Alessio Bazzica bc1c93dc6e Add remote-outbound stats for audio streams
Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats`
via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams.

`GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and
`ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`.

Bug: webrtc:12529
Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33452}
2021-03-12 20:39:50 +00:00
2021-03-01 16:16:03 +00:00
2021-03-09 09:58:58 +00:00
2021-01-20 15:01:07 +00:00
2020-07-13 11:42:07 +00:00
2021-02-14 19:14:44 +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 here 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

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