webrtc/audio/test/nack_test.cc
Philipp Hancke 6a7bf10d60 Replace "rcvd" with "received" for readability
following guidance in
  https://google.github.io/styleguide/cppguide.html#General_Naming_Rules

BUG=None

Change-Id: I105591a7f709d65a3d3320f7f44137d432cf7ce0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302282
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39937}
2023-04-24 15:30:07 +00:00

59 lines
2 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 "system_wrappers/include/sleep.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
using NackTest = CallTest;
TEST_F(NackTest, ShouldNackInLossyNetwork) {
class NackTest : public AudioEndToEndTest {
public:
const int kTestDurationMs = 2000;
const int64_t kRttMs = 30;
const int64_t kLossPercent = 30;
const int kNackHistoryMs = 1000;
BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
BuiltInNetworkBehaviorConfig pipe_config;
pipe_config.queue_delay_ms = kRttMs / 2;
pipe_config.loss_percent = kLossPercent;
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].rtp.nack.rtp_history_ms = kNackHistoryMs;
AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs);
}
void PerformTest() override { SleepMs(kTestDurationMs); }
void OnStreamsStopped() override {
AudioReceiveStreamInterface::Stats recv_stats =
receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
EXPECT_GT(recv_stats.nacks_sent, 0U);
AudioSendStream::Stats send_stats = send_stream()->GetStats();
EXPECT_GT(send_stats.retransmitted_packets_sent, 0U);
EXPECT_GT(send_stats.nacks_received, 0U);
}
} test;
RunBaseTest(&test);
}
} // namespace test
} // namespace webrtc