[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

 /*
  *  Copyright (c) 2012 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 47 matching lines @@
     return 0;
   }
 };
 
 class RTPCallback : public NullRtpFeedback {
  public:
   int32_t OnInitializeDecoder(const int32_t id,
                               const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
-                              const uint8_t channels,
+                              const size_t channels,
                               const uint32_t rate) override {
     if (payloadType == 96) {
       EXPECT_EQ(test_rate, rate) <<
           "The rate should be 64K for this payloadType";
     }
     return 0;
   }
 };
 
 class RtpRtcpAudioTest : public ::testing::Test {
@@ skipping 268 matching lines @@
   }
   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
 
   for (;timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }