diff options
Diffstat (limited to 'third_party/libwebrtc/audio/test/nack_test.cc')
-rw-r--r-- | third_party/libwebrtc/audio/test/nack_test.cc | 59 |
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..b36adf8991 --- /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_received, 0U); + } + } test; + + RunBaseTest(&test); +} + +} // namespace test +} // namespace webrtc |