Niels Möller 827cf3c1b2 Avoid overflow when parsing H.264 SPS.
Check that |log2_max_frame_num_minus4| and
|log2_max_pic_order_cnt_lsb_minus4| are at most 28, resulting in a
field width of at most 32 bits.

Bug: chromium:877843
Change-Id: I684f92b8f0f2fcdbab24732d8e8381bc51a92752
Reviewed-on: https://webrtc-review.googlesource.com/101760
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24862}
2018-09-27 11:50:10 +00:00
2018-09-27 10:00:40 +00:00
.gn
2018-08-13 13:54:05 +00:00
2018-07-23 15:28:48 +00:00
2018-09-04 11:14:07 +00:00
2018-07-23 15:28:48 +00:00
2018-02-23 10:34:16 +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
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