Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 59 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/test/nack_test.cc b/third_party/libwebrtc/audio/test/nack_test.cc
new file mode 100644
index 0000000000..c4bfe8306f
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/nack_test.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/test/audio_end_to_end_test.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+using NackTest = CallTest;
+
+TEST_F(NackTest, ShouldNackInLossyNetwork) {
+  class NackTest : public AudioEndToEndTest {
+   public:
+    const int kTestDurationMs = 2000;
+    const int64_t kRttMs = 30;
+    const int64_t kLossPercent = 30;
+    const int kNackHistoryMs = 1000;
+
+    BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
+      BuiltInNetworkBehaviorConfig pipe_config;
+      pipe_config.queue_delay_ms = kRttMs / 2;
+      pipe_config.loss_percent = kLossPercent;
+      return pipe_config;
+    }
+
+    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                            std::vector<AudioReceiveStreamInterface::Config>*
+                                receive_configs) override {
+      ASSERT_EQ(receive_configs->size(), 1U);
+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackHistoryMs;
+      AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs);
+    }
+
+    void PerformTest() override { SleepMs(kTestDurationMs); }
+
+    void OnStreamsStopped() override {
+      AudioReceiveStreamInterface::Stats recv_stats =
+          receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
+      EXPECT_GT(recv_stats.nacks_sent, 0U);
+      AudioSendStream::Stats send_stats = send_stream()->GetStats();
+      EXPECT_GT(send_stats.retransmitted_packets_sent, 0U);
+      EXPECT_GT(send_stats.nacks_rcvd, 0U);
+    }
+  } test;
+
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc