Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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