Use SingleThreadedTaskQueue in DirectTransport

webrtc/test/direct_transport.cc

 /*
  *  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.
  */
 #include "webrtc/test/direct_transport.h"
 
 #include "webrtc/call/call.h"
+#include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/single_threaded_task_queue.h"
 
 namespace webrtc {
 namespace test {
 
 DirectTransport::DirectTransport(
     Call* send_call,
     const std::map<uint8_t, MediaType>& payload_type_map)
     : DirectTransport(FakeNetworkPipe::Config(), send_call, payload_type_map) {}
 
 DirectTransport::DirectTransport(
     const FakeNetworkPipe::Config& config,
     Call* send_call,
     const std::map<uint8_t, MediaType>& payload_type_map)
     : DirectTransport(
           config,
           send_call,
           std::unique_ptr<Demuxer>(new DemuxerImpl(payload_type_map))) {}
 
 DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
                                  Call* send_call,
                                  std::unique_ptr<Demuxer> demuxer)
+    : DirectTransport(nullptr, config, send_call, std::move(demuxer)) {}
+
+DirectTransport::DirectTransport(
+    SingleThreadedTaskQueueForTesting* task_queue,
+    Call* send_call,
+    const std::map<uint8_t, MediaType>& payload_type_map)
+    : DirectTransport(task_queue,
+                      FakeNetworkPipe::Config(),
+                      send_call,
+                      payload_type_map) {
+}
+
+DirectTransport::DirectTransport(
+    SingleThreadedTaskQueueForTesting* task_queue,
+    const FakeNetworkPipe::Config& config,
+    Call* send_call,
+    const std::map<uint8_t, MediaType>& payload_type_map)
+    : DirectTransport(
+          task_queue,
+          config,
+          send_call,
+          std::unique_ptr<Demuxer>(new DemuxerImpl(payload_type_map))) {
+}
+
+DirectTransport::DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
+                                 const FakeNetworkPipe::Config& config,
+                                 Call* send_call,
+                                 std::unique_ptr<Demuxer> demuxer)
     : send_call_(send_call),
-      packet_event_(false, false),
-      thread_(NetworkProcess, this, "NetworkProcess"),
       clock_(Clock::GetRealTimeClock()),
-      shutting_down_(false),
+      task_queue_(task_queue),
       fake_network_(clock_, config, std::move(demuxer)) {
-  thread_.Start();
+  // TODO(eladalon): When the deprecated ctors are removed, this check
+  // can be restored. https://bugs.chromium.org/p/webrtc/issues/detail?id=8125
+  // RTC_DCHECK(task_queue);
+  if (!task_queue) {
+    deprecated_task_queue_ =
+        rtc::MakeUnique<SingleThreadedTaskQueueForTesting>("deprecated_queue");
+    task_queue_ = deprecated_task_queue_.get();
+  }
+
   if (send_call_) {
     send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
     send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
   }
+  SendPackets();
 }
 
-DirectTransport::~DirectTransport() { StopSending(); }
+DirectTransport::~DirectTransport() {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
+  // Constructor updates |next_scheduled_task_|, so it's guaranteed to
+  // be initialized.
+  task_queue_->CancelTask(next_scheduled_task_);
+}
 
 void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
   fake_network_.SetConfig(config);
 }
 
 void DirectTransport::StopSending() {
-  {
-    rtc::CritScope crit(&lock_);
-    shutting_down_ = true;
-  }
-
-  packet_event_.Set();
-  thread_.Stop();
+  task_queue_->CancelTask(next_scheduled_task_);
 }
 
 void DirectTransport::SetReceiver(PacketReceiver* receiver) {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
   fake_network_.SetReceiver(receiver);
 }
 
 bool DirectTransport::SendRtp(const uint8_t* data,
                               size_t length,
                               const PacketOptions& options) {
   if (send_call_) {
     rtc::SentPacket sent_packet(options.packet_id,
                                 clock_->TimeInMilliseconds());
     send_call_->OnSentPacket(sent_packet);
   }
   fake_network_.SendPacket(data, length);
-  packet_event_.Set();
   return true;
 }
 
 bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
   fake_network_.SendPacket(data, length);
-  packet_event_.Set();
   return true;
 }
 
 int DirectTransport::GetAverageDelayMs() {
   return fake_network_.AverageDelay();
 }
 
 DirectTransport::ForceDemuxer::ForceDemuxer(MediaType media_type)
     : media_type_(media_type) {}
 
 void DirectTransport::ForceDemuxer::SetReceiver(PacketReceiver* receiver) {
   packet_receiver_ = receiver;
 }
 
 void DirectTransport::ForceDemuxer::DeliverPacket(
     const NetworkPacket* packet,
     const PacketTime& packet_time) {
   if (!packet_receiver_)
     return;
   packet_receiver_->DeliverPacket(media_type_, packet->data(),
                                   packet->data_length(), packet_time);
 }
 
-bool DirectTransport::NetworkProcess(void* transport) {
-  return static_cast<DirectTransport*>(transport)->SendPackets();
-}
+void DirectTransport::SendPackets() {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
 
-bool DirectTransport::SendPackets() {
   fake_network_.Process();
-  int64_t wait_time_ms = fake_network_.TimeUntilNextProcess();
-  if (wait_time_ms > 0) {
-    packet_event_.Wait(static_cast<int>(wait_time_ms));
-  }
-  rtc::CritScope crit(&lock_);
-  return shutting_down_ ? false : true;
+
+  int64_t delay_ms = fake_network_.TimeUntilNextProcess();
+  next_scheduled_task_ = task_queue_->PostDelayedTask([this]() {
+    SendPackets();
+  }, delay_ms);
 }
 }  // namespace test
 }  // namespace webrtc