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 6262c43450f51ac44d144bfd9decdf3ac2f3395a..3f46406c65366c42788536a8d6f2e943c425c7f8 100644 |
--- a/content/renderer/media/media_stream_audio_processor_unittest.cc |
+++ b/content/renderer/media/media_stream_audio_processor_unittest.cc |
@@ -9,6 +9,7 @@ |
#include "base/path_service.h" |
#include "base/time/time.h" |
#include "content/public/common/content_switches.h" |
+#include "content/public/common/media_stream_request.h" |
#include "content/renderer/media/media_stream_audio_processor.h" |
#include "media/audio/audio_parameters.h" |
#include "media/base/audio_bus.h" |
@@ -156,7 +157,8 @@ TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) { |
new WebRtcAudioDeviceImpl()); |
scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
- constraints, 0, webrtc_audio_device.get())); |
+ constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE, |
+ webrtc_audio_device.get())); |
EXPECT_FALSE(audio_processor->has_audio_processing()); |
audio_processor->OnCaptureFormatChanged(params_); |
@@ -178,7 +180,8 @@ TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { |
new WebRtcAudioDeviceImpl()); |
scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
- constraints, 0, webrtc_audio_device.get())); |
+ constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE, |
+ webrtc_audio_device.get())); |
EXPECT_TRUE(audio_processor->has_audio_processing()); |
audio_processor->OnCaptureFormatChanged(params_); |
VerifyDefaultComponents(audio_processor); |
@@ -192,4 +195,36 @@ TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { |
audio_processor = NULL; |
} |
+TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) { |
+ // Setup the audio processor with enabling the flag. |
+ CommandLine::ForCurrentProcess()->AppendSwitch( |
+ switches::kEnableAudioTrackProcessing); |
+ blink::WebMediaConstraints constraints; |
+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( |
+ new WebRtcAudioDeviceImpl()); |
+ // Create MediaStreamAudioProcessor instance for MEDIA_TAB_AUDIO_CAPTURE type. |
+ scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
+ new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
+ constraints, 0, MEDIA_TAB_AUDIO_CAPTURE, |
+ webrtc_audio_device.get())); |
+ EXPECT_FALSE(audio_processor->has_audio_processing()); |
+ audio_processor->OnCaptureFormatChanged(params_); |
+ |
+ ProcessDataAndVerifyFormat(audio_processor, |
+ params_.sample_rate(), |
+ params_.channels(), |
+ params_.sample_rate() / 100); |
+ |
+ // Create MediaStreamAudioProcessor instance for MEDIA_LOOPBACK_AUDIO_CAPTURE. |
+ audio_processor = |
+ new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
+ constraints, 0, MEDIA_LOOPBACK_AUDIO_CAPTURE, |
+ webrtc_audio_device.get()); |
+ EXPECT_FALSE(audio_processor->has_audio_processing()); |
+ |
+ // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives |
+ // |audio_processor|. |
+ audio_processor = NULL; |
+} |
+ |
} // namespace content |