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audio_processing_unittest: unbreak clang compilation.

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BUG=2735
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5313 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
fischman@webrtc.org 2013-12-17 23:46:39 +00:00
parent 179908c81c
commit f8be8df33a
1 changed files with 2 additions and 0 deletions
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2
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
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@ -146,6 +146,7 @@ int16_t MaxAudioFrame(const AudioFrame& frame) {
return max_data; return max_data;
} }
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
void TestStats(const AudioProcessing::Statistic& test, void TestStats(const AudioProcessing::Statistic& test,
const webrtc::audioproc::Test::Statistic& reference) { const webrtc::audioproc::Test::Statistic& reference) {
EXPECT_EQ(reference.instant(), test.instant); EXPECT_EQ(reference.instant(), test.instant);
@ -161,6 +162,7 @@ void WriteStatsMessage(const AudioProcessing::Statistic& output,
message->set_maximum(output.maximum); message->set_maximum(output.maximum);
message->set_minimum(output.minimum); message->set_minimum(output.minimum);
} }
#endif
void WriteMessageLiteToFile(const std::string filename, void WriteMessageLiteToFile(const std::string filename,
const ::google::protobuf::MessageLite& message) { const ::google::protobuf::MessageLite& message) {