Only turn on the audio processing by default for MEDIA_DEVICE_AUDIO_CAPTURE

content/renderer/media/media_stream_audio_processor_unittest.cc

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..f04b76c21053ca253d1f20366f08c3de75681368 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>(
-          params_, constraints, 0, webrtc_audio_device.get()));
+          params_, constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE,
+          webrtc_audio_device.get()));
   EXPECT_FALSE(audio_processor->has_audio_processing());
 
   ProcessDataAndVerifyFormat(audio_processor,
@@ -177,7 +179,8 @@ TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
-          params_, constraints, 0, webrtc_audio_device.get()));
+          params_, constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE,
+          webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
   VerifyDefaultComponents(audio_processor);
 
@@ -190,4 +193,35 @@ TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
   audio_processor = NULL;
 }
 
+TEST_F(MediaStreamAudioProcessorTest, VerifyCastWithoutAudioProcessing) {

[Alpha Left Google, 2014/03/07 21:27:29]

Should be VerifyTabCapture...

[no longer working on chromium, 2014/03/07 21:56:52]

But we only verified MEDIA_LOOPBACK_AUDIO_CAPTURE from line 217...

[Alpha Left Google, 2014/03/07 22:09:42]

Cast is really using tab capture. This enum is for desktop capture.

[no longer working on chromium, 2014/03/09 15:12:57]

Done.

+  // 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>(
+          params_, constraints, 0, MEDIA_TAB_AUDIO_CAPTURE,
+          webrtc_audio_device.get()));
+  EXPECT_FALSE(audio_processor->has_audio_processing());
+
+  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>(
+          params_, 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