Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_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 "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::tr1::get;
 
 namespace webrtc {
 
 AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
                                          int input_sampling_khz,
                                          int output_sampling_khz)
     : block_duration_ms_(block_duration_ms),
       input_sampling_khz_(input_sampling_khz),
       output_sampling_khz_(output_sampling_khz),
       input_length_sample_(
           static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
       output_length_sample_(
           static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
       data_pointer_(0),
       loop_length_samples_(0),
       max_bytes_(0),
       encoded_bytes_(0),
       encoding_time_ms_(0.0),
       decoding_time_ms_(0.0),
       out_file_(NULL) {
 }
 
 void AudioCodecSpeedTest::SetUp() {
-  channels_ = get<0>(GetParam());
+  channels_ = static_cast<size_t>(get<0>(GetParam()));

[minyue-webrtc, 2015/12/29 10:42:47]

can change the def of param 0 if you like

[Peter Kasting, 2016/01/08 02:45:07]

Done.

   bit_rate_ = get<1>(GetParam());
   in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
   save_out_data_ = get<4>(GetParam());
 
   FILE* fp = fopen(in_filename_.c_str(), "rb");
   assert(fp != NULL);
 
   // Obtain file size.
   fseek(fp, 0, SEEK_END);
   loop_length_samples_ = ftell(fp) / sizeof(int16_t);
@@ skipping 41 matching lines @@
 void AudioCodecSpeedTest::TearDown() {
   if (save_out_data_) {
     fclose(out_file_);
   }
 }
 
 void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
   size_t time_now_ms = 0;
   float time_ms;
 
-  printf("Coding %d kHz-sampled %d-channel audio at %d bps ...\n",
+  printf("Coding %d kHz-sampled %" PRIuS "-channel audio at %d bps ...\n",
          input_sampling_khz_, channels_, bit_rate_);
 
   while (time_now_ms < audio_duration_sec * 1000) {
     // Encode & decode.
     time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0],
                            max_bytes_, &encoded_bytes_);
     encoding_time_ms_ += time_ms;
     time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
     decoding_time_ms_ += time_ms;
     if (save_out_data_) {
       fwrite(&out_data_[0], sizeof(int16_t),
              output_length_sample_ * channels_, out_file_);
     }
     data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
         loop_length_samples_;
     time_now_ms += block_duration_ms_;
   }
 
   printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n",
          (encoding_time_ms_ / audio_duration_sec) / 10.0,
          (decoding_time_ms_ / audio_duration_sec) / 10.0);
 }
 
 }  // namespace webrtc