Cast: Removed unnecessary ref counters.

media/cast/audio_receiver/audio_decoder_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/memory/ref_counted.h"
 #include "media/cast/audio_receiver/audio_decoder.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
 namespace cast {
 
 class AudioDecoderTest : public ::testing::Test {
  protected:
   AudioDecoderTest() {}
   virtual ~AudioDecoderTest() {}
 
   void Configure(const AudioReceiverConfig& audio_config) {
-    audio_decoder_ = new AudioDecoder(audio_config);
+    audio_decoder_.reset(new AudioDecoder(audio_config));
   }
 
-  scoped_refptr<AudioDecoder> audio_decoder_;
+  scoped_ptr<AudioDecoder> audio_decoder_;
 };
 
 TEST_F(AudioDecoderTest, Pcm16MonoNoResampleOnePacket) {
   AudioReceiverConfig audio_config;
   audio_config.rtp_payload_type = 127;
   audio_config.frequency = 16000;
   audio_config.channels = 1;
   audio_config.codec = kPcm16;
   audio_config.use_external_decoder = false;
   Configure(audio_config);
 
   RtpCastHeader rtp_header;
   rtp_header.webrtc.header.payloadType = 127;
   rtp_header.webrtc.header.sequenceNumber = 1234;
   rtp_header.webrtc.header.timestamp = 0x87654321;
   rtp_header.webrtc.header.ssrc = 0x12345678;
   rtp_header.webrtc.header.paddingLength = 0;
   rtp_header.webrtc.header.headerLength = 12;
   rtp_header.webrtc.type.Audio.channel = 1;
   rtp_header.webrtc.type.Audio.isCNG = false;
 
   std::vector<int16> payload(640, 0x1234);
+  int number_of_10ms_blocks = 4;
+  int desired_frequency = 16000;
+  PcmAudioFrame audio_frame;
+  uint32 rtp_timestamp;
+
+  EXPECT_FALSE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
+                                                desired_frequency,
+                                                &audio_frame,
+                                                &rtp_timestamp));
 
   uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
   size_t payload_size = payload.size() * sizeof(int16);
 
   audio_decoder_->IncomingParsedRtpPacket(payload_data,
       payload_size, rtp_header);
 
-  int number_of_10ms_blocks = 4;
-  int desired_frequency = 16000;
-  PcmAudioFrame audio_frame;
-  uint32 rtp_timestamp;
-
   EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
                                                desired_frequency,
                                                &audio_frame,
                                                &rtp_timestamp));
-
   EXPECT_EQ(1, audio_frame.channels);
   EXPECT_EQ(16000, audio_frame.frequency);
   EXPECT_EQ(640ul, audio_frame.samples.size());
   // First 10 samples per channel are 0 from NetEq.
   for (size_t i = 10; i < audio_frame.samples.size(); ++i) {
     EXPECT_EQ(0x3412, audio_frame.samples[i]);
   }
 }
 
 TEST_F(AudioDecoderTest, Pcm16StereoNoResampleTwoPackets) {
@@ skipping 106 matching lines @@
   // Resampling makes the variance worse.
   for (size_t i = 100 * audio_config.channels; i < audio_frame.samples.size();
       ++i) {
     EXPECT_NEAR(0x3412, audio_frame.samples[i], 400);
     if (0x3412 == audio_frame.samples[i])  count++;
   }
 }
 
 }  // namespace cast
 }  // namespace media