Paul Hallak 95f1e5192c Do not attempt setting the absolute capture time extension if we don't
get a timestamp.

Also, use -1 to signify an unset timestamp. This is what other callers
do: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/acm2/audio_coding_module.cc;l=306;drc=bbe4aed2302dc763935496b80a5cefb6a42d912d

No-Try because of lack of infra lack of capacity on macs.

No-Try: True
Bug: webrtc:11327
Change-Id: Ide0c0633579b6b2be3eea9912b13f858760de0ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219781
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34074}
2021-05-21 10:55:15 +00:00
2018-10-05 14:40:21 +00:00
2021-05-19 12:57:03 +00:00
2021-01-20 15:01:07 +00:00
2021-04-26 16:39:07 +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.

More info

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