Use SingleThreadedTaskQueue in DirectTransport

webrtc/call/bitrate_estimator_tests.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 <functional>
@@ skipping 84 matching lines @@
 static const int kTOFExtensionId = 4;
 static const int kASTExtensionId = 5;
 
 class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest() : receive_config_(nullptr) {}
 
   virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
 
   virtual void SetUp() {
-    Call::Config config(event_log_.get());
-    receiver_call_.reset(Call::Create(config));
-    sender_call_.reset(Call::Create(config));
+    task_queue_.SendTask([this]() {
+      Call::Config config(event_log_.get());
+      receiver_call_.reset(Call::Create(config));
+      sender_call_.reset(Call::Create(config));
 
-    send_transport_.reset(
-        new test::DirectTransport(sender_call_.get(), payload_type_map_));
-    send_transport_->SetReceiver(receiver_call_->Receiver());
-    receive_transport_.reset(
-        new test::DirectTransport(receiver_call_.get(), payload_type_map_));
-    receive_transport_->SetReceiver(sender_call_->Receiver());
+      send_transport_.reset(new test::DirectTransport(
+          &task_queue_, sender_call_.get(), payload_type_map_));
+      send_transport_->SetReceiver(receiver_call_->Receiver());
+      receive_transport_.reset(new test::DirectTransport(
+          &task_queue_, receiver_call_.get(), payload_type_map_));
+      receive_transport_->SetReceiver(sender_call_->Receiver());
 
-    video_send_config_ = VideoSendStream::Config(send_transport_.get());
-    video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
-    // Encoders will be set separately per stream.
-    video_send_config_.encoder_settings.encoder = nullptr;
-    video_send_config_.encoder_settings.payload_name = "FAKE";
-    video_send_config_.encoder_settings.payload_type =
-        kFakeVideoSendPayloadType;
-    test::FillEncoderConfiguration(1, &video_encoder_config_);
+      video_send_config_ = VideoSendStream::Config(send_transport_.get());
+      video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
+      // Encoders will be set separately per stream.
+      video_send_config_.encoder_settings.encoder = nullptr;
+      video_send_config_.encoder_settings.payload_name = "FAKE";
+      video_send_config_.encoder_settings.payload_type =
+          kFakeVideoSendPayloadType;
+      test::FillEncoderConfiguration(1, &video_encoder_config_);
 
-    receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
-    // receive_config_.decoders will be set by every stream separately.
-    receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
-    receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
-    receive_config_.rtp.remb = true;
-    receive_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
-    receive_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
+      receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
+      // receive_config_.decoders will be set by every stream separately.
+      receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
+      receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+      receive_config_.rtp.remb = true;
+      receive_config_.rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
+      receive_config_.rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
+    });
   }
 
   virtual void TearDown() {
-    std::for_each(streams_.begin(), streams_.end(),
-                  std::mem_fun(&Stream::StopSending));
+    task_queue_.SendTask([this]() {
+      std::for_each(streams_.begin(), streams_.end(),
+                    std::mem_fun(&Stream::StopSending));
 
-    send_transport_->StopSending();

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

Why is StopSending no longer needed?

Also dropped in call_test.cc.

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

reset() both stops as well as deallocates, so StopSending() became superfluous,
and so I deleted it.

-    receive_transport_->StopSending();
+      while (!streams_.empty()) {
+        delete streams_.back();
+        streams_.pop_back();
+      }
 
-    while (!streams_.empty()) {
-      delete streams_.back();
-      streams_.pop_back();
-    }
+      send_transport_.reset();
+      receive_transport_.reset();
 
-    receiver_call_.reset();
-    sender_call_.reset();
+      receiver_call_.reset();
+      sender_call_.reset();
+    });
   }
 
  protected:
   friend class Stream;
 
   class Stream {
    public:
     explicit Stream(BitrateEstimatorTest* test)
         : test_(test),
           is_sending_receiving_(false),
@@ skipping 74 matching lines @@
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };
 
 static const char* kAbsSendTimeLog =
     "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
 static const char* kSingleStreamLog =
     "RemoteBitrateEstimatorSingleStream: Instantiating.";
 
 TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  streams_.push_back(new Stream(this));
+  task_queue_.SendTask([this]() {
+    video_send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    streams_.push_back(new Stream(this));
+  });
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
-  streams_.push_back(new Stream(this));
+  task_queue_.SendTask([this]() {
+    video_send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+    receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+    streams_.push_back(new Stream(this));
+  });
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  streams_.push_back(new Stream(this));
+  task_queue_.SendTask([this]() {
+    video_send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    streams_.push_back(new Stream(this));
+  });
   EXPECT_TRUE(receiver_log_.Wait());
 
-  video_send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
-  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
-  streams_.push_back(new Stream(this));
+  task_queue_.SendTask([this]() {
+    video_send_config_.rtp.extensions[0] =
+        RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
+    receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+    receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+    streams_.push_back(new Stream(this));
+  });
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
 // This test is flaky. See webrtc:5790.
 TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  streams_.push_back(new Stream(this));
+  task_queue_.SendTask([this]() {
+    video_send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+    receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+    streams_.push_back(new Stream(this));
+  });
   EXPECT_TRUE(receiver_log_.Wait());
 
-  video_send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
-  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
-  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  streams_.push_back(new Stream(this));
+  task_queue_.SendTask([this]() {
+    video_send_config_.rtp.extensions[0] =
+        RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
+    receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+    receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+    streams_.push_back(new Stream(this));
+  });
   EXPECT_TRUE(receiver_log_.Wait());
 
-  video_send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
-  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
-  receiver_log_.PushExpectedLogLine(
-      "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
-  streams_.push_back(new Stream(this));
-  streams_[0]->StopSending();
-  streams_[1]->StopSending();
+  task_queue_.SendTask([this]() {
+    video_send_config_.rtp.extensions[0] =
+        RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
+    receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+    receiver_log_.PushExpectedLogLine(
+        "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
+    streams_.push_back(new Stream(this));
+    streams_[0]->StopSending();
+    streams_[1]->StopSending();
+  });
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc