Index: webrtc/modules/audio_processing/audio_processing_impl_unittest.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc |
index ed20daaa61ed27ceb2677fde95eecd4a5805b093..965e7eb60b72807e41c7161c589329472115a98c 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc |
@@ -65,12 +65,10 @@ TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { |
frame.num_channels_ = 2; |
EXPECT_NOERR(mock.AnalyzeReverseStream(&frame)); |
- // A new sample rate passed to AnalyzeReverseStream should be an error and |
- // not cause an init. |
+ // A new sample rate passed to AnalyzeReverseStream should cause an init. |
SetFrameSampleRate(&frame, 16000); |
- EXPECT_CALL(mock, InitializeLocked()) |
- .Times(0); |
- EXPECT_EQ(mock.kBadSampleRateError, mock.AnalyzeReverseStream(&frame)); |
+ EXPECT_CALL(mock, InitializeLocked()).Times(1); |
+ EXPECT_NOERR(mock.AnalyzeReverseStream(&frame)); |
} |
} // namespace webrtc |