tor-browser

The Tor Browser
git clone https://git.dasho.dev/tor-browser.git
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non_sender_rtt_test.cc (3325B)

      1 /*
      2 *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
      3 *
      4 *  Use of this source code is governed by a BSD-style license
      5 *  that can be found in the LICENSE file in the root of the source
      6 *  tree. An additional intellectual property rights grant can be found
      7 *  in the file PATENTS.  All contributing project authors may
      8 *  be found in the AUTHORS file in the root of the source tree.
      9 */
     10 
     11 #include <cstdint>
     12 #include <vector>
     13 
     14 #include "api/task_queue/task_queue_base.h"
     15 #include "api/test/rtc_error_matchers.h"
     16 #include "api/test/simulated_network.h"
     17 #include "api/units/time_delta.h"
     18 #include "audio/test/audio_end_to_end_test.h"
     19 #include "call/audio_receive_stream.h"
     20 #include "call/audio_send_stream.h"
     21 #include "rtc_base/task_queue_for_test.h"
     22 #include "test/call_test.h"
     23 #include "test/gmock.h"
     24 #include "test/gtest.h"
     25 #include "test/wait_until.h"
     26 
     27 namespace webrtc {
     28 namespace test {
     29 
     30 using ::testing::IsTrue;
     31 using NonSenderRttTest = CallTest;
     32 
     33 TEST_F(NonSenderRttTest, NonSenderRttStats) {
     34  class NonSenderRttTest : public AudioEndToEndTest {
     35   public:
     36    const int kLongTimeoutMs = 20000;
     37    const int64_t kRttMs = 30;
     38 
     39    explicit NonSenderRttTest(TaskQueueBase* task_queue)
     40        : task_queue_(task_queue) {}
     41 
     42    BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
     43      BuiltInNetworkBehaviorConfig pipe_config;
     44      pipe_config.queue_delay_ms = kRttMs / 2;
     45      return pipe_config;
     46    }
     47 
     48    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
     49                            std::vector<AudioReceiveStreamInterface::Config>*
     50                                receive_configs) override {
     51      ASSERT_EQ(receive_configs->size(), 1U);
     52      (*receive_configs)[0].enable_non_sender_rtt = true;
     53      AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs);
     54      send_config->send_codec_spec->enable_non_sender_rtt = true;
     55    }
     56 
     57    void PerformTest() override {
     58      // Wait until we have an RTT measurement, but no longer than
     59      // `kLongTimeoutMs`. This usually takes around 5 seconds, but in rare
     60      // cases it can take more than 10 seconds.
     61      EXPECT_THAT(
     62          WaitUntil([&] { return HasRoundTripTimeMeasurement(); }, IsTrue(),
     63                    {.timeout = TimeDelta::Millis(kLongTimeoutMs)}),
     64          IsRtcOk());
     65    }
     66 
     67    void OnStreamsStopped() override {
     68      AudioReceiveStreamInterface::Stats recv_stats =
     69          receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
     70      EXPECT_GT(recv_stats.round_trip_time_measurements, 0);
     71      ASSERT_TRUE(recv_stats.round_trip_time.has_value());
     72      EXPECT_GT(recv_stats.round_trip_time->ms(), 0);
     73      EXPECT_GE(recv_stats.total_round_trip_time.ms(),
     74                recv_stats.round_trip_time->ms());
     75    }
     76 
     77   protected:
     78    bool HasRoundTripTimeMeasurement() {
     79      bool has_rtt = false;
     80      // GetStats() can only be called on `task_queue_`, block while we check.
     81      SendTask(task_queue_, [this, &has_rtt]() {
     82        if (receive_stream() &&
     83            receive_stream()->GetStats(true).round_trip_time_measurements > 0) {
     84          has_rtt = true;
     85        }
     86      });
     87      return has_rtt;
     88    }
     89 
     90   private:
     91    TaskQueueBase* task_queue_;
     92  } test(task_queue());
     93 
     94  RunBaseTest(&test);
     95 }
     96 
     97 }  // namespace test
     98 }  // namespace webrtc