admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Adds test case that would have found potential dead-lock in pacer.

Browse Source 
https://webrtc-review.googlesource.com/c/src/+/178100 reverted a change
that could result in a deadlock if WebRTC-Audio-SendSideBwe was enabled
and WebRTC-Audio-ABWENoTWCC was not while using send-side BWE in a
mixed audio/video setting.

This CL adds an integration test that fails on tsan if above commit is
cherry-picked.

Bug: webrtc:10809
Change-Id: I5028d5794e5c9e970ccd9b7eb25d5b76a7fa4e58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178203
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31574}
This commit is contained in:
Erik Språng 2020-06-26 17:03:19 +02:00 committed by Commit Bot
parent efc55b0134
commit 000953c8d1
1 changed files with 45 additions and 0 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

45
test/scenario/scenario_unittest.cc
View File

@ -11,6 +11,7 @@
#include <atomic>
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/logging/memory_log_writer.h"
#include "test/scenario/stats_collection.h"
@ -141,5 +142,49 @@ TEST(ScenarioTest, WritesToRtcEventLog) {
EXPECT_GE(storage.logs().at("alice.rtc.dat").size(), 1u);
}
TEST(ScenarioTest,
RetransmitsVideoPacketsInAudioAndVideoCallWithSendSideBweAndLoss) {
// Make sure audio packets are included in transport feedback.
test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/WebRTC-Audio-ABWENoTWCC/Disabled/");
Scenario s;
CallClientConfig call_client_config;
call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
auto* alice = s.CreateClient("alice", call_client_config);
auto* bob = s.CreateClient("bob", call_client_config);
NetworkSimulationConfig network_config;
// Add some loss and delay.
network_config.delay = TimeDelta::Millis(200);
network_config.loss_rate = 0.05;
auto alice_net = s.CreateSimulationNode(network_config);
auto bob_net = s.CreateSimulationNode(network_config);
auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
// First add an audio stream, then a video stream.
// Needed to make sure audio RTP module is selected first when sending
// transport feedback message.
AudioStreamConfig audio_stream_config;
audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6);
audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64);
audio_stream_config.encoder.allocate_bitrate = true;
audio_stream_config.stream.in_bandwidth_estimation = true;
s.CreateAudioStream(route->forward(), audio_stream_config);
s.CreateAudioStream(route->reverse(), audio_stream_config);
VideoStreamConfig video_stream_config;
auto video = s.CreateVideoStream(route->forward(), video_stream_config);
s.CreateVideoStream(route->reverse(), video_stream_config);
// Run for 10 seconds.
s.RunFor(TimeDelta::Seconds(10));
// Make sure retransmissions have happened.
int retransmit_packets = 0;
for (const auto& substream : video->send()->GetStats().substreams) {
retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
}
EXPECT_GT(retransmit_packets, 0);
}
} // namespace test
} // namespace webrtc