Index: webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h |
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h |
index ee7a2f1340beff6c85d2fa979d460df16c3fe112..559c3fc7a0e2211ab423fb08a656cd065d8e1030 100644 |
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h |
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h |
@@ -19,35 +19,6 @@ |
namespace webrtc { |
/////////////////////////////////////////////////////////////////////////// |
-// enum AudioPlayoutMode |
-// An enumerator for different playout modes. |
-// |
-// -voice : This is the standard mode for VoIP calls. The trade-off |
-// between low delay and jitter robustness is optimized |
-// for high-quality two-way communication. |
-// NetEQs packet loss concealment and signal processing |
-// capabilities are fully employed. |
-// -fax : The fax mode is optimized for decodability of fax signals |
-// rather than for perceived audio quality. When this mode |
-// is selected, NetEQ will do as few delay changes as possible, |
-// trying to maintain a high and constant delay. Meanwhile, |
-// the packet loss concealment efforts are reduced. |
-// |
-// -streaming : In the case of one-way communication such as passive |
-// conference participant, a webinar, or a streaming application, |
-// this mode can be used to improve the jitter robustness at |
-// the cost of increased delay. |
-// -off : Turns off most of NetEQ's features. Stuffs zeros for lost |
-// packets and during buffer increases. |
-// |
-enum AudioPlayoutMode { |
- voice = 0, |
- fax = 1, |
- streaming = 2, |
- off = 3, |
-}; |
- |
-/////////////////////////////////////////////////////////////////////////// |
// enum ACMSpeechType |
// An enumerator for possible labels of a decoded frame. |
// |