Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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.
  */
 
@@ skipping 1504 matching lines @@
 
   // Pull audio once and verify that the output is speech again.
   ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
   EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
             PlayoutTimestamp());
 }
 
 uint32_t NetEqDecodingTest::PlayoutTimestamp() {
-  uint32_t playout_timestamp = 0;
-  EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&playout_timestamp));
-  return playout_timestamp;
+  rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+  EXPECT_TRUE(playout_timestamp);
+  return playout_timestamp ? *playout_timestamp : 0;

[minyue-webrtc, 2016/04/04 15:53:50]

consider using value_or()

[hlundin-webrtc, 2016/04/04 21:02:54]

Nice. I didn't know about that one. Done.

 }
 
 TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
 
 TEST_F(NetEqDecodingTest, CngFirst) {
   uint16_t seq_no = 0;
   uint32_t timestamp = 0;
   const int kFrameSizeMs = 10;
   const int kSampleRateKhz = 16;
   const int kSamples = kFrameSizeMs * kSampleRateKhz;
@@ skipping 25 matching lines @@
     ++seq_no;
     timestamp += kSamples;
 
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }
-
 }  // namespace webrtc