Taylor Brandstetter bd7392829a Revert "Reland "Remove our stream << overloads from non-test build targets.""
This reverts commit d7ee72041f882c023c73e27a7436c626c4e43604.

Reason for revert: Broke downstream build which was using SdpAudioFormat operator<<

Original change's description:
> Reland "Remove our stream << overloads from non-test build targets."
> 
> This is a reland of c841d18d257ba8e4ed7d77d105e3c46006bb1e7e
> 
> Original change's description:
> > Remove our stream << overloads from non-test build targets.
> >
> > Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
> > SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
> >
> > Bug: webrtc:8982
> > Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
> > Reviewed-on: https://webrtc-review.googlesource.com/64143
> > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> > Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22916}
> 
> TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org
> 
> Bug: webrtc:8982
> Change-Id: Ibe08c6270e5e693eb661a6ce9e8f074b34ef8123
> Reviewed-on: https://webrtc-review.googlesource.com/71161
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22949}

TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org

Change-Id: I3c2b18ec2877d68a522ecbae7a2955c4eecf36df
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8982
Reviewed-on: https://webrtc-review.googlesource.com/71446
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22963}
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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 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
Readme BSD-3-Clause 446 MiB
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