Convert channel counts to size_t.

webrtc/modules/rtp_rtcp/source/rtp_payload_registry_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.
  */
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 
 using ::testing::Eq;
 using ::testing::Return;
 using ::testing::_;
 
 static const char* kTypicalPayloadName = "name";
-static const uint8_t kTypicalChannels = 1;
+static const size_t kTypicalChannels = 1;
 static const int kTypicalFrequency = 44000;
 static const int kTypicalRate = 32 * 1024;
 
 class RtpPayloadRegistryTest : public ::testing::Test {
  public:
   void SetUp() {
     // Note: the payload registry takes ownership of the strategy.
     mock_payload_strategy_ = new testing::NiceMock<MockRTPPayloadStrategy>();
     rtp_payload_registry_.reset(new RTPPayloadRegistry(mock_payload_strategy_));
   }
@@ skipping 44 matching lines @@
   EXPECT_EQ(returned_payload_on_heap, retrieved_payload);
 
   // Now forget about it and verify it's gone.
   EXPECT_EQ(0, rtp_payload_registry_->DeRegisterReceivePayload(payload_type));
   EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type));
 }
 
 TEST_F(RtpPayloadRegistryTest, AudioRedWorkProperly) {
   const uint8_t kRedPayloadType = 127;
   const int kRedSampleRate = 8000;
-  const int kRedChannels = 1;
+  const size_t kRedChannels = 1;
   const int kRedBitRate = 0;
 
   // This creates an audio RTP payload strategy.
   rtp_payload_registry_.reset(
       new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true)));
 
   bool new_payload_created = false;
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
                    "red", kRedPayloadType, kRedSampleRate, kRedChannels,
                    kRedBitRate, &new_payload_created));
@@ skipping 283 matching lines @@
   rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(true);
   TestRtxPacket(rtp_payload_registry_.get(), 105, 95, true);
   TestRtxPacket(rtp_payload_registry_.get(), 106, 0, false);
 }
 
 INSTANTIATE_TEST_CASE_P(TestDynamicRange,
                         RtpPayloadRegistryGenericTest,
                         testing::Range(96, 127 + 1));
 
 }  // namespace webrtc