diff options
Diffstat (limited to 'third_party/libwebrtc/audio/test/non_sender_rtt_test.cc')
-rw-r--r-- | third_party/libwebrtc/audio/test/non_sender_rtt_test.cc | 84 |
1 files changed, 84 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/test/non_sender_rtt_test.cc b/third_party/libwebrtc/audio/test/non_sender_rtt_test.cc new file mode 100644 index 0000000000..278193e335 --- /dev/null +++ b/third_party/libwebrtc/audio/test/non_sender_rtt_test.cc @@ -0,0 +1,84 @@ +/* + * 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 "rtc_base/gunit.h" +#include "rtc_base/task_queue_for_test.h" +#include "system_wrappers/include/sleep.h" +#include "test/gtest.h" + +namespace webrtc { +namespace test { + +using NonSenderRttTest = CallTest; + +TEST_F(NonSenderRttTest, NonSenderRttStats) { + class NonSenderRttTest : public AudioEndToEndTest { + public: + const int kLongTimeoutMs = 20000; + const int64_t kRttMs = 30; + + explicit NonSenderRttTest(TaskQueueBase* task_queue) + : task_queue_(task_queue) {} + + BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override { + BuiltInNetworkBehaviorConfig pipe_config; + pipe_config.queue_delay_ms = kRttMs / 2; + 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].enable_non_sender_rtt = true; + AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs); + send_config->send_codec_spec->enable_non_sender_rtt = true; + } + + void PerformTest() override { + // Wait until we have an RTT measurement, but no longer than + // `kLongTimeoutMs`. This usually takes around 5 seconds, but in rare + // cases it can take more than 10 seconds. + EXPECT_TRUE_WAIT(HasRoundTripTimeMeasurement(), kLongTimeoutMs); + } + + void OnStreamsStopped() override { + AudioReceiveStreamInterface::Stats recv_stats = + receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true); + EXPECT_GT(recv_stats.round_trip_time_measurements, 0); + ASSERT_TRUE(recv_stats.round_trip_time.has_value()); + EXPECT_GT(recv_stats.round_trip_time->ms(), 0); + EXPECT_GE(recv_stats.total_round_trip_time.ms(), + recv_stats.round_trip_time->ms()); + } + + protected: + bool HasRoundTripTimeMeasurement() { + bool has_rtt = false; + // GetStats() can only be called on `task_queue_`, block while we check. + SendTask(task_queue_, [this, &has_rtt]() { + if (receive_stream() && + receive_stream()->GetStats(true).round_trip_time_measurements > 0) { + has_rtt = true; + } + }); + return has_rtt; + } + + private: + TaskQueueBase* task_queue_; + } test(task_queue()); + + RunBaseTest(&test); +} + +} // namespace test +} // namespace webrtc |