henrik.lundin 5fac3f0892 NetEq: Don't check sample rate and frame size upon error
If an error happens in the GetAudio call, for instance when corrupt
payloads are inserted, GetAudio wil return an error. In this case, the
audio frame is not always correctly populated, which is to be expected.

BUG=webrtc:5447

Review-Url: https://codereview.webrtc.org/2272963002
Cr-Commit-Position: refs/heads/master@{#13902}
2016-08-24 18:18:54 +00:00
2016-08-24 06:17:32 +00:00
2016-08-11 14:01:03 +00:00
2016-08-03 17:11:35 +00:00
.gn
2016-08-24 13:55:05 +00:00
2016-06-14 09:39:40 +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.

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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