Fixes for memory and threading issues in cast

media/cast/audio_receiver/audio_receiver_unittest.cc

 // Copyright 2013 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 "base/bind.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/audio_receiver/audio_receiver.h"
 #include "media/cast/cast_defines.h"
@@ skipping 57 matching lines @@
 class AudioReceiverTest : public ::testing::Test {
  protected:
   AudioReceiverTest() {
     // Configure the audio receiver to use PCM16.
     audio_config_.rtp_payload_type = 127;
     audio_config_.frequency = 16000;
     audio_config_.channels = 1;
     audio_config_.codec = transport::kPcm16;
     audio_config_.use_external_decoder = false;
     audio_config_.feedback_ssrc = 1234;
-    testing_clock_.Advance(
+    testing_clock_ = new base::SimpleTestTickClock();
+    testing_clock_->Advance(
         base::TimeDelta::FromMilliseconds(kStartMillisecond));
-    task_runner_ = new test::FakeTaskRunner(&testing_clock_);
-    cast_environment_ = new CastEnvironment(&testing_clock_, task_runner_,
+    task_runner_ = new test::FakeTaskRunner(testing_clock_);
+    cast_environment_ = new CastEnvironment(
+        scoped_ptr<base::TickClock>(testing_clock_).Pass(), task_runner_,
         task_runner_, task_runner_, task_runner_, task_runner_,
         task_runner_, GetDefaultCastReceiverLoggingConfig());
     test_audio_encoder_callback_ = new TestAudioEncoderCallback();
   }
 
   void Configure(bool use_external_decoder) {
     audio_config_.use_external_decoder = use_external_decoder;
     receiver_.reset(new PeerAudioReceiver(cast_environment_, audio_config_,
                                           &mock_transport_));
   }
 
   virtual ~AudioReceiverTest() {}
 
   static void DummyDeletePacket(const uint8* packet) {};
 
   virtual void SetUp() {
     payload_.assign(kMaxIpPacketSize, 0);
     rtp_header_.is_key_frame = true;
     rtp_header_.frame_id = 0;
     rtp_header_.packet_id = 0;
     rtp_header_.max_packet_id = 0;
     rtp_header_.is_reference = false;
     rtp_header_.reference_frame_id = 0;
     rtp_header_.webrtc.header.timestamp = 0;
   }
 
   AudioReceiverConfig audio_config_;
   std::vector<uint8> payload_;
   RtpCastHeader rtp_header_;
-  base::SimpleTestTickClock testing_clock_;
+  base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment.
   transport::MockPacedPacketSender mock_transport_;
   scoped_refptr<test::FakeTaskRunner> task_runner_;
   scoped_ptr<PeerAudioReceiver> receiver_;
   scoped_refptr<CastEnvironment> cast_environment_;
   scoped_refptr<TestAudioEncoderCallback> test_audio_encoder_callback_;
 };
 
 TEST_F(AudioReceiverTest, GetOnePacketEncodedframe) {
   Configure(true);
   EXPECT_CALL(mock_transport_, SendRtcpPacket(testing::_)).Times(1);
 
   receiver_->IncomingParsedRtpPacket(payload_.data(),
       payload_.size(), rtp_header_);
   transport::EncodedAudioFrame audio_frame;
   base::TimeTicks playout_time;
-  test_audio_encoder_callback_->SetExpectedResult(0, testing_clock_.NowTicks());
+  test_audio_encoder_callback_->SetExpectedResult(
+      0, testing_clock_->NowTicks());
 
   AudioFrameEncodedCallback frame_encoded_callback =
       base::Bind(&TestAudioEncoderCallback::DeliverEncodedAudioFrame,
                  test_audio_encoder_callback_.get());
 
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
 }
 
 TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
   Configure(true);
   EXPECT_CALL(mock_transport_, SendRtcpPacket(testing::_)).WillRepeatedly(
       testing::Return(true));
 
   AudioFrameEncodedCallback frame_encoded_callback =
       base::Bind(&TestAudioEncoderCallback::DeliverEncodedAudioFrame,
                  test_audio_encoder_callback_.get());
 
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
 
   receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
                                      rtp_header_);
 
   transport::EncodedAudioFrame audio_frame;
   base::TimeTicks playout_time;
-  test_audio_encoder_callback_->SetExpectedResult(0, testing_clock_.NowTicks());
+  test_audio_encoder_callback_->SetExpectedResult(
+      0, testing_clock_->NowTicks());
 
   task_runner_->RunTasks();
   EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
 
   TestRtcpPacketBuilder rtcp_packet;
 
   uint32 ntp_high;
   uint32 ntp_low;
-  ConvertTimeTicksToNtp(testing_clock_.NowTicks(), &ntp_high, &ntp_low);
+  ConvertTimeTicksToNtp(testing_clock_->NowTicks(), &ntp_high, &ntp_low);
   rtcp_packet.AddSrWithNtp(audio_config_.feedback_ssrc, ntp_high, ntp_low,
       rtp_header_.webrtc.header.timestamp);
 
-  testing_clock_.Advance(base::TimeDelta::FromMilliseconds(20));
+  testing_clock_->Advance(base::TimeDelta::FromMilliseconds(20));
 
   receiver_->IncomingPacket(rtcp_packet.Packet(), rtcp_packet.Length(),
       base::Bind(AudioReceiverTest::DummyDeletePacket, rtcp_packet.Packet()));
 
   // Make sure that we are not continuous and that the RTP timestamp represent a
   // time in the future.
   rtp_header_.is_key_frame = false;
   rtp_header_.frame_id = 2;
   rtp_header_.is_reference = true;
   rtp_header_.reference_frame_id = 0;
   rtp_header_.webrtc.header.timestamp = 960;
   test_audio_encoder_callback_->SetExpectedResult(2,
-      testing_clock_.NowTicks() + base::TimeDelta::FromMilliseconds(100));
+      testing_clock_->NowTicks() + base::TimeDelta::FromMilliseconds(100));
 
   receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
                                      rtp_header_);
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
 
   // Frame 2 should not come out at this point in time.
   EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
 
   // Through on one more pending callback.
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
 
-  testing_clock_.Advance(base::TimeDelta::FromMilliseconds(100));
+  testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
 
   task_runner_->RunTasks();
   EXPECT_EQ(2, test_audio_encoder_callback_->number_times_called());
 
-  test_audio_encoder_callback_->SetExpectedResult(3, testing_clock_.NowTicks());
+  test_audio_encoder_callback_->SetExpectedResult(
+      3, testing_clock_->NowTicks());
 
   // Through on one more pending audio frame.
   rtp_header_.frame_id = 3;
   rtp_header_.is_reference = false;
   rtp_header_.reference_frame_id = 0;
   rtp_header_.webrtc.header.timestamp = 1280;
   receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
                                      rtp_header_);
 
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(3, test_audio_encoder_callback_->number_times_called());
 }
 
 // TODO(mikhal): Add encoded frames.
 TEST_F(AudioReceiverTest, GetRawFrame) {
 }
 
 }  // namespace cast
 }  // namespace media