Match existing type usage better.

webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc

 /*
  *  Copyright (c) 2014 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 74 matching lines @@
     clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
     CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_));
     if (input_frame_.num_channels_ > 1) {
       InputAudioFile::DuplicateInterleaved(input_frame_.data_,
                                            input_block_size_samples_,
                                            input_frame_.num_channels_,
                                            input_frame_.data_);
     }
     data_to_send_ = false;
     CHECK_GE(acm_->Add10MsData(input_frame_), 0);
-    input_frame_.timestamp_ += input_block_size_samples_;
+    input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
     if (data_to_send_) {
       // Encoded packet received.
       return CreatePacket();
     }
   }
   // Test ended.
   return NULL;
 }
 
 // This method receives the callback from ACM when a new packet is produced.
@@ skipping 41 matching lines @@
          last_payload_vec_.size());
   Packet* packet =
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
   assert(packet);
   assert(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc