Danil Chapovalov 97b59f060c Reduce RtpFrameReferenceFinder fuzzer input to more reasonable value
frame_id is unwraped from a 16bit value.
Getting to int64_t boundaries would take more than 2^48 packets.
That scenario considered unrealistic and thus untested.

Bug: chromium:1053482
Change-Id: Ib3f52d4528b20915b2330773f616d9304f45cac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168682
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30607}
2020-02-25 14:15:24 +00:00
2018-10-05 14:40:21 +00:00
2020-02-20 14:24:29 +00:00
2020-01-21 12:13:11 +00:00
2020-02-25 14:11:52 +00:00
2019-10-28 12:27:50 +00:00
2020-02-07 17:57:30 +00:00
2017-09-15 04:25:06 +00:00
2018-12-18 12:30:58 +00:00
2019-10-08 12:20:39 +00:00
2017-09-15 04:25:06 +00:00
2018-07-23 15:28:48 +00:00
2017-09-15 04:25:06 +00:00
2020-01-21 12:13:11 +00:00
2020-01-28 07:53:15 +00:00

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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%