blob: e23467e71cd244bd933794a366354b5189063e69 [file] [log] [blame]
/*
* Copyright (c) 2016 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 "video/send_delay_stats.h"
#include <cstdint>
#include <memory>
#include <vector>
#include "call/rtp_config.h"
#include "system_wrappers/include/metrics.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
const uint32_t kSsrc1 = 17;
const uint32_t kSsrc2 = 42;
const uint32_t kRtxSsrc1 = 18;
const uint32_t kRtxSsrc2 = 43;
const uint16_t kPacketId = 2345;
const TimeDelta kMaxPacketDelay = TimeDelta::Seconds(11);
const int kMinRequiredPeriodicSamples = 5;
const int kProcessIntervalMs = 2000;
} // namespace
class SendDelayStatsTest : public ::testing::Test {
public:
SendDelayStatsTest() : clock_(1234), config_(CreateConfig()) {}
virtual ~SendDelayStatsTest() {}
protected:
virtual void SetUp() {
stats_.reset(new SendDelayStats(&clock_));
stats_->AddSsrcs(config_);
}
VideoSendStream::Config CreateConfig() {
VideoSendStream::Config config(nullptr);
config.rtp.ssrcs.push_back(kSsrc1);
config.rtp.ssrcs.push_back(kSsrc2);
config.rtp.rtx.ssrcs.push_back(kRtxSsrc1);
config.rtp.rtx.ssrcs.push_back(kRtxSsrc2);
return config;
}
void OnSendPacket(uint16_t id, uint32_t ssrc) {
OnSendPacket(id, ssrc, clock_.CurrentTime());
}
void OnSendPacket(uint16_t id, uint32_t ssrc, Timestamp capture) {
stats_->OnSendPacket(id, capture, ssrc);
}
bool OnSentPacket(uint16_t id) {
return stats_->OnSentPacket(id, clock_.CurrentTime());
}
SimulatedClock clock_;
VideoSendStream::Config config_;
std::unique_ptr<SendDelayStats> stats_;
};
TEST_F(SendDelayStatsTest, SentPacketFound) {
EXPECT_FALSE(OnSentPacket(kPacketId));
OnSendPacket(kPacketId, kSsrc1);
EXPECT_TRUE(OnSentPacket(kPacketId)); // Packet found.
EXPECT_FALSE(OnSentPacket(kPacketId)); // Packet removed when found.
}
TEST_F(SendDelayStatsTest, SentPacketNotFoundForNonRegisteredSsrc) {
OnSendPacket(kPacketId, kSsrc1);
EXPECT_TRUE(OnSentPacket(kPacketId));
OnSendPacket(kPacketId + 1, kSsrc2);
EXPECT_TRUE(OnSentPacket(kPacketId + 1));
OnSendPacket(kPacketId + 2, kRtxSsrc1); // RTX SSRC not registered.
EXPECT_FALSE(OnSentPacket(kPacketId + 2));
}
TEST_F(SendDelayStatsTest, SentPacketFoundWithMaxSendDelay) {
OnSendPacket(kPacketId, kSsrc1);
clock_.AdvanceTime(kMaxPacketDelay - TimeDelta::Millis(1));
OnSendPacket(kPacketId + 1, kSsrc1); // kPacketId -> not old/removed.
EXPECT_TRUE(OnSentPacket(kPacketId)); // Packet found.
EXPECT_TRUE(OnSentPacket(kPacketId + 1)); // Packet found.
}
TEST_F(SendDelayStatsTest, OldPacketsRemoved) {
const Timestamp kCaptureTime = clock_.CurrentTime();
OnSendPacket(0xffffu, kSsrc1, kCaptureTime);
OnSendPacket(0u, kSsrc1, kCaptureTime);
OnSendPacket(1u, kSsrc1, kCaptureTime + TimeDelta::Millis(1));
clock_.AdvanceTime(kMaxPacketDelay); // 0xffff, 0 -> old.
OnSendPacket(2u, kSsrc1, kCaptureTime + TimeDelta::Millis(2));
EXPECT_FALSE(OnSentPacket(0xffffu)); // Old removed.
EXPECT_FALSE(OnSentPacket(0u)); // Old removed.
EXPECT_TRUE(OnSentPacket(1u));
EXPECT_TRUE(OnSentPacket(2u));
}
TEST_F(SendDelayStatsTest, HistogramsAreUpdated) {
metrics::Reset();
const int64_t kDelayMs1 = 5;
const int64_t kDelayMs2 = 15;
const int kNumSamples = kMinRequiredPeriodicSamples * kProcessIntervalMs /
(kDelayMs1 + kDelayMs2) +
1;
uint16_t id = 0;
for (int i = 0; i < kNumSamples; ++i) {
OnSendPacket(++id, kSsrc1);
clock_.AdvanceTimeMilliseconds(kDelayMs1);
EXPECT_TRUE(OnSentPacket(id));
OnSendPacket(++id, kSsrc2);
clock_.AdvanceTimeMilliseconds(kDelayMs2);
EXPECT_TRUE(OnSentPacket(id));
}
stats_.reset();
EXPECT_METRIC_EQ(2, metrics::NumSamples("WebRTC.Video.SendDelayInMs"));
EXPECT_METRIC_EQ(1,
metrics::NumEvents("WebRTC.Video.SendDelayInMs", kDelayMs1));
EXPECT_METRIC_EQ(1,
metrics::NumEvents("WebRTC.Video.SendDelayInMs", kDelayMs2));
}
} // namespace webrtc