Danil Chapovalov 95cb56bd89 Add extra input validation to RtpFrameReferenceFinder for codec-specific cases
wrap ids before unwrapping: should be noop for ids arrived from the
network, but avoids DCHECKs for ids arrived from fuzzer.

for vp9 double check number of references doesn't exceed maximum.
for vp8 drop key frames for non-zero temporal id.
for general by seqnum code path do not set last_picture_id_:
it is not used there, but may confuse vp8 codepath.

as a slight speed up avoid copying RTPVideoTypeHeader for vp8 and vp9.

Bug: chromium:1046995, chromium:1047024, chromium:1047095, chromium:1047165, chromium:1047190
Change-Id: I1ab0833d32e2c023cbf5e3cfcc9e74f1c558e44b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168040
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30426}
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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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