Match existing type usage better.

webrtc/modules/audio_coding/main/test/initial_delay_unittest.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.
  */
 
@@ skipping 126 matching lines @@
     for (int n = 0; n < samples; ++n) {
       in_audio_frame.data_[n] = kAmp;
     }
 
     uint32_t timestamp = 0;
     double rms = 0;
     ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
     acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
     while (rms < kAmp / 2) {
       in_audio_frame.timestamp_ = timestamp;
-      timestamp += in_audio_frame.samples_per_channel_;
+      timestamp += static_cast<uint32_t>(in_audio_frame.samples_per_channel_);
       ASSERT_GE(acm_a_->Add10MsData(in_audio_frame), 0);
       ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
       rms = FrameRms(out_audio_frame);
       ++num_frames;
     }
 
     ASSERT_GE(num_frames * 10, initial_delay_ms);
     ASSERT_LE(num_frames * 10, initial_delay_ms + 100);
   }
 
   rtc::scoped_ptr<AudioCodingModule> acm_a_;
   rtc::scoped_ptr<AudioCodingModule> acm_b_;
   Channel* channel_a2b_;
 };
 
 TEST_F(InitialPlayoutDelayTest, NbMono) { NbMono(); }
 
 TEST_F(InitialPlayoutDelayTest, WbMono) { WbMono(); }
 
 TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
 
 TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
 
 TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
 
 TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
 
 }  // namespace webrtc