Åsa Persson a6e7b88198 Move rtp_timestamp_to_frame_num_ map from VideoProcessor to Stats class.
Let Stats class handle rtp timestamp to frame number mapping.

Bug: none
Change-Id: I2a29c89a25c75c4bbd6c6368a5d10514f90b3c42
Reviewed-on: https://webrtc-review.googlesource.com/41220
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21709}
2018-01-22 09:02:56 +00:00
2018-01-19 10:37:44 +00:00
2018-01-18 14:24:48 +00:00
.gn
2018-01-18 16:55:58 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +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.

Authoritative list of directories that contain the native API header files.

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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