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Markus Handell 06a2bf09a4 NackModule2: Rename to NackRequester. 
The alternative new name proposed, NackTracker, is already in
use in audio_coding.

Fixed: webrtc:11594
Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34539}
2021-07-23 08:30:33 +00:00
api Update links to point at main branch 2021-07-22 16:41:26 +00:00
audio Adding packetsDiscarded to RTCReceivedRtpStreamStats. 2021-07-13 20:34:45 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Update WebRTC code version (2021-07-23T04:03:53). 2021-07-23 05:49:17 +00:00
common_audio Fix -Wunreachable-code-aggressive. 2021-06-30 11:14:37 +00:00
common_video Ignore prefix NAL unit for slice parsing. 2021-07-22 06:25:53 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update links to point at main branch 2021-07-22 16:41:26 +00:00
examples Replace assert() with RTC_DCHECK(). 2021-07-09 07:49:43 +00:00
g3doc Update links to point at main branch 2021-07-22 16:41:26 +00:00
logging Update links to point at main branch 2021-07-22 16:41:26 +00:00
media Move helpers to parse base rtp packet fields to rtp_rtcp module 2021-07-19 14:27:27 +00:00
modules NackModule2: Rename to NackRequester. 2021-07-23 08:30:33 +00:00
net/dcsctp Update links to point at main branch 2021-07-22 16:41:26 +00:00
p2p [sigslot] - Remove signal from StunPort::AddressResolver. 2021-07-22 19:35:56 +00:00
pc Update links to point at main branch 2021-07-22 16:41:26 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Remove all #include <assert.h>/<cassert> and usage in Obj-C code. 2021-07-22 14:00:26 +00:00
rtc_tools Replace assert() with RTC_DCHECK(). 2021-07-09 07:49:43 +00:00
sdk Hide AndroidVideoBuffer class and use factory function 2021-07-16 13:54:45 +00:00
stats Adding packetsDiscarded to RTCReceivedRtpStreamStats. 2021-07-13 20:34:45 +00:00
system_wrappers Replace assert() with RTC_DCHECK(). 2021-07-09 07:49:43 +00:00
test Update links to point at main branch 2021-07-22 16:41:26 +00:00
tools_webrtc Reland "Add WebRTC specific dcheck_always_on." 2021-07-21 13:26:14 +00:00
video NackModule2: Rename to NackRequester. 2021-07-23 08:30:33 +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 links to point at main branch 2021-07-22 16:41:26 +00:00
AUTHORS Added PeerConnectionObserverJni::OnRemoveTrack() 2021-06-03 19:24:55 +00:00
BUILD.gn Reland "Add WebRTC specific dcheck_always_on." 2021-07-21 13:26:14 +00:00
CODE_OF_CONDUCT.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +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 89e5d40511..03a29cf406 (904223:904346) 2021-07-22 17:25:16 +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 Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
PATENTS
PRESUBMIT.py Fix PRESUBMIT.py to not run pylint on deleted files. 2021-07-23 06:45:57 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +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 doc: add g3doc sitemap to toplevel readme 2021-07-23 07:55:17 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni Reland "Add WebRTC specific dcheck_always_on." 2021-07-21 13:26:14 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +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