Alessio Bazzica c366d51836 Fix unit for inbound RTP stat lastPacketReceivedTimestamp (s -> ms)
Both inbound RTP stats `estimatedPlayoutTimestamp` and
`lastPacketReceivedTimestamp` are surfaced to JS land as
`DOMHighResTimeStamp` - i.e., time values in milliseconds.
This CL fixes `lastPacketReceivedTimestamp` which is incorrectly
surfaced as time value in seconds.

Bug: webrtc:12605
Change-Id: I290103071cca3331d2a3066b6b6b9fcb4f4fd0af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212742
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33530}
2021-03-22 18:47:33 +00:00
2018-10-05 14:40:21 +00:00
2021-03-01 16:16:03 +00:00
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2021-03-22 12:56:26 +00:00
2021-03-22 09:41:27 +00:00
2021-03-22 15:48:52 +00:00
2021-01-20 15:01:07 +00:00
2020-07-13 11:42:07 +00:00
2021-03-22 11:57:23 +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.

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