alessiob 8d23c050f2 MultiEndCall::CheckTiming() verifies that a set of audio tracks and timing information is valid to simulate conversational speech. Unordered turns are rejected. Self cross-talk and cross-talk with 3 or more speakers are not permitted since it would require mixing at the simulation step.
This CL includes extensive tests to match accept or reject decisions on several different timing setups. The setups are simulated using mocks (by far more light-weight than using actual timing and audio track files).

The client code, the unit tests in this case, passes information about the fake audio tracks to MockWavReaderFactory. MockWavReader instances are then created using the parameters defined in the client code. To improve the readability of the tests, generator_unittest.cc includes a docstring explaining how each MultiEndCallSetup* test is documented.

Run tests as follows:
$ out/Default/modules_unittests --gtest_filter=ConversationalSpeechTest.*

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2781573002
Cr-Commit-Position: refs/heads/master@{#17592}
2017-04-07 19:05:08 +00:00
2017-04-05 14:14:31 +00:00
2017-04-06 12:48:36 +00:00
2017-03-09 10:04:57 +00:00
2017-01-20 20:45:07 +00:00
2017-03-23 10:46:00 +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.

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