium_revision 570a173256..b452ca696d (892156:892948) 2021-06-16 10:50:17 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py sctp: Rename SctpTransport to UsrSctpTransport 2021-04-12 10:40:34 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +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.gni Turn on the RTC_ENABLE_WIN_WGC build flag. 2021-05-10 20:16:52 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Reland "Triggering CI." 2021-03-22 11:57:23 +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