Add a PacketOptions struct to webrtc::Transport.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>
@@ skipping 43 matching lines @@
   EndToEndTest() {}
 
   virtual ~EndToEndTest() {
     EXPECT_EQ(nullptr, send_stream_);
     EXPECT_TRUE(receive_streams_.empty());
   }
 
  protected:
   class UnusedTransport : public Transport {
    private:
-    bool SendRtp(const uint8_t* packet, size_t length) override {
+    bool SendRtp(const uint8_t* packet,
+                 size_t length,
+                 const PacketOptions& options) override {
       ADD_FAILURE() << "Unexpected RTP sent.";
       return false;
     }
 
     bool SendRtcp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected RTCP sent.";
       return false;
     }
   };
 
@@ skipping 1266 matching lines @@
         : done_(EventWrapper::Create()),
           parser_(RtpHeaderParser::Create()),
           last_seq_(0),
           padding_observed_(false),
           rtx_padding_observed_(false) {
       parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
                                           kExtensionId);
     }
     virtual ~RtpExtensionHeaderObserver() {}
 
-    bool SendRtp(const uint8_t* data, size_t length) override {
+    bool SendRtp(const uint8_t* data,
+                 size_t length,
+                 const PacketOptions& options) override {
       if (IsDone())
         return false;
 
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(data, length, &header));
       if (header.extension.hasTransportSequenceNumber) {
+        EXPECT_EQ(options.packet_id,
+                  header.extension.transportSequenceNumber);
         if (!streams_observed_.empty()) {
           EXPECT_EQ(static_cast<uint16_t>(last_seq_ + 1),
                     header.extension.transportSequenceNumber);
         }
         last_seq_ = header.extension.transportSequenceNumber;
 
         size_t payload_length =
             length - (header.headerLength + header.paddingLength);
         if (payload_length == 0) {
           padding_observed_ = true;
         } else if (header.payloadType == kSendRtxPayloadType) {
           rtx_padding_observed_ = true;
         } else {
           streams_observed_.insert(header.ssrc);
         }
 
         if (IsDone())
           done_->Set();
       }
-      return test::DirectTransport::SendRtp(data, length);
+      return test::DirectTransport::SendRtp(data, length, options);
     }
 
     bool IsDone() {
       return streams_observed_.size() == MultiStreamTest::kNumStreams &&
              padding_observed_ && rtx_padding_observed_;
     }
 
     EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
 
     rtc::scoped_ptr<EventWrapper> done_;
@@ skipping 1734 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc