Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc

Index: webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index 07a15ff5787b056b3dfe1597609ac829ec8e3d3c..069e06fd9b0e1fbf60ff2753de9f1108b56a6152 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -11,6 +11,7 @@
 #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;
@@ -37,7 +38,7 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
 }
 
 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());
@@ -99,7 +100,7 @@ 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) {