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Unified Diff: webrtc/modules/audio_processing/test/audio_processing_unittest.cc

Issue 1510873004: Using buffered signal to calculate the level of echo cancellation. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: reducing buffer for nearLevel and farLevel Created 4 years, 11 months ago
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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,

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