A new PeerConnection level perf test.
This test creates a one way audio and video call, allows for bandwidth
estimation to ramp up and then runs the call for 10 seconds. The
average bandwidth estimate over this time is recorded as a perf metric.
This is done at the PeerConnection level with the intention to catch
regressions related to ICE configurations. Stats are taken from
PeerConnection for BWE, and the network simulation is done with a
VirtualSocketServer.
Bug: webrtc:7668
Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db
Reviewed-on: https://webrtc-review.googlesource.com/78361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23758}diff --git a/BUILD.gn b/BUILD.gn
index 7566f2a..8cedb5b 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -546,13 +546,13 @@
rtc_test("webrtc_perf_tests") {
testonly = true
configs += [ ":rtc_unittests_config" ]
-
deps = [
"audio:audio_perf_tests",
"call:call_perf_tests",
"modules/audio_coding:audio_coding_perf_tests",
"modules/audio_processing:audio_processing_perf_tests",
"modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests",
+ "pc:peerconnection_perf_tests",
"test:test_main",
"video:video_full_stack_tests",
]