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Benjamin Wright 1295b0def0 Add basic fuzzing for rtp_header_parser.h/cc. 
rtp_header_parser currently has 0% fuzzing coverage. To improve this I have
added a basic fuzzer which fuzzes all of the available paths.

Bug: webrtc:10395
Change-Id: I30324b2bfa7629b0110527258b33b7e048e89fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27115}
2019-03-13 23:31:16 +00:00
api
Reland "NetEQ RTP Play: Optionally write output audio file"
2019-03-13 15:33:29 +00:00
audio
Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly
2019-03-11 16:49:21 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Pass flexfec_sender only to the protected media send stream.
2019-03-12 14:18:08 +00:00
common_audio
webrtc: Remove semicolons.
2019-02-20 16:02:59 +00:00
common_video
Generic Frame Descriptor (GFD) VP8 templates.
2019-03-06 11:03:09 +00:00
crypto
Adding new top-level directory crypto/
2019-03-08 00:35:05 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Finally delete deprecated mac capturer.
2019-03-07 10:35:03 +00:00
logging
Add log based GoogCC simulation to visualizer.
2019-03-13 09:39:14 +00:00
media
Modifying MediaChannel to accept CopyOnWriteBuffer by value.
2019-03-12 23:49:57 +00:00
modules
Making UpdatesTargetRateBasedOnLinkCapacity more robust.
2019-03-13 15:36:55 +00:00
p2p
Implicitly suppress //build/config/clang:find_bad_constructs.
2019-03-01 10:18:17 +00:00
pc
Modifying MediaChannel to accept CopyOnWriteBuffer by value.
2019-03-12 23:49:57 +00:00
resources
2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883
2019-02-22 09:59:01 +00:00
rtc_base
Fixing documentation for CopyOnWriteBuffer.
2019-03-13 17:50:16 +00:00
rtc_tools
Add log based GoogCC simulation to visualizer.
2019-03-13 09:39:14 +00:00
sdk
Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly
2019-03-11 16:49:21 +00:00
stats
Expose relative packet arrival delay metric in stats API.
2019-03-06 16:35:16 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Add metrics::Samples to facilitate easier testing
2019-03-05 17:51:14 +00:00
test
Add basic fuzzing for rtp_header_parser.h/cc.
2019-03-13 23:31:16 +00:00
tools_webrtc
Bump iOS min supported version to 10.0
2019-03-07 13:08:17 +00:00
video
Preserve legacy behavior for old OveruseFrameDetector
2019-03-13 16:42:38 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Bump iOS min supported version to 10.0
2019-03-07 13:08:17 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Ban absl::StrSplit and absl::StrJoin
2019-02-26 00:45:11 +00:00
AUTHORS
Import proto_library.gni when rtc_enable_protobuf is true
2019-02-27 09:56:42 +00:00
BUILD.gn
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +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
common_types.h
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +00:00
DEPS
Roll chromium_revision a9ac2956aa..acb568e104 (640406:640514)
2019-03-13 23:26:06 +00:00
ENG_REVIEW_OWNERS
Enforce LGTM from owners of depends-on paths in DEPS via presubmit.
2018-09-28 12:49:54 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Clean up root OWNERS.
2018-11-09 14:23:59 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
Add a presubmit check for absl/memory/memory.h inclusion for WrapUnique
2019-02-28 14:12:48 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Modify pc/ WATCHLISTS definition
2019-01-31 22:09:16 +00:00
webrtc.gni
Remove legacy empty task_queue BUILD targets and build arg
2019-03-12 10:06:04 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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