Add histogram stats for send delay for a sent video stream.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 108 matching lines @@
                     size_t bytes,
                     bool retransmission));
 };
 
 class MockTransportSequenceNumberAllocator
     : public TransportSequenceNumberAllocator {
  public:
   MOCK_METHOD0(AllocateSequenceNumber, uint16_t());
 };
 
+class MockSendPacketObserver : public SendPacketObserver {
+ public:
+  MOCK_METHOD3(OnSendPacket, void(uint16_t, int64_t, uint32_t));
+};
+
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
       : fake_clock_(kStartTime),
         mock_rtc_event_log_(),
         mock_paced_sender_(),
         rtp_sender_(),
         payload_(kPayload),
         transport_(),
         kMarkerBit(true) {
   }
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
-    rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
-                                    pacer ? &mock_paced_sender_ : nullptr,
-                                    &seq_num_allocator_, nullptr, nullptr,
-                                    nullptr, nullptr, &mock_rtc_event_log_));
+    rtp_sender_.reset(new RTPSender(
+        false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
+        &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
+        &mock_rtc_event_log_, &send_packet_observer_));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
   SimulatedClock fake_clock_;
   MockRtcEventLog mock_rtc_event_log_;
   MockRtpPacketSender mock_paced_sender_;
   MockTransportSequenceNumberAllocator seq_num_allocator_;
+  MockSendPacketObserver send_packet_observer_;
   rtc::scoped_ptr<RTPSender> rtp_sender_;
   int payload_;
   LoopbackTransportTest transport_;
   const bool kMarkerBit;
   uint8_t packet_[kMaxPacketLength];
 
   void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
     VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
   }
 
@@ skipping 17 matching lines @@
     uint32_t timestamp = capture_time_ms * 90;
     int32_t rtp_length = rtp_sender_->BuildRTPheader(
         packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
     ASSERT_GE(rtp_length, 0);
 
     // Packet should be stored in a send bucket.
     EXPECT_EQ(0, rtp_sender_->SendToNetwork(
                      packet_, payload_length, rtp_length, capture_time_ms,
                      kAllowRetransmission, RtpPacketSender::kNormalPriority));
   }
+
+  void SendGenericPayload() {
+    const uint8_t kPayload[] = {47, 11, 32, 93, 89};
+    const uint32_t kTimestamp = 1234;
+    const uint8_t kPayloadType = 127;
+    const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
+    char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
+    EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
+                                              0, 1500));
+
+    EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
+                     kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
+                     kPayload, sizeof(kPayload), nullptr));
+  }
 };
 
 // TODO(pbos): Move tests over from WithoutPacer to RtpSenderTest as this is our
 // default code path.
 class RtpSenderTestWithoutPacer : public RtpSenderTest {
  public:
   void SetUp() override { SetUpRtpSender(false); }
 };
 
 class RtpSenderVideoTest : public RtpSenderTest {
@@ skipping 274 matching lines @@
 
 TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
   // Ignore rtc event calls.
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
 
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
 
-  char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
-  const uint8_t payload_type = 127;
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
-                                            0, 1500));
-  // Create a dummy payload of 5 bytes.
-  uint8_t payload[] = {47, 11, 32, 93, 89};
-
-  const uint16_t kTransportSequenceNumber = 17;
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
-  const uint32_t kTimestamp = 1234;
-  const int64_t kCaptureTimeMs = 4321;
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
-                   kVideoFrameKey, payload_type, kTimestamp, kCaptureTimeMs,
-                   payload, sizeof(payload), nullptr));
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber, _, _))
+      .Times(1);
+
+  SendGenericPayload();
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransportSequenceNumber,
                kTransportSequenceNumberExtensionId);
   EXPECT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
   EXPECT_EQ(kTransportSequenceNumber,
             rtp_header.extension.transportSequenceNumber);
   EXPECT_EQ(transport_.last_packet_id_,
             rtp_header.extension.transportSequenceNumber);
 }
 
+TEST_F(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
+  EXPECT_CALL(mock_rtc_event_log_,  // Ignore rtc event calls.
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber, _, _))
+      .Times(1);
+
+  SendGenericPayload();
+}
+
 // Test CVO header extension is only set when marker bit is true.
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
   rtp_sender_->SetVideoRotation(kRotation);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
   EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
 
@@ skipping 418 matching lines @@
   // Verify sequence number and timestamp.
   EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
   EXPECT_EQ(timestamp, rtp_header.timestamp);
   // Verify transmission time offset. This packet is sent without delay.
   EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
   uint64_t expected_send_time =
       ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
+TEST_F(RtpSenderTest, OnSendPacketUpdated) {
+  EXPECT_CALL(mock_rtc_event_log_,  // Ignore rtc event calls.
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber, _, _))
+      .Times(1);
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
+
+  SendGenericPayload();  // Packet passed to pacer.
+  const bool kIsRetransmit = false;
+  rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
+                                kIsRetransmit);
+  EXPECT_EQ(1, transport_.packets_sent_);
+}
+
+TEST_F(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) {
+  EXPECT_CALL(mock_rtc_event_log_,  // Ignore rtc event calls.
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+
+  EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
+
+  SendGenericPayload();  // Packet passed to pacer.
+  const bool kIsRetransmit = true;
+  rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
+                                kIsRetransmit);
+  EXPECT_EQ(1, transport_.packets_sent_);
+}
+
+TEST_F(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, &mock_paced_sender_,
+      nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
+      nullptr, nullptr, &send_packet_observer_));
+  rtp_sender_->SetSequenceNumber(kSeqNum);
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+
+  EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
+
+  SendGenericPayload();  // Packet passed to pacer.
+  const bool kIsRetransmit = false;
+  rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
+                                kIsRetransmit);
+  EXPECT_EQ(1, transport_.packets_sent_);
+}
+
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
-      nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_));
+      false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
+      nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr));
+
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
 
   uint16_t seq_num = kSeqNum;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   int32_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
@@ skipping 120 matching lines @@
       ++num_calls_;
       ssrc_ = ssrc;
       frame_counts_ = frame_counts;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     FrameCounts frame_counts_;
   } callback;
 
-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
-                                  &mock_paced_sender_, nullptr, nullptr,
-                                  nullptr, &callback, nullptr, nullptr));
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,
+      nullptr, &callback, nullptr, nullptr, nullptr));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
@@ skipping 33 matching lines @@
       ssrc_ = ssrc;
       total_stats_ = total_stats;
       retransmit_stats_ = retransmit_stats;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     BitrateStatistics total_stats_;
     BitrateStatistics retransmit_stats_;
   } callback;
-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
-                                  nullptr, nullptr, nullptr, &callback, nullptr,
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+                                  nullptr, nullptr, &callback, nullptr, nullptr,
                                   nullptr, nullptr));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals.
   const uint32_t kNumPackets = 15;
   const uint32_t kPacketInterval = 20;
   // Overhead = 12 bytes RTP header + 1 byte generic header.
   const uint32_t kPacketOverhead = 13;
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
@@ skipping 31 matching lines @@
 
   rtp_sender_.reset();
 }
 
 class RtpSenderAudioTest : public RtpSenderTest {
  protected:
   RtpSenderAudioTest() {}
 
   void SetUp() override {
     payload_ = kAudioPayload;
-    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_,
+    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
                                     nullptr, nullptr, nullptr, nullptr, nullptr,
                                     nullptr, nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
 
 TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
     TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
@@ skipping 307 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc