Stefan Holmer 1da4d79ba3 Move allocation and rtp conversion logic out of payload router.
Makes it easier to write tests, and allows for moving rtp module
ownership into the payload router in the future.

The RtpPayloadParams class is split into declaration and definition and
moved into separate files.

Bug: webrtc:9517
Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
Reviewed-on: https://webrtc-review.googlesource.com/88564
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23983}
2018-07-16 13:34:37 +00:00
2018-05-30 08:30:00 +00:00
2018-07-16 07:31:07 +00:00
2018-05-15 16:41:02 +00:00
.gn
2018-06-29 09:36:17 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2018-06-20 12:39:11 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2018-06-26 13:57:35 +00:00
2018-02-23 10:34:16 +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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