| Index: webrtc/modules/audio_coding/include/audio_coding_module.h | 
| diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h | 
| index fc8ae1ed513cb5098c2406f09a4d63a2e1f9720a..615e8d949658e51b1b5ed08f418391ddf1a3ae13 100644 | 
| --- a/webrtc/modules/audio_coding/include/audio_coding_module.h | 
| +++ b/webrtc/modules/audio_coding/include/audio_coding_module.h | 
| @@ -252,6 +252,9 @@ class AudioCodingModule { | 
| /////////////////////////////////////////////////////////////////////////// | 
| // Sets the bitrate to the specified value in bits/sec. If the value is not | 
| // supported by the codec, it will choose another appropriate value. | 
| +  // | 
| +  // This is only used in audio_coding_module_unittest_oldapi.cc. | 
| +  // TODO(minyue): Remove it when possible. | 
| virtual void SetBitRate(int bitrate_bps) = 0; | 
|  | 
| // int32_t RegisterTransportCallback() | 
| @@ -371,6 +374,8 @@ class AudioCodingModule { | 
| //   -1 if failed to set packet loss rate, | 
| //   0 if succeeded. | 
| // | 
| +  // This is only used in audio_coding_module_unittest_oldapi.cc. | 
| +  // TODO(minyue): Remove it when possible. | 
| virtual int SetPacketLossRate(int packet_loss_rate) = 0; | 
|  | 
| /////////////////////////////////////////////////////////////////////////// | 
|  |