nisse c8ee882753 Replace use of ASSERT in test code.
In top level test functions, replaced with gtest ASSERT_*. In helper
methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a
case-by-case basis.

In separate mock/fake classes used by tests (which might be of some
use also in tests of third-party applications), ASSERT was replaced
with RTC_CHECK, using

  git grep -l ' ASSERT(' | grep -v common.h | \
    xargs sed -i 's/ ASSERT(/ RTC_CHECK(/'

followed by additional includes of base/checks.h in affected files,
and git cl format.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2622413005
Cr-Commit-Position: refs/heads/master@{#16150}
2017-01-18 15:20:55 +00:00
2016-06-14 09:39:40 +00:00
2015-09-11 09:04:09 +00:00
2016-11-23 16:42:57 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%