Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 6f809e852e63b849519956dd1995554e40608cd9..0aa667effb156da2e9bd3ac9c6cf9dc26506924f 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -197,23 +197,6 @@ void AudioEncoderOpus::SetMaxPlaybackRate(int frequency_hz) {
   RTC_CHECK(RecreateEncoderInstance(conf));
 }
 
-void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
-  double opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_);
-  if (packet_loss_rate_ != opt_loss_rate) {
-    packet_loss_rate_ = opt_loss_rate;
-    RTC_CHECK_EQ(
-        0, WebRtcOpus_SetPacketLossRate(
-               inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
-  }
-}
-
-void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
-  config_.bitrate_bps = rtc::Optional<int>(
-      std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps));
-  RTC_DCHECK(config_.IsOk());
-  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
-}
-
 bool AudioEncoderOpus::EnableAudioNetworkAdaptor(
     const std::string& config_string,
     const Clock* clock) {
@@ -234,10 +217,8 @@ void AudioEncoderOpus::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {
 
 void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
-  if (!audio_network_adaptor_) {
-    return AudioEncoder::OnReceivedTargetAudioBitrate(
-        uplink_packet_loss_fraction);
-  }
+  if (!audio_network_adaptor_)
+    SetProjectedPacketLossRate(uplink_packet_loss_fraction);

[kwiberg-webrtc, 2016/10/20 21:46:23]

Shouldn't there be an early retirn here? Or an else?

[minyue-webrtc, 2016/11/08 13:37:24]

Yes, it should early return :)

This is self-solved after rebasing.

   audio_network_adaptor_->SetUplinkPacketLossFraction(
       uplink_packet_loss_fraction);
   ApplyAudioNetworkAdaptor();
@@ -246,7 +227,7 @@ void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
 void AudioEncoderOpus::OnReceivedTargetAudioBitrate(
     int target_audio_bitrate_bps) {
   if (!audio_network_adaptor_)
-    return AudioEncoder::OnReceivedTargetAudioBitrate(target_audio_bitrate_bps);
+    return SetTargetBitrate(target_audio_bitrate_bps);
   audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
   ApplyAudioNetworkAdaptor();
 }
@@ -381,6 +362,23 @@ void AudioEncoderOpus::SetNumChannelsToEncode(size_t num_channels_to_encode) {
   num_channels_to_encode_ = num_channels_to_encode;
 }
 
+void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
+  config_.bitrate_bps = rtc::Optional<int>(
+      std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps));
+  RTC_DCHECK(config_.IsOk());
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
+}
+
+void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
+  double opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_);
+  if (packet_loss_rate_ != opt_loss_rate) {
+    packet_loss_rate_ = opt_loss_rate;
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetPacketLossRate(
+               inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  }
+}
+
 void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
   auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
   // |audio_network_adaptor_| is supposed to be configured to output all