Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/opus/opus_fec_test.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.
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::string;
 using ::std::tr1::tuple;
 using ::std::tr1::get;
 using ::testing::TestWithParam;
 
 namespace webrtc {
 
 // Define coding parameter as <channels, bit_rate, filename, extension>.
-typedef tuple<int, int, string, string> coding_param;
+typedef tuple<size_t, int, string, string> coding_param;
 typedef struct mode mode;
 
 struct mode {
   bool fec;
   uint8_t target_packet_loss_rate;
 };
 
 const int kOpusBlockDurationMs = 20;
 const int kOpusSamplingKhz = 48;
 
 class OpusFecTest : public TestWithParam<coding_param> {
  protected:
   OpusFecTest();
 
   virtual void SetUp();
   virtual void TearDown();
 
   virtual void EncodeABlock();
 
   virtual void DecodeABlock(bool lost_previous, bool lost_current);
 
   int block_duration_ms_;
   int sampling_khz_;
   size_t block_length_sample_;
 
-  int channels_;
+  size_t channels_;
   int bit_rate_;
 
   size_t data_pointer_;
   size_t loop_length_samples_;
   size_t max_bytes_;
   size_t encoded_bytes_;
 
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
 
   string in_filename_;
 
   rtc::scoped_ptr<int16_t[]> in_data_;
   rtc::scoped_ptr<int16_t[]> out_data_;
   rtc::scoped_ptr<uint8_t[]> bit_stream_;
 };
 
 void OpusFecTest::SetUp() {
   channels_ = get<0>(GetParam());
   bit_rate_ = get<1>(GetParam());
-  printf("Coding %d channel signal at %d bps.\n", channels_, bit_rate_);
+  printf("Coding %" PRIuS " channel signal at %d bps.\n", channels_, bit_rate_);
 
   in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
 
   FILE* fp = fopen(in_filename_.c_str(), "rb");
   ASSERT_FALSE(fp == NULL);
 
   // Obtain file size.
   fseek(fp, 0, SEEK_END);
   loop_length_samples_ = ftell(fp) / sizeof(int16_t);
   rewind(fp);
@@ skipping 149 matching lines @@
      ::std::tr1::make_tuple(1, 32000, string("audio_coding/testfile32kHz"),
                             string("pcm")),
      ::std::tr1::make_tuple(2, 64000, string("audio_coding/teststereo32kHz"),
                             string("pcm"))};
 
 // 64 kbps, stereo
 INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest,
                         ::testing::ValuesIn(param_set));
 
 }  // namespace webrtc