815c5c0179
Bug: webrtc:381524905 Change-Id: If153ae0f284873a35cfce7d72e3dcba41f768190 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373741 Commit-Queue: Per Åhgren <peah@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43717}
99 lines
3.3 KiB
C++
99 lines
3.3 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 <cstdint>
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#include <vector>
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#include "api/task_queue/task_queue_base.h"
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#include "api/test/rtc_error_matchers.h"
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#include "api/test/simulated_network.h"
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#include "api/units/time_delta.h"
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#include "audio/test/audio_end_to_end_test.h"
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#include "call/audio_receive_stream.h"
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#include "call/audio_send_stream.h"
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#include "rtc_base/task_queue_for_test.h"
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#include "test/call_test.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "test/wait_until.h"
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namespace webrtc {
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namespace test {
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using ::testing::IsTrue;
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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 kLongTimeoutMs = 20000;
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const int64_t kRttMs = 30;
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explicit NonSenderRttTest(TaskQueueBase* task_queue)
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: task_queue_(task_queue) {}
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BuiltInNetworkBehaviorConfig GetSendTransportConfig() 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 {
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// Wait until we have an RTT measurement, but no longer than
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// `kLongTimeoutMs`. This usually takes around 5 seconds, but in rare
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// cases it can take more than 10 seconds.
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EXPECT_THAT(
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WaitUntil([&] { return HasRoundTripTimeMeasurement(); }, IsTrue(),
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{.timeout = webrtc::TimeDelta::Millis(kLongTimeoutMs)}),
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IsRtcOk());
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}
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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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protected:
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bool HasRoundTripTimeMeasurement() {
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bool has_rtt = false;
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// GetStats() can only be called on `task_queue_`, block while we check.
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SendTask(task_queue_, [this, &has_rtt]() {
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if (receive_stream() &&
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receive_stream()->GetStats(true).round_trip_time_measurements > 0) {
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has_rtt = true;
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}
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});
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return has_rtt;
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}
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private:
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TaskQueueBase* task_queue_;
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} test(task_queue());
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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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