tommi ae695e95a6 Refactor RtpSender and SSRCDatabase.
* SSRCDatabase doesn't need to be a global instance, so I've changed it to be a "regular" class (i.e. construct via ctor, not maybe via GetSSRCDatabase( + release via ReturnSSRCDatabase())).  If we ever have parallel tests running in the same process, they won't have the problem of using the same ssrc database.

* Made RtpSender a more const.  Also added some todos for myself and holmer to look into clarifying the threading model.

* Switched from CriticalSectionWrapper to rtc::CriticalSection

* Changed the random seeding to use TickTime::Now().Ticks() since TimeInMicroseconds() could return 0 when the process was starting.  This is what TimeInMicroseconds() does anyway but now we don't need to access a global clock object.

BUG=webrtc:3062

Review URL: https://codereview.webrtc.org/1623543002

Cr-Commit-Position: refs/heads/master@{#11462}
2016-02-02 16:34:16 +00:00
2014-06-17 08:54:03 +00:00
2015-09-11 09:04:09 +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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

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Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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