mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-20 00:57:49 +01:00
0c126ed47a
It takes several seconds until we get an RTT measurement because that requires RTCP packets to be received and those are not sent very often. This CL makes the test faster on average by unblocking it as soon as we see an RTT measurement (as opposed to always blocking for 10 seconds), this usually unblocks after around 5 seconds. But to de-flake those rare instances where the test takes more than 10s to run, the maximum timeout is extended to 20 seconds. Patch Set 4: also fix use-of-uninitialized value. Bug: webrtc:14981 Change-Id: Ieca94c90dfb52c3b17584a06660ff66c6462aa8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296822 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Auto-Submit: Henrik Boström <hbos@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39531}
84 lines
2.8 KiB
C++
84 lines
2.8 KiB
C++
/*
|
|
* 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
|