From 000953c8d126f3a65f7d6f8a0943e18456013a74 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Fri, 26 Jun 2020 17:03:19 +0200
Subject: [PATCH] Adds test case that would have found potential dead-lock in
 pacer.
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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}
---
 test/scenario/scenario_unittest.cc | 45 ++++++++++++++++++++++++++++++
 1 file changed, 45 insertions(+)

diff --git a/test/scenario/scenario_unittest.cc b/test/scenario/scenario_unittest.cc
index f74c1a5bdf..7c05ea39dd 100644
--- a/test/scenario/scenario_unittest.cc
+++ b/test/scenario/scenario_unittest.cc
@@ -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