Add tests for inital probing and mid-call probing by reconfiguring max bitrate.
BUG=none
Review-Url: https://codereview.webrtc.org/2760623002
Cr-Commit-Position: refs/heads/master@{#17316}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
The WebRTC valgrind wrapper scripts needs to be moved in order to
unlock us from depending on a Chromium checkout.
BUG=webrtc:5006
TBR=ehmaldonado@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2578093002
Cr-Commit-Position: refs/heads/master@{#15662}
