Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate."

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 126 matching lines @@
   // The default implementation just returns false.
   enum class Application { kSpeech, kAudio };
   virtual bool SetApplication(Application application);
 
   // Tells the encoder about the highest sample rate the decoder is expected to
   // use when decoding the bitstream. The encoder would typically use this
   // information to adjust the quality of the encoding. The default
   // 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
   // live as long as this encoder). Unless the returned array is empty, you may
   // not call any methods on this encoder afterwards, except for the
   // destructor. The default implementation just returns an empty array.
   // NOTE: This method is subject to change. Do not call or override it.
   virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
   ReclaimContainedEncoders();
 
   // Enables audio network adaptor. Returns true if successful.
   virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
                                          const Clock* clock);
 
   // Disables audio network adaptor.
   virtual void DisableAudioNetworkAdaptor();
 
   // Provides uplink bandwidth to this encoder to allow it to adapt.
   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);
 
   // Provides target audio bitrate to this encoder to allow it to adapt.
   virtual void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps);
 
   // Provides RTT to this encoder to allow it to adapt.
   virtual void OnReceivedRtt(int rtt_ms);
 
   // To allow encoder to adapt its frame length, it must be provided the frame
   // length range that receivers can accept.
   virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                            int max_frame_length_ms);
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_