Reland "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 2c8d9ce5ece0bf97fa35820087ad0773f827c641..c913765fe48fd87ddc9fd45c7eb534dee016c014 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -144,16 +144,12 @@ 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);
-
+  // This is to be deprecated. Please use |OnReceivedTargetAudioBitrate|
+  // instead.
   // 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);
+  RTC_DEPRECATED 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
@@ -175,6 +171,7 @@ 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);