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; |
/////////////////////////////////////////////////////////////////////////// |