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) { |