deadbeef f5f03e823c Reland of: Improving the fake clock and using it to fix a flaky STUN timeout test.
When the fake clock's time is advanced, it now ensures all pending
queued messages have been dispatched. This allows us to write a
"SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
until the target time.

Useful in this case, where we know the STUN timeout should take a total
of 9500ms, but it would be overly complex to write test code that waits
for each individual timeout, ensures a STUN packet has been
retransmited, etc.

(The test described above *should* be written, but it belongs in
p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).

Review-Url: https://codereview.webrtc.org/2024813004
Cr-Commit-Position: refs/heads/master@{#13052}
2016-06-06 18:16:13 +00:00
2016-06-03 13:07:41 +00:00
2014-06-17 08:54:03 +00:00
2015-09-11 09:04:09 +00:00
2016-04-16 17:24:53 +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.

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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