This reverts commit 26d5e2e2809558148dc1e977ec1bc8318a2047bc.
Reverted originally because it dependend on a CL which was reverted. That CL has been reinstated in: https://chromium-review.googlesource.com/#/c/572070/
Bug: webrtc:7969
Change-Id: I404c3a42ad447312d981646dca0aa4cf0ec3134e
Reviewed-on: https://chromium-review.googlesource.com/572403
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19034}
Check that ice_regather_interval_range is set only when continual
regathering is also set.
Bug: webrtc:7969
Change-Id: Ifcfeee744d817cf00914418d7e682f11528faf05
Reviewed-on: https://chromium-review.googlesource.com/569358
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19009}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
Reason for revert:
Broken downstream project.
Original issue's description:
> Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
>
> BUG=webrtc:7395
>
> Review-Url: https://codereview.webrtc.org/2888303005
> Cr-Commit-Position: refs/heads/master@{#18417}
> Committed: 9641c13327TBR=deadbeef@webrtc.org,stefan@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,holmer@google.com,zstein@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2914413002
Cr-Commit-Position: refs/heads/master@{#18420}
Also renames "peerconnection_unittests" to "peerconnection_integrationtests",
and moves the ICE URL parsing code to separate files.
The main problem previously was that the test assertions
occurred in various places in the main test class, and this shared test
code was overly complex and stateful. As a result, it was difficult to
tell what a test even does, let alone what assertions it's meant to be
making. And writing a new test that does what you want can be a
frustrating ordeal.
The new code still uses helper methods, but they have intuitive names
and a smaller role; all of the important parts of the test's logic are
in the test case itself.
We're planning on merging PeerConnection and WebRtcSession at some point
soon, so it seemed valuable to do this, so that the WebRtcSession tests
can be rewritten as PeerConnection tests using better patterns.
BUG=None
Review-Url: https://codereview.webrtc.org/2738353003
Cr-Commit-Position: refs/heads/master@{#17458}
This will later allow calling the "PeerConnectionObserver::OnIceCandidate"
method asynchronously while keeping the object alive.
BUG=webrtc:3721
Review-Url: https://codereview.webrtc.org/2748253003
Cr-Commit-Position: refs/heads/master@{#17380}
for consistency with the WebRTC 1.0 standard as suggested in a TODO.
BUG=None
Review-Url: https://codereview.webrtc.org/2732663004
Cr-Commit-Position: refs/heads/master@{#17077}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
