Fix expectation bug in the RTPSender unit test.

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 107 matching lines @@
 
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
       : fake_clock_(kStartTime),
         mock_paced_sender_(),
         rtp_sender_(),
         payload_(kPayload),
         transport_(),
         kMarkerBit(true) {
-    EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
-        .WillRepeatedly(testing::Return());
   }
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                     pacer ? &mock_paced_sender_ : nullptr,
                                     nullptr, nullptr, nullptr, nullptr,
                                     nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
@@ skipping 481 matching lines @@
   EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
   EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
   EXPECT_FALSE(rtp_header2.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header2.extension.transportSequenceNumber);
 }
 
 TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                _, kSeqNum, _, _, _))
-      .WillRepeatedly(testing::Return());
+      .WillOnce(testing::Return());

[stefan-webrtc, 2016/01/14 15:03:42]

I _think_ you don't need this if the method is void?

[terelius, 2016/01/14 15:41:01]

You're right. Done.

 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
@@ skipping 31 matching lines @@
   // Verify transmission time offset.
   EXPECT_EQ(kStoredTimeInMs * 90, 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, TrafficSmoothingRetransmits) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                _, kSeqNum, _, _, _))
-      .WillRepeatedly(testing::Return());
+      .WillOnce(testing::Return());
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // 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_);
 
-  EXPECT_CALL(mock_paced_sender_,
-              InsertPacket(RtpPacketSender::kHighPriority, _, kSeqNum, _, _, _))
-      .WillRepeatedly(testing::Return());
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
+                                               _, kSeqNum, _, _, _))
+      .WillOnce(testing::Return());
 
   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);
 
   // Process send bucket. Packet should now be sent.
@@ skipping 15 matching lines @@
   EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
   uint64_t expected_send_time =
       ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
 // This test sends 1 regular video packet, then 4 padding packets, and then
 // 1 more regular packet.
 TEST_F(RtpSenderTest, SendPadding) {
   // Make all (non-padding) packets go to send queue.
-  EXPECT_CALL(mock_paced_sender_,
-              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
-      .WillRepeatedly(testing::Return());
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
+                                               _, kSeqNum, _, _, _))
+      .WillOnce(testing::Return());
 
   uint16_t seq_num = kSeqNum;
   uint32_t timestamp = kTimestamp;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   size_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
   EXPECT_EQ(
@@ skipping 68 matching lines @@
     timestamp += 90 * kPaddingPeriodMs;
   }
 
   // Send a regular video packet again.
   capture_time_ms = fake_clock_.TimeInMilliseconds();
   rtp_length_int = rtp_sender_->BuildRTPheader(packet_, kPayload, kMarkerBit,
                                                timestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   rtp_length = static_cast<size_t>(rtp_length_int);
 
+  EXPECT_CALL(mock_paced_sender_,
+              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
+      .WillOnce(testing::Return());
+
   // 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);
   // Process send bucket.
   EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
   EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
   // Parse sent packet.
@@ skipping 10 matching lines @@
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
                                   &mock_paced_sender_, nullptr, nullptr,
                                   nullptr, nullptr, nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
-  // Make all packets go through the pacer.
-  EXPECT_CALL(mock_paced_sender_,
-              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
-      .WillRepeatedly(testing::Return());
 
   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.
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
   rtp_sender_->SetRtxSsrc(1234);
 
   // Create and set up parser.
   rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
                                          kTransmissionTimeOffsetExtensionId);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                          kAbsoluteSendTimeExtensionId);
   rtp_sender_->SetTargetBitrate(300000);
   const size_t kNumPayloadSizes = 10;
   const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
                                                   750, 800, 850, 900, 950};
+  // Expect all packets go through the pacer.
+  EXPECT_CALL(mock_paced_sender_,
+              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
+      .Times(kNumPayloadSizes)
+      .WillRepeatedly(testing::Return());
   // 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);
     fake_clock_.AdvanceTimeMilliseconds(33);
   }
   // The amount of padding to send it too small to send a payload packet.
   EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
@@ skipping 92 matching lines @@
                                   nullptr, &callback, 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(_, _, _, _, _, _))
+      .Times(::testing::AtLeast(2))
+      .WillRepeatedly(testing::Return());
+
   ASSERT_EQ(
       0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
                                        payload, sizeof(payload), nullptr));
 
   EXPECT_EQ(1U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(0, callback.frame_counts_.delta_frames);
 
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
@@ skipping 402 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