This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.
Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
The decision to route audio packets to a separate overuse detector
is off by default and requires the field trial
WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/
The parameters control the threshold for switching over to the
audio overuse detector if we stop receiving feedback for video.
Bug: webrtc:10932
Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30694}
This makes it easier to maintain consistency between real time
and simulated time modes.
The RealTimeController is updated to use an explicit main thread,
this ensures that pending destruction tasks are run as the network
emulator goes out of scope.
Bug: webrtc:11255
Change-Id: Ie73ab778c78a68d7c58c0f857f14a8d8ac027c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166164
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30342}
This is a reland of b70c5c5ce97e7dcf2e1d8453f5ea0639d4b60453
Original change's description:
> Using simulated rtc::Thread for peer connection scenario tests.
>
> Bug: webrtc:11255
> Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30258}
Bug: webrtc:11255
Change-Id: If65cd56b59158cebec5609407a721fbdb47cfd1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30294}
This CL contains some preparatory cleanup that can be done
outside the main CL.
Bug: webrtc:11255
Change-Id: Ib0dcd81d352bafc446dcd2f7f82ba81f5e82e210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165766
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30247}
Based on similar code in the call level scenario test framework.
Bug: webrtc:10839
Change-Id: I262a890aa2cf905bb81b0f07957c08d0df5f7651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29361}
This is a reland of ad5c4accad00e04de08e2b62d366cc1f8e0320a5
It was flaky due to starting ICE signaling before SDP negotiation
finished. This was solved by adding an helper for adding ice candidates
which will wait until the peer connection is ready if needed.
Original change's description:
> Adds PeerConnection scenario test framework.
>
> Bug: webrtc:10839
> Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28754}
Bug: webrtc:10839
Change-Id: I6eb8f482561c87e7b0f20d2431d21a41b26c91d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147877
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28777}
Bug: webrtc:10839
Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28754}
