Evan Shrubsole 9f9671fe7f Revert "Reland "Ensure RTCRtpSenders are always created with one encoding""
This reverts commit fc5d627cef71f906e921476c2e6b1e01d07732fe.

Reason for revert: Breaks upstream WPT tests

Original change's description:
> Reland "Ensure RTCRtpSenders are always created with one encoding"
>
> This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
>
> Original change's description:
> > Ensure RTCRtpSenders are always created with one encoding
> >
> > It is possible to have AddTransceiver calls with an empty array
> > of encodings or AddTrack calls. In both cases, before negotiation,
> > the sender's encodings array would be empty and it was not possible
> > to update any value.
> >
> > Now, a default entry should be created in those cases, allowing to
> > update the configuration before negotiation.
> >
> > Bug: webrtc:10567
> > Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> > Auto-Submit: Florent Castelli <orphis@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39126}
>
> Bug: webrtc:10567
> Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39145}

Bug: webrtc:10567
Change-Id: If9b5adb5debb7c87a15662a8d0f232405a0e8136
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291221
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39147}
2023-01-19 14:02:26 +00:00
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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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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