cast: Split Rtcp into two for sender and receiver

media/cast/sender/audio_sender_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/sender/audio_sender.h"
 
 #include <stdint.h>
 #include <utility>
 
 #include "base/bind.h"
@@ skipping 24 matching lines @@
   *out_status = in_status;
 }
 
 }  // namespace
 
 class TestPacketSender : public PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
   bool SendPacket(PacketRef packet, const base::Closure& cb) final {
-    if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
+    if (IsRtcpPacket(&packet->data[0], packet->data.size())) {
       ++number_of_rtcp_packets_;
     } else {
       // Check that at least one RTCP packet was sent before the first RTP
       // packet.  This confirms that the receiver will have the necessary lip
       // sync info before it has to calculate the playout time of the first
       // frame.
       if (number_of_rtp_packets_ == 0)
         EXPECT_LE(1, number_of_rtcp_packets_);
       ++number_of_rtp_packets_;
     }
@@ skipping 98 matching lines @@
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kRtcpReportIntervalMs * 3 / 2);
   testing_clock_->Advance(max_rtcp_timeout);
   task_runner_->RunTasks();
   EXPECT_LE(1, transport_.number_of_rtp_packets());
   EXPECT_LE(1, transport_.number_of_rtcp_packets());
 }
 
 }  // namespace cast
 }  // namespace media