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Optimize execution time of RTPSender::UpdateDelayStatistics

Browse Source 
Bug: webrtc:9439
Change-Id: I908e9ced10031c614678a89657d089cb9a66b9ce
Reviewed-on: https://webrtc-review.googlesource.com/92391
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24295}
This commit is contained in:
Bjorn Terelius
2018-08-15 14:47:41 +02:00
committed by Commit Bot
parent cdb87f1a65
commit 733df738e3
3 changed files with 133 additions and 17 deletions
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70
modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -133,6 +133,11 @@ class MockTransportSequenceNumberAllocator
MOCK_METHOD0(AllocateSequenceNumber, uint16_t());
};
class MockSendSideDelayObserver : public SendSideDelayObserver {
public:
MOCK_METHOD3(SendSideDelayUpdated, void(int, int, uint32_t));
};
class MockSendPacketObserver : public SendPacketObserver {
public:
MOCK_METHOD3(OnSendPacket, void(uint16_t, int64_t, uint32_t));
@ -485,6 +490,71 @@ TEST_P(RtpSenderTestWithoutPacer, NoAllocationIfNotRegistered) {
SendGenericPayload();
}
TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
testing::StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, &send_side_delay_observer_,
&mock_rtc_event_log_, nullptr, nullptr, nullptr, false));
rtp_sender_->SetSSRC(kSsrc);
const uint8_t kPayloadType = 127;
const uint32_t kCaptureTimeMsToRtpTimestamp = 90; // 90 kHz clock
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
RTPVideoHeader video_header;
EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType,
1000 * kCaptureTimeMsToRtpTimestamp,
0, 1500));
// Send packet with 10 ms send-side delay. The average and max should be 10
// ms.
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(10, 10, kSsrc))
.Times(1);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
fake_clock_.AdvanceTimeMilliseconds(10);
EXPECT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet with 20 ms delay. The average
// and max should be 15 and 20 ms respectively.
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(15, 20, kSsrc))
.Times(1);
fake_clock_.AdvanceTimeMilliseconds(10);
EXPECT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet at the same time, which replaces the last packet.
// Since this packet has 0 ms delay, the average is now 5 ms and max is 10 ms.
// TODO(terelius): Is is not clear that this is the right behavior.
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(5, 10, kSsrc))
.Times(1);
capture_time_ms = fake_clock_.TimeInMilliseconds();
EXPECT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
kDefaultExpectedRetransmissionTimeMs));
// Send a packet 1 second later. The earlier packets should have timed
// out, so both max and average should be the delay of this packet.
fake_clock_.AdvanceTimeMilliseconds(1000);
capture_time_ms = fake_clock_.TimeInMilliseconds();
fake_clock_.AdvanceTimeMilliseconds(1);
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(1, 1, kSsrc))
.Times(1);
EXPECT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
kDefaultExpectedRetransmissionTimeMs));
}
TEST_P(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,