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3176ef79e9
Bug: webrtc:7484 Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36965}
58 lines
2 KiB
C++
58 lines
2 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio/test/audio_end_to_end_test.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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using NonSenderRttTest = CallTest;
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TEST_F(NonSenderRttTest, NonSenderRttStats) {
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class NonSenderRttTest : public AudioEndToEndTest {
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public:
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const int kTestDurationMs = 10000;
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const int64_t kRttMs = 30;
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BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const override {
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BuiltInNetworkBehaviorConfig pipe_config;
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pipe_config.queue_delay_ms = kRttMs / 2;
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return pipe_config;
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}
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void ModifyAudioConfigs(AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStreamInterface::Config>*
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receive_configs) override {
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ASSERT_EQ(receive_configs->size(), 1U);
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(*receive_configs)[0].enable_non_sender_rtt = true;
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AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs);
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send_config->send_codec_spec->enable_non_sender_rtt = true;
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}
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void PerformTest() override { SleepMs(kTestDurationMs); }
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void OnStreamsStopped() override {
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AudioReceiveStreamInterface::Stats recv_stats =
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receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
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EXPECT_GT(recv_stats.round_trip_time_measurements, 0);
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ASSERT_TRUE(recv_stats.round_trip_time.has_value());
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EXPECT_GT(recv_stats.round_trip_time->ms(), 0);
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EXPECT_GE(recv_stats.total_round_trip_time.ms(),
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recv_stats.round_trip_time->ms());
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}
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} test;
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RunBaseTest(&test);
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}
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} // namespace test
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} // namespace webrtc
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