Tomas Gunnarsson 191bf5c653 Revert "Reland "Adds support for multiple or no media stream ids.""
This reverts commit f351c3408a0c7f695447a2a9f4e6a1719a0d6a26.

Reason for revert: Breaks chromium import

https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/58012

Failin tests:
WebRtcRtpBrowserTest.TrackAddedToSecondStream
WebRtcRtpBrowserTest.TrackSwitchingStream

Original change's description:
> Reland "Adds support for multiple or no media stream ids."
> 
> This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
> 
> Original change's description:
> > Adds support for multiple or no media stream ids.
> > 
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> > 
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
> 
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org

Change-Id: I1835419f963762bc308a91d81c423d8e7bf65026
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7932, webrtc:7933
Reviewed-on: https://webrtc-review.googlesource.com/65700
Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22690}
2018-03-30 10:44:53 +00:00
2018-03-29 22:56:38 +00:00
2018-03-28 11:58:06 +00:00
.gn
2018-02-19 15:07:45 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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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