Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

 /*
  *  Copyright (c) 2012 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 384 matching lines @@
   }
   void Reset() override { return enc_->Reset(); }
   bool SetFec(bool enable) override { return enc_->SetFec(enable); }
   bool SetDtx(bool enable) override { return enc_->SetDtx(enable); }
   bool SetApplication(Application application) override {
     return enc_->SetApplication(application);
   }
   void SetMaxPlaybackRate(int frequency_hz) override {
     return enc_->SetMaxPlaybackRate(frequency_hz);
   }
-  void SetProjectedPacketLossRate(double fraction) override {
-    return enc_->SetProjectedPacketLossRate(fraction);
-  }
-  void SetTargetBitrate(int target_bps) override {
-    return enc_->SetTargetBitrate(target_bps);
-  }
 
  private:
   AudioEncoder* enc_;
 };
 
 // Return false on error.
 bool CreateSpeechEncoderIfNecessary(EncoderFactory* ef) {
   auto* sp = ef->codec_manager.GetStackParams();
   if (sp->speech_encoder) {
     // Do nothing; we already have a speech encoder.
@@ skipping 221 matching lines @@
                  "SendFrequency Failed, no codec is registered");
     return -1;
   }
 
   return encoder_stack_->SampleRateHz();
 }
 
 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
   rtc::CritScope lock(&acm_crit_sect_);
   if (encoder_stack_) {
-    encoder_stack_->SetTargetBitrate(bitrate_bps);
+    encoder_stack_->OnReceivedTargetAudioBitrate(bitrate_bps);
   }
 }
 
 // Register a transport callback which will be called to deliver
 // the encoded buffers.
 int AudioCodingModuleImpl::RegisterTransportCallback(
     AudioPacketizationCallback* transport) {
   rtc::CritScope lock(&callback_crit_sect_);
   packetization_callback_ = transport;
   return 0;
@@ skipping 231 matching lines @@
     return sp->use_codec_fec ? 0 : -1;
   } else {
     RTC_DCHECK(!sp->use_codec_fec);
     return 0;
   }
 }
 
 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
   rtc::CritScope lock(&acm_crit_sect_);
   if (HaveValidEncoder("SetPacketLossRate")) {
-    encoder_stack_->SetProjectedPacketLossRate(loss_rate / 100.0);
+    encoder_stack_->OnReceivedUplinkPacketLossFraction(loss_rate / 100.0);
   }
   return 0;
 }
 
 /////////////////////////////////////////
 //   (VAD) Voice Activity Detection
 //
 int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
                                   bool enable_vad,
                                   ACMVADMode mode) {
@@ skipping 441 matching lines @@
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
                  "Invalid codec setting");
   return valid;
 }
 
 }  // namespace webrtc