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 3ffed874444b70f120dfa58b20db894fd34e707e..53f667d50d7815287ae1b873606598bd02120149 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
@@ -204,10 +204,10 @@ int16_t MaxAudioFrame(const AudioFrame& frame) { |
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
void TestStats(const AudioProcessing::Statistic& test, |
const audioproc::Test::Statistic& reference) { |
- EXPECT_NEAR(reference.instant(), test.instant, 1); |
- EXPECT_EQ(reference.average(), test.average); |
- EXPECT_EQ(reference.maximum(), test.maximum); |
- EXPECT_NEAR(reference.minimum(), test.minimum, 1); |
+ EXPECT_NEAR(reference.instant(), test.instant, 2); |
+ EXPECT_NEAR(reference.average(), test.average, 2); |
+ EXPECT_NEAR(reference.maximum(), test.maximum, 3); |
+ EXPECT_NEAR(reference.minimum(), test.minimum, 2); |
} |
void WriteStatsMessage(const AudioProcessing::Statistic& output, |