Taylor Brandstetter d65ae4a8ba Fixing DCHECK in turnport.cc and doing some related cleanup.
Namely:
* Changing destruction_timestamp_ to rtc::Optional, instead of using 0
  as a magic value.
* Adding some comments.
* Adding a log statement that would have helped debugging the issue
  that hit this DCHECK.
* Getting rid of a 2-line method called in one place, which was not
  really helping code readability.

Bug: None
Change-Id: I5fb1ce60edea29cab0c2a8c97e735f26c08aba62
Reviewed-on: https://webrtc-review.googlesource.com/7440
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20196}
2017-10-08 19:20:37 +00:00
2017-10-05 06:59:15 +00:00
2017-10-06 13:58:14 +00:00
2017-10-06 11:20:14 +00:00
.gn
2017-09-25 15:34:41 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +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.

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%