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/*
* 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 <cstdint>
#include <vector>
#include "api/task_queue/task_queue_base.h"
#include "api/test/rtc_error_matchers.h"
#include "api/test/simulated_network.h"
#include "api/units/time_delta.h"
#include "audio/test/audio_end_to_end_test.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/call_test.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/wait_until.h"
namespace webrtc {
namespace test {
using ::testing::IsTrue;
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_THAT(
WaitUntil([&] { return HasRoundTripTimeMeasurement(); }, IsTrue(),
{.timeout = TimeDelta::Millis(kLongTimeoutMs)}),
IsRtcOk());
}
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
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