Index: content/renderer/media/media_stream_audio_processor_unittest.cc |
diff --git a/content/renderer/media/media_stream_audio_processor_unittest.cc b/content/renderer/media/media_stream_audio_processor_unittest.cc |
index fb2f0ab540bceb5ee364b200cbed22b1b583fa9a..6262c43450f51ac44d144bfd9decdf3ac2f3395a 100644 |
--- a/content/renderer/media/media_stream_audio_processor_unittest.cc |
+++ b/content/renderer/media/media_stream_audio_processor_unittest.cc |
@@ -156,8 +156,9 @@ TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) { |
new WebRtcAudioDeviceImpl()); |
scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
- params_, constraints, 0, webrtc_audio_device.get())); |
+ constraints, 0, webrtc_audio_device.get())); |
EXPECT_FALSE(audio_processor->has_audio_processing()); |
+ audio_processor->OnCaptureFormatChanged(params_); |
ProcessDataAndVerifyFormat(audio_processor, |
params_.sample_rate(), |
@@ -177,8 +178,9 @@ TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { |
new WebRtcAudioDeviceImpl()); |
scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
- params_, constraints, 0, webrtc_audio_device.get())); |
+ constraints, 0, webrtc_audio_device.get())); |
EXPECT_TRUE(audio_processor->has_audio_processing()); |
+ audio_processor->OnCaptureFormatChanged(params_); |
VerifyDefaultComponents(audio_processor); |
ProcessDataAndVerifyFormat(audio_processor, |