Index: talk/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java |
diff --git a/talk/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java b/talk/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java |
index 90c6610a872ddd25afcaca79c9825d760e962d77..9660cc59a3b4e6f2a852c213bdb3b1972d3062b6 100644 |
--- a/talk/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java |
+++ b/talk/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java |
@@ -151,10 +151,11 @@ public class AppRTCAudioManager { |
audioManager.requestAudioFocus(null, AudioManager.STREAM_VOICE_CALL, |
AudioManager.AUDIOFOCUS_GAIN_TRANSIENT); |
- // Start by setting RINGTONE as default audio mode. The native WebRTC |
- // audio layer will switch to COMMUNICATION mode when the first streaming |
- // session starts and return to RINGTONE mode when all streaming stops. |
- audioManager.setMode(AudioManager.MODE_RINGTONE); |
+ // Start by setting MODE_IN_COMMUNICATION as default audio mode. It is |
+ // required to be in this mode when playout and/or recording starts for |
+ // best possible VoIP performance. |
+ // TODO(henrika): we migh want to start with RINGTONE mode here instead. |
+ audioManager.setMode(AudioManager.MODE_IN_COMMUNICATION); |
// Always disable microphone mute during a WebRTC call. |
setMicrophoneMute(false); |