Cleans up code related to legacy pre-pacing fec generation.

Bug: webrtc:11340
Change-Id: If3493db9fafdd3ad041f78999e304c8714be517f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186562
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32349}
This commit is contained in:
Erik Språng
2020-10-05 14:20:51 +02:00
committed by Commit Bot
parent 0d1b044db8
commit b6477858ac
9 changed files with 27 additions and 167 deletions

View File

@ -171,25 +171,6 @@ TEST(VideoStreamTest, SendsFecWithFlexFec) {
EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
}
TEST(VideoStreamTest, SendsFecWithDeferredFlexFec) {
ScopedFieldTrials trial("WebRTC-DeferredFecGeneration/Enabled/");
Scenario s;
auto route =
s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
{s.CreateSimulationNode([](NetworkSimulationConfig* c) {
c->loss_rate = 0.1;
c->delay = TimeDelta::Millis(100);
})},
s.CreateClient("callee", CallClientConfig()),
{s.CreateSimulationNode(NetworkSimulationConfig())});
auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
c->stream.use_flexfec = true;
});
s.RunFor(TimeDelta::Seconds(5));
VideoSendStream::Stats video_stats = video->send()->GetStats();
EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
}
TEST(VideoStreamTest, ResolutionAdaptsToAvailableBandwidth) {
// Declared before scenario to avoid use after free.
std::atomic<size_t> num_qvga_frames_(0);