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Harald Alvestrand 7af57c6e48 Remove RTP data implementation 
Bug: webrtc:6625
Change-Id: Ie68d7a938d8b7be95a01cca74a176104e4e44e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215321
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33759}
2021-04-16 13:10:54 +00:00
api Add Stable Writable Connection Ping Interval parameter to RTCConfiguration. 2021-04-16 07:11:10 +00:00
audio Speed up FrameCombiner::Combine by 3x 2021-04-13 17:18:47 +00:00
build_overrides Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492) 2021-04-12 18:25:58 +00:00
call Fix unsignalled ssrc race in WebRtcVideoChannel. 2021-04-16 09:33:42 +00:00
common_audio Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
common_video Provide a default implementation of NV12BufferInterface::CropAndScale. 2021-03-22 11:09:36 +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 Revert "Expose AV1 encoder&decoder from Android SDK." 2021-04-16 07:40:23 +00:00
g3doc Add video/g3doc/stats.md to the doc site menu 2021-04-16 11:23:43 +00:00
logging Remove use of istream in RTC event log parser. 2021-03-31 13:21:58 +00:00
media Remove RTP data implementation 2021-04-16 13:10:54 +00:00
modules Add conceptual docs for modules/video_coding 2021-04-16 08:46:12 +00:00
net/dcsctp dcsctp: Add operators on TimeMs and DurationMs 2021-04-14 09:21:15 +00:00
p2p Add Stable Writable Connection Ping Interval parameter to RTCConfiguration. 2021-04-16 07:11:10 +00:00
pc Remove RTP data implementation 2021-04-16 13:10:54 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Delete StreamAdapterInterface 2021-04-16 08:47:17 +00:00
rtc_tools Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium" 2021-02-25 10:48:55 +00:00
sdk Revert "Expose AV1 encoder&decoder from Android SDK." 2021-04-16 07:40:23 +00:00
stats Remove RTCRemoteInboundRtpStreamStats duplicate members. 2021-04-08 09:06:24 +00:00
style-guide Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
system_wrappers Consolidate the different NTP clocks into one. 2021-04-08 13:54:04 +00:00
test Remove RTP data implementation 2021-04-16 13:10:54 +00:00
tools_webrtc Change from sakal@webrtc.org to xalep@webrtc.org in OWNERS files. 2021-04-14 08:27:54 +00:00
video Add documentation for video/stats. 2021-04-16 09:18:42 +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 Configure GN to use python3 to exec_script. 2021-04-14 17:54:11 +00:00
.vpython Reland "Add protobuf-py2_py3 3.13.0 to .vpython." 2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md Polish the "Using Abseil in WebRTC" docs 2020-10-16 13:42:00 +00:00
AUTHORS Adds missing header to fix compilation error when compiling with use_custom_libcxx set to false. 2021-03-25 09:57:00 +00:00
BUILD.gn Move RTC_ENABLE_WIN_WGC define to the top level BUILD.gn 2021-04-08 16:31:49 +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 0bde1c5411..1a13f11499 (871876:872016) 2021-04-13 18:43:25 +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 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 Add titovartem@webrtc.org as owner for /g3doc 2021-04-12 13:40:47 +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 doc: move bug reporting instructions to the repository 2020-10-21 14:47:49 +00:00
style-guide.md Add deprecation section to webrtc style guide 2021-02-22 13:34:40 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni Remove unused a gn variable related to gtk 2021-04-16 06:29:20 +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 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