Add a PacketOptions struct to webrtc::Transport.

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 60 matching lines @@
  public:
   LoopbackTransportTest()
       : packets_sent_(0),
         last_sent_packet_len_(0),
         total_bytes_sent_(0),
         last_sent_packet_(nullptr) {}
 
   ~LoopbackTransportTest() {
     STLDeleteContainerPointers(sent_packets_.begin(), sent_packets_.end());
   }
-  bool SendRtp(const uint8_t *data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
     packets_sent_++;
     rtc::Buffer* buffer =
         new rtc::Buffer(reinterpret_cast<const uint8_t*>(data), len);
     last_sent_packet_ = buffer->data();
     last_sent_packet_len_ = len;
     total_bytes_sent_ += len;
     sent_packets_.push_back(buffer);
     return true;
   }
   bool SendRtcp(const uint8_t* data, size_t len) override {
@@ skipping 765 matching lines @@
                                          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};
   // 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));
+    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))
+  EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(kMaxPaddingSize, rtp_sender_->TimeToSendPadding(49));
 
   EXPECT_CALL(transport,
-              SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize))
+              SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(kPayloadSizes[0], rtp_sender_->TimeToSendPadding(500));
 
   EXPECT_CALL(transport, SendRtp(_, kPayloadSizes[kNumPayloadSizes - 1] +
-                                           rtp_header_len + kRtxHeaderSize))
+                                        rtp_header_len + kRtxHeaderSize,
+                                 _))
       .WillOnce(testing::Return(true));
-  EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len))
+  EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
             rtp_sender_->TimeToSendPadding(999));
 }
 
 TEST_F(RtpSenderTest, 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));
@@ skipping 506 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