Call OnSentPacket for all packets sent in the test framework.

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 98 matching lines @@
 
   Callback callback_;
 };
 }  // namespace
 
 static const int kTOFExtensionId = 4;
 static const int kASTExtensionId = 5;
 
 class BitrateEstimatorTest : public test::CallTest {
  public:
-  BitrateEstimatorTest()
-      : receiver_trace_(),
-        send_transport_(),
-        receive_transport_(),
-        sender_call_(),
-        receiver_call_(),
-        receive_config_(nullptr),
-        streams_() {
-  }
+  BitrateEstimatorTest() : receive_config_(nullptr) {}
 
   virtual ~BitrateEstimatorTest() {
     EXPECT_TRUE(streams_.empty());
   }
 
   virtual void SetUp() {
     Call::Config config;
     config.voice_engine = &fake_voice_engine_;
     receiver_call_.reset(Call::Create(config));
     sender_call_.reset(Call::Create(config));
 
-    send_transport_.SetReceiver(receiver_call_->Receiver());
-    receive_transport_.SetReceiver(sender_call_->Receiver());
+    send_transport_.reset(new test::DirectTransport(sender_call_.get()));
+    send_transport_->SetReceiver(receiver_call_->Receiver());
+    receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
+    receive_transport_->SetReceiver(sender_call_->Receiver());
 
-    send_config_ = VideoSendStream::Config(&send_transport_);
+    send_config_ = VideoSendStream::Config(send_transport_.get());
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
     // Encoders will be set separately per stream.
     send_config_.encoder_settings.encoder = nullptr;
     send_config_.encoder_settings.payload_name = "FAKE";
     send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
     encoder_config_.streams = test::CreateVideoStreams(1);
 
-    receive_config_ = VideoReceiveStream::Config(&receive_transport_);
+    receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
     // receive_config_.decoders will be set by every stream separately.
     receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
     receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
     receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
   }
 
   virtual void TearDown() {
     std::for_each(streams_.begin(), streams_.end(),
         std::mem_fun(&Stream::StopSending));
 
-    send_transport_.StopSending();
-    receive_transport_.StopSending();
+    send_transport_->StopSending();
+    receive_transport_->StopSending();
 
     while (!streams_.empty()) {
       delete streams_.back();
       streams_.pop_back();
     }
 
     receiver_call_.reset();
   }
 
  protected:
@@ skipping 27 matching lines @@
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
         // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
         // the AudioReceiveStream. Every receive stream has to correspond to
         // an underlying channel id.
         receive_config.voe_channel_id = 0;
         receive_config.rtp.extensions.push_back(
             RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
         receive_config.combined_audio_video_bwe = true;
-        audio_receive_stream_ = test_->receiver_call_->CreateAudioReceiveStream(
-            receive_config);
+        audio_receive_stream_ =
+            test_->receiver_call_->CreateAudioReceiveStream(receive_config);
       } else {
         VideoReceiveStream::Decoder decoder;
         decoder.decoder = &fake_decoder_;
         decoder.payload_type =
             test_->send_config_.encoder_settings.payload_type;
         decoder.payload_name =
             test_->send_config_.encoder_settings.payload_name;
         test_->receive_config_.decoders.push_back(decoder);
         test_->receive_config_.rtp.remote_ssrc =
             test_->send_config_.rtp.ssrcs[0];
@@ skipping 37 matching lines @@
     VideoSendStream* send_stream_;
     AudioReceiveStream* audio_receive_stream_;
     VideoReceiveStream* video_receive_stream_;
     rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
 
   test::FakeVoiceEngine fake_voice_engine_;
   TraceObserver receiver_trace_;
-  test::DirectTransport send_transport_;
-  test::DirectTransport receive_transport_;
+  rtc::scoped_ptr<test::DirectTransport> send_transport_;
+  rtc::scoped_ptr<test::DirectTransport> receive_transport_;
   rtc::scoped_ptr<Call> sender_call_;
   rtc::scoped_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };
 
 static const char* kAbsSendTimeLog =
     "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
 static const char* kSingleStreamLog =
     "RemoteBitrateEstimatorSingleStream: Instantiating.";
@@ skipping 78 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_trace_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
 }
 }  // namespace webrtc