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authorErik Språng <sprang@webrtc.org>2020-07-02 12:02:36 +0000
committerCommit Bot <commit-bot@chromium.org>2020-07-02 12:03:07 +0000
commita1888ae791b5f4e4cdb01acf72e378a33908aeab (patch)
tree6a731c1e899ba2f7e1d3799b3b9f1d37e0ba7f4a /test
parent19df870d924662e3b6efb86078d31a8e086b38b5 (diff)
downloadwebrtc-a1888ae791b5f4e4cdb01acf72e378a33908aeab.tar.gz
Revert "Reland "Allows FEC generation after pacer step.""
This reverts commit 19df870d924662e3b6efb86078d31a8e086b38b5. Reason for revert: Downstream project failure Original change's description: > Reland "Allows FEC generation after pacer step." > > This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0 > > Patchset 2 contains a fix. Old code can in factor call > RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec > is not supported there - we shouldn't crash. > > Original change's description: > > Allows FEC generation after pacer step. > > > > Split out from https://webrtc-review.googlesource.com/c/src/+/173708 > > This CL enables FEC packets to be generated as media packets are sent, > > rather than generated, i.e. media packets are inserted into the fec > > generator after the pacing stage rather than at packetization time. > > > > This may have some small impact of performance. FEC packets are > > typically only generated when a new packet with a marker bit is added, > > which means FEC packets protecting a frame will now be sent after all > > of the media packets, rather than (potentially) interleaved with them. > > Therefore this feature is currently behind a flag so we can examine the > > impact. Once we are comfortable with the behavior we'll make it default > > and remove the old code. > > > > Note that this change does not include the "protect all header > > extensions" part of the original CL - that will be a follow-up. > > > > Bug: webrtc:11340 > > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760 > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#31558} > > Bug: webrtc:11340 > Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984 > Commit-Queue: Erik Språng <sprang@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31613} TBR=sprang@webrtc.org,srte@webrtc.org Change-Id: I3b2b25898ce88b64c2322f68ef83f9f86ac2edb0 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11340 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178563 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31614}
Diffstat (limited to 'test')
-rw-r--r--test/scenario/video_stream_unittest.cc20
1 files changed, 0 insertions, 20 deletions
diff --git a/test/scenario/video_stream_unittest.cc b/test/scenario/video_stream_unittest.cc
index 873ef639ba..37afe1b1e7 100644
--- a/test/scenario/video_stream_unittest.cc
+++ b/test/scenario/video_stream_unittest.cc
@@ -9,7 +9,6 @@
*/
#include <atomic>
-#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/scenario/scenario.h"
@@ -171,25 +170,6 @@ 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);