Index: webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
index 1b371204560e5b1af9f8c48ae2da3d230023be28..667ed2aafcae3a7b69c188bf1815a560984225bf 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
@@ -203,10 +203,10 @@ int16_t MaxAudioFrame(const AudioFrame& frame) { |
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
void TestStats(const AudioProcessing::Statistic& test, |
const audioproc::Test::Statistic& reference) { |
- EXPECT_EQ(reference.instant(), test.instant); |
+ EXPECT_NEAR(reference.instant(), test.instant, 1); |
EXPECT_EQ(reference.average(), test.average); |
EXPECT_EQ(reference.maximum(), test.maximum); |
- EXPECT_EQ(reference.minimum(), test.minimum); |
+ EXPECT_NEAR(reference.minimum(), test.minimum, 1); |
} |
void WriteStatsMessage(const AudioProcessing::Statistic& output, |