These tests are already built into rtc_unittests, so they end up being
run three times. Fixed by creating a "p2p_test_utils" target that
contains the test utils that ortc_unittests and rtc_media_unittests
depend on, but not the tests themselves.
BUG=None
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2820263004
Cr-Commit-Position: refs/heads/master@{#17752}
Reason for revert:
Breaks checkdeps rules. Need to make a "p2p_test_utils" build target to include things like fakeicetransport.h.
Original issue's description:
> Remove rtc_p2p_unittests from ortc_unittests executable.
>
> These tests are already built into rtc_unittests; they shouldn't be
> built into two test executables.
>
> BUG=None
> TBR=kjellander@webrtc.org
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2820263004
> Cr-Commit-Position: refs/heads/master@{#17748}
> Committed: fe9d38f515TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2826703002
Cr-Commit-Position: refs/heads/master@{#17749}
The function converts the cricket::Codec, cricket::RtpHeaderExtensions
and cricket::StreamParamsVec to webrtc::RtpParameters.
BUG=webrtc:7311
Review-Url: https://codereview.webrtc.org/2735853004
Cr-Commit-Position: refs/heads/master@{#17124}
Introduce new target //webrtc/p2p:rtc_p2p_test_utils to host
test-related utilities.
Previously uncovered header "base/fakecandidatepair.h" is now also in a target.
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2714263004
Cr-Commit-Position: refs/heads/master@{#17036}
Create the SrtpTransportInterface, a subclass of RtpTransportInterface, which
allows the user to set the send and receive keys. The functionalities are
implemented inside the RtpTransportAdapters on top of BaseChannel.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2714813004
Cr-Commit-Position: refs/heads/master@{#17023}
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}
