| 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
|
|
|