Move rtp dump writer from quality test to test transport

webrtc/test/direct_transport.h

 /*
  *  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.
  */
 #ifndef WEBRTC_TEST_DIRECT_TRANSPORT_H_
 #define WEBRTC_TEST_DIRECT_TRANSPORT_H_
 
 #include <assert.h>
 
 #include <memory>
 
 #include "webrtc/api/call/transport.h"
 #include "webrtc/call/call.h"
 #include "webrtc/rtc_base/sequenced_task_checker.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/test/fake_network_pipe.h"
+#include "webrtc/test/rtp_file_writer.h"
 #include "webrtc/test/single_threaded_task_queue.h"
 
 namespace webrtc {
 
 class Clock;
 class PacketReceiver;
 
 namespace test {
 
-// Objects of this class are expected to be allocated and destroyed  on the
+// Objects of this class are expected to be allocated and destroyed on the
 // same task-queue - the one that's passed in via the constructor.
-class DirectTransport : public Transport {
+class DirectTransport : public Transport, private PacketReceiver {
  public:
   DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
                   Call* send_call,
                   const std::map<uint8_t, MediaType>& payload_type_map);
 
   DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
                   const FakeNetworkPipe::Config& config,
                   Call* send_call,
-                  const std::map<uint8_t, MediaType>& payload_type_map);
+                  const std::map<uint8_t, MediaType>& payload_type_map,
+                  std::unique_ptr<test::RtpFileWriter> rtp_file_writer);
 
   DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
                   const FakeNetworkPipe::Config& config,
                   Call* send_call,
-                  std::unique_ptr<Demuxer> demuxer);
+                  std::unique_ptr<Demuxer> demuxer,
+                  std::unique_ptr<test::RtpFileWriter> rtp_file_writer);
 
   ~DirectTransport() override;
 
   void SetConfig(const FakeNetworkPipe::Config& config);
 
   RTC_DEPRECATED void StopSending();
 
   // TODO(holmer): Look into moving this to the constructor.
   virtual void SetReceiver(PacketReceiver* receiver);
 
   bool SendRtp(const uint8_t* data,
                size_t length,
                const PacketOptions& options) override;
   bool SendRtcp(const uint8_t* data, size_t length) override;
 
   int GetAverageDelayMs();
 
  private:
+  DeliveryStatus DeliverPacket(MediaType media_type,
+                               const uint8_t* packet,
+                               size_t length,
+                               const PacketTime& packet_time) override;
+
   void SendPackets();
 
   Call* const send_call_;
   Clock* const clock_;
 
   // TODO(eladalon): Make |task_queue_| const.
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=8125
   SingleThreadedTaskQueueForTesting* task_queue_;
   SingleThreadedTaskQueueForTesting::TaskId next_scheduled_task_
       GUARDED_BY(&sequence_checker_);
 
   FakeNetworkPipe fake_network_;
 
   rtc::SequencedTaskChecker sequence_checker_;
+
+  PacketReceiver* receiver_;
+  const int64_t start_ms_;
+  std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_DIRECT_TRANSPORT_H_