Propagate probing cluster id to SendTimeHistory.

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 748 matching lines @@
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
 
   EXPECT_EQ(0, transport_.packets_sent_);
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
 
-  rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false);
+  rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false, -1);
 
   // Process send bucket. Packet should now be sent.
   EXPECT_EQ(1, transport_.packets_sent_);
   EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
   // Parse sent packet.
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                                  rtp_length);
   webrtc::RTPHeader rtp_header;
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
@@ skipping 38 matching lines @@
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                _, kSeqNum, _, _, _));
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
 
   EXPECT_EQ(rtp_length_int, rtp_sender_->ReSendPacket(kSeqNum));
   EXPECT_EQ(0, transport_.packets_sent_);
 
-  rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false);
+  rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false, -1);
 
   // Process send bucket. Packet should now be sent.
   EXPECT_EQ(1, transport_.packets_sent_);
   EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
 
   // Parse sent packet.
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                                  rtp_length);
   webrtc::RTPHeader rtp_header;
   RtpHeaderExtensionMap map;
@@ skipping 55 matching lines @@
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
 
   int total_packets_sent = 0;
   EXPECT_EQ(total_packets_sent, transport_.packets_sent_);
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
-  rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false);
+  rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false, -1);
   // Packet should now be sent. This test doesn't verify the regular video
   // packet, since it is tested in another test.
   EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
   timestamp += 90 * kStoredTimeInMs;
 
   // Send padding 4 times, waiting 50 ms between each.
   for (int i = 0; i < 4; ++i) {
     const int kPaddingPeriodMs = 50;
     const size_t kPaddingBytes = 100;
     const size_t kMaxPaddingLength = 224;  // Value taken from rtp_sender.cc.
     // Padding will be forced to full packets.
-    EXPECT_EQ(kMaxPaddingLength, rtp_sender_->TimeToSendPadding(kPaddingBytes));
+    EXPECT_EQ(kMaxPaddingLength,
+              rtp_sender_->TimeToSendPadding(kPaddingBytes, -1));
 
     // Process send bucket. Padding should now be sent.
     EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
     EXPECT_EQ(kMaxPaddingLength + rtp_header_len,
               transport_.last_sent_packet_len_);
     // Parse sent packet.
     ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
                                   transport_.last_sent_packet_len_,
                                   &rtp_header));
     EXPECT_EQ(kMaxPaddingLength, rtp_header.paddingLength);
@@ skipping 20 matching lines @@
   rtp_length = static_cast<size_t>(rtp_length_int);
 
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _));
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
 
-  rtp_sender_->TimeToSendPacket(seq_num, capture_time_ms, false);
+  rtp_sender_->TimeToSendPacket(seq_num, capture_time_ms, false, -1);
   // Process send bucket.
   EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
   EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
   // Parse sent packet.
   ASSERT_TRUE(
       rtp_parser->Parse(transport_.last_sent_packet_, rtp_length, &rtp_header));
 
   // Verify sequence number and timestamp.
   EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
   EXPECT_EQ(timestamp, rtp_header.timestamp);
@@ skipping 12 matching lines @@
   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);
+                                kIsRetransmit, -1);
   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);
+                                kIsRetransmit, -1);
   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);
+                                kIsRetransmit, -1);
   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_, nullptr));
 
   rtp_sender_->SetSequenceNumber(kSeqNum);
@@ skipping 29 matching lines @@
       .Times(kNumPayloadSizes);
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(kNumPayloadSizes);
 
   // Send 10 packets of increasing size.
   for (size_t i = 0; i < kNumPayloadSizes; ++i) {
     int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
     EXPECT_CALL(transport, SendRtp(_, _, _)).WillOnce(testing::Return(true));
     SendPacket(capture_time_ms, kPayloadSizes[i]);
-    rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false);
+    rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false, -1);
     fake_clock_.AdvanceTimeMilliseconds(33);
   }
 
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(::testing::AtLeast(4));
 
   // The amount of padding to send it too small to send a payload packet.
   EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
       .WillOnce(testing::Return(true));
-  EXPECT_EQ(kMaxPaddingSize, rtp_sender_->TimeToSendPadding(49));
+  EXPECT_EQ(kMaxPaddingSize, rtp_sender_->TimeToSendPadding(49, -1));
 
   EXPECT_CALL(transport,
               SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
       .WillOnce(testing::Return(true));
-  EXPECT_EQ(kPayloadSizes[0], rtp_sender_->TimeToSendPadding(500));
+  EXPECT_EQ(kPayloadSizes[0], rtp_sender_->TimeToSendPadding(500, -1));
 
   EXPECT_CALL(transport, SendRtp(_, kPayloadSizes[kNumPayloadSizes - 1] +
                                         rtp_header_len + kRtxHeaderSize,
                                  _))
       .WillOnce(testing::Return(true));
   EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
-            rtp_sender_->TimeToSendPadding(999));
+            rtp_sender_->TimeToSendPadding(999, -1));
 }
 
 TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
   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};
 
   // Send keyframe
@@ skipping 242 matching lines @@
   expected.transmitted.payload_bytes = 12;
   expected.transmitted.header_bytes = 24;
   expected.transmitted.packets = 2;
   expected.retransmitted.payload_bytes = 6;
   expected.retransmitted.header_bytes = 12;
   expected.retransmitted.padding_bytes = 0;
   expected.retransmitted.packets = 1;
   callback.Matches(ssrc, expected);
 
   // Send padding.
-  rtp_sender_->TimeToSendPadding(kMaxPaddingSize);
+  rtp_sender_->TimeToSendPadding(kMaxPaddingSize, -1);
   expected.transmitted.payload_bytes = 12;
   expected.transmitted.header_bytes = 36;
   expected.transmitted.padding_bytes = kMaxPaddingSize;
   expected.transmitted.packets = 3;
   callback.Matches(ssrc, expected);
 
   // Send FEC.
   rtp_sender_->SetGenericFECStatus(true, kRedPayloadType, kUlpfecPayloadType);
   FecProtectionParams fec_params;
   fec_params.fec_mask_type = kFecMaskRandom;
@@ skipping 136 matching lines @@
 
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   ASSERT_EQ(
       0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321,
                                        payload, sizeof(payload), 0));
 
   // Will send 2 full-size padding packets.
-  rtp_sender_->TimeToSendPadding(1);
-  rtp_sender_->TimeToSendPadding(1);
+  rtp_sender_->TimeToSendPadding(1, -1);
+  rtp_sender_->TimeToSendPadding(1, -1);
 
   StreamDataCounters rtp_stats;
   StreamDataCounters rtx_stats;
   rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
 
   // Payload + 1-byte generic header.
   EXPECT_GT(rtp_stats.first_packet_time_ms, -1);
   EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(payload) + 1);
   EXPECT_EQ(rtp_stats.transmitted.header_bytes, 12u);
   EXPECT_EQ(rtp_stats.transmitted.padding_bytes, 0u);
@@ skipping 69 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