Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 19dc3327578ec3c1853da96599052612f82b3e77..3879b267401d95d9ef39cb67fca2489425d1dbbf 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -144,17 +144,6 @@ class AudioEncoder {
   // implementation does nothing.
   virtual void SetMaxPlaybackRate(int frequency_hz);
 
-  // Tells the encoder what the projected packet loss rate is. The rate is in
-  // the range [0.0, 1.0]. The encoder would typically use this information to
-  // adjust channel coding efforts, such as FEC. The default implementation
-  // does nothing.
-  virtual void SetProjectedPacketLossRate(double fraction);
-
-  // Tells the encoder what average bitrate we'd like it to produce. The
-  // encoder is free to adjust or disregard the given bitrate (the default
-  // implementation does the latter).
-  virtual void SetTargetBitrate(int target_bps);
-
   // Causes this encoder to let go of any other encoders it contains, and
   // returns a pointer to an array where they are stored (which is required to
   // live as long as this encoder). Unless the returned array is empty, you may
@@ -175,6 +164,7 @@ class AudioEncoder {
   virtual void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps);
 
   // Provides uplink packet loss fraction to this encoder to allow it to adapt.
+  // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);