Switching WebRTC browser test from AudioManager::HasAudioOutputDevices() to AudioSystem interface.

content/browser/webrtc/webrtc_audio_browsertest.cc

Index: content/browser/webrtc/webrtc_audio_browsertest.cc
diff --git a/content/browser/webrtc/webrtc_audio_browsertest.cc b/content/browser/webrtc/webrtc_audio_browsertest.cc
index 082793d172988a461ef5aa781e72b7cf40ace050..5b66e50b2a372dd4cf779db4965244eefa01aa75 100644
--- a/content/browser/webrtc/webrtc_audio_browsertest.cc
+++ b/content/browser/webrtc/webrtc_audio_browsertest.cc
@@ -13,7 +13,6 @@
 #include "content/public/test/browser_test_utils.h"
 #include "content/public/test/content_browser_test_utils.h"
 #include "content/public/test/test_utils.h"
-#include "media/audio/audio_manager.h"
 #include "media/base/media_switches.h"
 #include "net/test/embedded_test_server/embedded_test_server.h"
 
@@ -44,7 +43,7 @@ class MAYBE_WebRtcAudioBrowserTest : public WebRtcContentBrowserTestBase {
   // has some special prerequisites, such that there needs to be an audio output
   // device on the executing machine).
   void MakeAudioDetectingPeerConnectionCall(const std::string& javascript) {
-    if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
+    if (!HasAudioOutputDevices()) {
       // Bots with no output devices will force the audio code into a state
       // where it doesn't manage to set either the low or high latency path.
       // This test will compute useless values in that case, so skip running on