Cast: Refactor RTCP handling

media/cast/sender/audio_sender_unittest.cc

 // Copyright 2013 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 <stdint.h>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/base/media.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/net/cast_transport_config.h"
 #include "media/cast/net/cast_transport_sender_impl.h"
 #include "media/cast/net/rtcp/rtcp.h"

[miu, 2014/07/16 19:58:03]

This, and possibly other #includes can now be removed.

[Alpha Left Google, 2014/07/17 01:01:46]

Done.

+#include "media/cast/net/rtcp/rtcp_receiver.h"
 #include "media/cast/sender/audio_sender.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/utility/audio_utility.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
 
 class TestPacketSender : public PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
   virtual bool SendPacket(PacketRef packet,
                           const base::Closure& cb) OVERRIDE {
-    if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
+    if (RtcpReceiver::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 90 matching lines @@
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 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