Simplify interface for media/cast: CastTransportSenderImpl

media/cast/sender/audio_sender_unittest.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/sender/audio_sender.h"
 
 #include <stdint.h>
 #include <utility>
 
 #include "base/bind.h"
@@ skipping 17 matching lines @@
 
 namespace {
 
 void SaveOperationalStatus(OperationalStatus* out_status,
                            OperationalStatus in_status) {
   DVLOG(1) << "OperationalStatus transitioning from " << *out_status << " to "
            << in_status;
   *out_status = in_status;
 }
 
+class TransportClient : public CastTransportSender::Client {
+ public:
+  TransportClient() {}
+
+  void OnStatusChanged(CastTransportStatus status) final {
+    EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
+  };
+  void OnLoggingEventsReceived(
+      scoped_ptr<std::vector<FrameEvent>> frame_events,
+      scoped_ptr<std::vector<PacketEvent>> packet_events) final{};
+  void ProcessRtpPacket(scoped_ptr<Packet> packet) final{};
+
+  DISALLOW_COPY_AND_ASSIGN(TransportClient);
+};
+
 }  // namespace
 
 class TestPacketSender : public PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
   bool SendPacket(PacketRef packet, const base::Closure& cb) final {
     if (IsRtcpPacket(&packet->data[0], packet->data.size())) {
       ++number_of_rtcp_packets_;
     } else {
       // Check that at least one RTCP packet was sent before the first RTP
       // packet.  This confirms that the receiver will have the necessary lip
       // sync info before it has to calculate the playout time of the first
       // frame.
       if (number_of_rtp_packets_ == 0)
         EXPECT_LE(1, number_of_rtcp_packets_);
       ++number_of_rtp_packets_;
     }
     return true;
   }
 
   int64_t GetBytesSent() final { return 0; }
 
+  void StartReceiving(
+      const PacketReceiverCallbackWithStatus& packet_receiver) final {}
+
+  void StopReceiving() final {}
+
   int number_of_rtp_packets() const { return number_of_rtp_packets_; }
 
   int number_of_rtcp_packets() const { return number_of_rtcp_packets_; }
 
  private:
   int number_of_rtp_packets_;
   int number_of_rtcp_packets_;
 
   DISALLOW_COPY_AND_ASSIGN(TestPacketSender);
 };
 
 class AudioSenderTest : public ::testing::Test {
  protected:
   AudioSenderTest() {
     InitializeMediaLibrary();
     testing_clock_ = new base::SimpleTestTickClock();
     testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
     task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
     cast_environment_ =
         new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_),
                             task_runner_, task_runner_, task_runner_);
     audio_config_.codec = CODEC_AUDIO_OPUS;
     audio_config_.use_external_encoder = false;
     audio_config_.frequency = kDefaultAudioSamplingRate;
     audio_config_.channels = 2;
     audio_config_.bitrate = kDefaultAudioEncoderBitrate;
     audio_config_.rtp_payload_type = 127;
 
-    net::IPEndPoint dummy_endpoint;
-
-    transport_sender_.reset(new CastTransportSenderImpl(
-        NULL,
-        testing_clock_,
-        net::IPEndPoint(),
-        dummy_endpoint,
-        make_scoped_ptr(new base::DictionaryValue),
-        base::Bind(&UpdateCastTransportStatus),
-        BulkRawEventsCallback(),
-        base::TimeDelta(),
-        task_runner_,
-        PacketReceiverCallback(),
-        &transport_));
+    transport_ = new TestPacketSender();
+    transport_sender_.reset(
+        new CastTransportSenderImpl(testing_clock_, base::TimeDelta(),
+                                    make_scoped_ptr(new TransportClient()),
+                                    make_scoped_ptr(transport_), task_runner_));
     OperationalStatus operational_status = STATUS_UNINITIALIZED;
     audio_sender_.reset(new AudioSender(
         cast_environment_,
         audio_config_,
         base::Bind(&SaveOperationalStatus, &operational_status),
         transport_sender_.get()));
     task_runner_->RunTasks();
     CHECK_EQ(STATUS_INITIALIZED, operational_status);
   }
 
   ~AudioSenderTest() override {}
 
-  static void UpdateCastTransportStatus(CastTransportStatus status) {
-    EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
-  }
-
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
-  TestPacketSender transport_;
+  TestPacketSender* transport_;               // Owned by CastTransportSender.
   scoped_ptr<CastTransportSenderImpl> transport_sender_;
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_ptr<AudioSender> audio_sender_;
   scoped_refptr<CastEnvironment> cast_environment_;
   AudioSenderConfig audio_config_;
 };
 
 TEST_F(AudioSenderTest, Encode20ms) {
   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
   scoped_ptr<AudioBus> bus(
       TestAudioBusFactory(audio_config_.channels,
                           audio_config_.frequency,
                           TestAudioBusFactory::kMiddleANoteFreq,
                           0.5f).NextAudioBus(kDuration));
 
   audio_sender_->InsertAudio(std::move(bus), testing_clock_->NowTicks());
   task_runner_->RunTasks();
-  EXPECT_LE(1, transport_.number_of_rtp_packets());
-  EXPECT_LE(1, transport_.number_of_rtcp_packets());
+  EXPECT_LE(1, transport_->number_of_rtp_packets());
+  EXPECT_LE(1, transport_->number_of_rtcp_packets());
 }
 
 TEST_F(AudioSenderTest, RtcpTimer) {
   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
   scoped_ptr<AudioBus> bus(
       TestAudioBusFactory(audio_config_.channels,
                           audio_config_.frequency,
                           TestAudioBusFactory::kMiddleANoteFreq,
                           0.5f).NextAudioBus(kDuration));
 
   audio_sender_->InsertAudio(std::move(bus), testing_clock_->NowTicks());
   task_runner_->RunTasks();
 
   // Make sure that we send at least one RTCP packet.
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kRtcpReportIntervalMs * 3 / 2);
   testing_clock_->Advance(max_rtcp_timeout);
   task_runner_->RunTasks();
-  EXPECT_LE(1, transport_.number_of_rtp_packets());
-  EXPECT_LE(1, transport_.number_of_rtcp_packets());
+  EXPECT_LE(1, transport_->number_of_rtp_packets());
+  EXPECT_LE(1, transport_->number_of_rtcp_packets());
 }
 
 }  // namespace cast
 }  // namespace media