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Java 2.9%
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Victor Boivie 54e4e35c89 dcsctp: Add consistency check for assembled msgs 
The buffer of reassembled messages in ReassemblyQueue is only to be
used while processing a DATA/I-DATA or FORWARD-TSN as processing these
chunks may result in assembling messages.

When the socket is idle - between API calls - it's supposed to be empty.

Instead of having it as a member in ReassemblyQueue, it could be
provided as an argument to ReassemblyQueue::Add and
ReassemblyQueue::Handle(ForwardTSN), but that would be a quite big
refactoring. That will be investigated separately.

Bug: None
Change-Id: I41238de28f32f2a622c1d045debe3ea11e7c23f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232000
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35014}
2021-09-16 13:19:31 +00:00
api New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. 2021-09-15 09:57:29 +00:00
audio Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Reland "Reland "Enable WebRTC-Vp9DependencyDescriptor by default"" 2021-09-15 13:48:58 +00:00
common_audio Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
common_video Update h264 sps parsers and sps vui rewriter to use BitstreamReader 2021-09-16 10:48:41 +00:00
data
docs Update Mac prerequisites 2021-08-23 19:52:17 +00:00
examples Replace AV1X with AV1 2021-09-14 08:29:02 +00:00
g3doc Use absl instead of self-made function for low-level bit counting 2021-08-26 08:56:37 +00:00
logging Migrate rtc event log from rtc::BitBuffer to BitstreamReader 2021-09-07 14:03:27 +00:00
media Reland "Handle scalability mode in QueryCodecSupport" 2021-09-15 13:12:58 +00:00
modules Updae bitexactness tests to match new canonical results 2021-09-15 18:38:41 +00:00
net/dcsctp dcsctp: Add consistency check for assembled msgs 2021-09-16 13:19:31 +00:00
p2p Delete BasicPacketSocketFactory default constructor 2021-09-03 10:46:29 +00:00
pc Return proxied object in OnTransceiver 2021-09-16 09:40:52 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base GCC: fix template specialization in webrtc::BitstreamReader 2021-09-15 17:20:50 +00:00
rtc_tools Improve points visualization in metrics_plotter. 2021-09-10 10:59:58 +00:00
sdk Replace AV1X with AV1 2021-09-14 08:29:02 +00:00
stats Reland "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-06 14:26:55 +00:00
system_wrappers Use GTEST_SKIP() instead of early return. 2021-08-12 15:24:13 +00:00
test New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. 2021-09-15 09:57:29 +00:00
tools_webrtc Fix iOS sim bot to for the new Chromium roll 2021-09-14 16:16:33 +00:00
video New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. 2021-09-15 09:57:29 +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
.vpython3 Add .vpython3 file to webrtc 2021-09-15 16:56:30 +00:00
AUTHORS Fix potential crash during SimulcastEncoderAdapter tear down. 2021-09-14 09:15:22 +00:00
BUILD.gn Allow export of Obj-C symbols without C++ ones. 2021-07-30 22:54:59 +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 Updae bitexactness tests to match new canonical results 2021-09-15 18:38:41 +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 Add .vpython3 file to webrtc 2021-09-15 16:56:30 +00:00
PATENTS
PRESUBMIT.py fix some typos 2021-08-12 18:37:10 +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 Update WATCHLISTS 2021-08-23 13:37:55 +00:00
webrtc.gni Revert "Reland "PipeWire capturer: implement proper DMA-BUFs support""" 2021-09-03 11:28:26 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2021-09-08 18:30:03 +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