Use SingleThreadedTaskQueue in DirectTransport

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 <deque>
 
 #include "webrtc/api/call/transport.h"
 #include "webrtc/call/call.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/event.h"
-#include "webrtc/rtc_base/platform_thread.h"
+#include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/test/fake_network_pipe.h"
+#include "webrtc/test/single_threaded_task_queue.h"
 
 namespace webrtc {
 
 class Clock;
 class PacketReceiver;
 
 namespace test {
 
 class DirectTransport : public Transport {
  public:
-  DirectTransport(Call* send_call,
-                  const std::map<uint8_t, MediaType>& payload_type_map);
-  DirectTransport(const FakeNetworkPipe::Config& config,
+  DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
                   Call* send_call,
                   const std::map<uint8_t, MediaType>& payload_type_map);
-  DirectTransport(const FakeNetworkPipe::Config& config,
+
+  DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
+                  const FakeNetworkPipe::Config& config,
+                  Call* send_call,
+                  const std::map<uint8_t, MediaType>& payload_type_map);
+
+  DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
+                  const FakeNetworkPipe::Config& config,
                   Call* send_call,
                   std::unique_ptr<Demuxer> demuxer);
 
   // This deprecated variant always uses MediaType::VIDEO.
   RTC_DEPRECATED explicit DirectTransport(Call* send_call)
       : DirectTransport(
+  // TODO(eladalon): !!! Deprecate older APIs gently. (Before landing.)
+            nullptr,
             FakeNetworkPipe::Config(),
             send_call,
             std::unique_ptr<Demuxer>(new ForceDemuxer(MediaType::VIDEO))) {}
 
   ~DirectTransport() override;
 
   void SetConfig(const FakeNetworkPipe::Config& config);
 
-  virtual void StopSending();

[nisse-webrtc, 2017/08/21 09:07:07]

Ok, so you're deleting this method? Worth mentioning in the cl description.

[eladalon, 2017/08/21 10:56:53]

Done.

   // 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:
   // TODO(minyue): remove when the deprecated ctors of DirectTransport that
   // create ForceDemuxer are removed.
   class ForceDemuxer : public Demuxer {
    public:
     explicit ForceDemuxer(MediaType media_type);
     void SetReceiver(PacketReceiver* receiver) override;
     void DeliverPacket(const NetworkPacket* packet,
                        const PacketTime& packet_time) override;
 
    private:
     const MediaType media_type_;
     PacketReceiver* packet_receiver_;
     RTC_DISALLOW_COPY_AND_ASSIGN(ForceDemuxer);
   };
 
-  static bool NetworkProcess(void* transport);
-  bool SendPackets();
+  void SendPackets();
 
-  rtc::CriticalSection lock_;
   Call* const send_call_;
-  rtc::Event packet_event_;
-  rtc::PlatformThread thread_;
   Clock* const clock_;
 
-  bool shutting_down_;
+  SingleThreadedTaskQueueForTesting* const task_queue_;
+  SingleThreadedTaskQueueForTesting::TaskId next_scheduled_task_;
 
   FakeNetworkPipe fake_network_;
+
+  rtc::ThreadChecker thread_checker_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_DIRECT_TRANSPORT_H_