Tommi 3ed36c0521 Reland "Remove stopped_ from AudioRtpReceiver and VideoRtpReceiver."
This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f

The difference from the original CL is that a check for
`state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
This caused a side effect that registering the sink while the source
was in an "initializing" state, failed. The last remaining state
however, is `kEnded` - but since there's no logic in the class around
the expected value of the states, the check inside of AddSink()
doesn't provide an additional value - it's rather a surprise for
developers if it doesn't succeed. So, now removed.

Original change's description:
> Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
>
> This simplifies the logic in these classes a bit, which makes upcoming
> change easier. The `stopped_` flag in these classes was essentially
> the same thing as `media_channel_ == nullptr`, which is what's
> consistently used now for the same checks.
>
> Bug: webrtc:13540
> Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35907}

Bug: webrtc:13540
Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35958}
2022-02-08 21:41:14 +00:00
2021-11-03 14:59:46 +00:00
2022-02-08 09:13:33 +00:00
2022-02-08 12:56:45 +00:00
2021-01-20 15:01:07 +00:00
2021-12-08 08:53:00 +00:00
2021-07-22 16:41:26 +00:00
2022-02-08 09:13:33 +00:00
2021-12-08 08:53:00 +00:00
2020-07-13 11:42:07 +00:00
2021-08-23 13:37:55 +00:00
2021-12-16 17:45:31 +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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