Refactoring: CastTransport InitializeAudio/InitializeVideo.

media/cast/net/rtp/rtp_packetizer_unittest.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/net/rtp/rtp_packetizer.h"
 
 #include <stddef.h>
 #include <stdint.h>
 
 #include <memory>
@@ skipping 104 matching lines @@
   RtpPacketizerTest()
       : task_runner_(new FakeSingleThreadTaskRunner(&testing_clock_)) {
     config_.sequence_number = kSeqNum;
     config_.ssrc = kSsrc;
     config_.payload_type = kPayload;
     config_.max_payload_length = kMaxPacketLength;
     transport_.reset(new TestRtpPacketTransport(config_));
     pacer_.reset(new PacedSender(kTargetBurstSize, kMaxBurstSize,
                                  &testing_clock_, nullptr, transport_.get(),
                                  task_runner_));
-    pacer_->RegisterVideoSsrc(config_.ssrc);
+    pacer_->RegisterSsrc(config_.ssrc, false);
     rtp_packetizer_.reset(new RtpPacketizer(
         pacer_.get(), &packet_storage_, config_));
     video_frame_.dependency = EncodedFrame::DEPENDENT;
     video_frame_.frame_id = FrameId::first() + 1;
     video_frame_.referenced_frame_id = video_frame_.frame_id - 1;
     video_frame_.data.assign(kFrameSize, 123);
     video_frame_.rtp_timestamp = RtpTimeTicks().Expand(UINT32_C(0x0055aa11));
   }
 
   void RunTasks(int during_ms) {
@@ skipping 54 matching lines @@
   video_frame_.reference_time = testing_clock_.NowTicks();
   rtp_packetizer_->SendFrameAsPackets(video_frame_);
   RunTasks(33 + 1);
   EXPECT_EQ(expected_num_of_packets, rtp_packetizer_->send_packet_count());
   EXPECT_EQ(kFrameSize, rtp_packetizer_->send_octet_count());
   EXPECT_EQ(expected_num_of_packets, transport_->number_of_packets_received());
 }
 
 }  // namespace cast
 }  // namespace media