Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.

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

 /*
  *  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 35 matching lines @@
   config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
   return config;
 }
 
 // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
 // the input loss rate rounded down to various levels, because a robustly good
 // audio quality is achieved by lowering the packet loss down.
 // Additionally, to prevent toggling, margins are used, i.e., when jumping to
 // a loss rate from below, a higher threshold is used than jumping to the same
 // level from above.
-double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) {
-  RTC_DCHECK_GE(new_loss_rate, 0.0);
-  RTC_DCHECK_LE(new_loss_rate, 1.0);
-  RTC_DCHECK_GE(old_loss_rate, 0.0);
-  RTC_DCHECK_LE(old_loss_rate, 1.0);
-  const double kPacketLossRate20 = 0.20;
-  const double kPacketLossRate10 = 0.10;
-  const double kPacketLossRate5 = 0.05;
-  const double kPacketLossRate1 = 0.01;
-  const double kLossRate20Margin = 0.02;
-  const double kLossRate10Margin = 0.01;
-  const double kLossRate5Margin = 0.01;
+float OptimizePacketLossRate(float new_loss_rate, float old_loss_rate) {
+  RTC_DCHECK_GE(new_loss_rate, 0.0f);
+  RTC_DCHECK_LE(new_loss_rate, 1.0f);
+  RTC_DCHECK_GE(old_loss_rate, 0.0f);
+  RTC_DCHECK_LE(old_loss_rate, 1.0f);
+  constexpr float kPacketLossRate20 = 0.20f;
+  constexpr float kPacketLossRate10 = 0.10f;
+  constexpr float kPacketLossRate5 = 0.05f;
+  constexpr float kPacketLossRate1 = 0.01f;
+  constexpr float kLossRate20Margin = 0.02f;
+  constexpr float kLossRate10Margin = 0.01f;
+  constexpr float kLossRate5Margin = 0.01f;
   if (new_loss_rate >=
       kPacketLossRate20 +
           kLossRate20Margin *
               (kPacketLossRate20 - old_loss_rate > 0 ? 1 : -1)) {
     return kPacketLossRate20;
   } else if (new_loss_rate >=
              kPacketLossRate10 +
                  kLossRate10Margin *
                      (kPacketLossRate10 - old_loss_rate > 0 ? 1 : -1)) {
     return kPacketLossRate10;
   } else if (new_loss_rate >=
              kPacketLossRate5 +
                  kLossRate5Margin *
                      (kPacketLossRate5 - old_loss_rate > 0 ? 1 : -1)) {
     return kPacketLossRate5;
   } else if (new_loss_rate >= kPacketLossRate1) {
     return kPacketLossRate1;
   } else {
-    return 0.0;
+    return 0.0f;
   }
 }
 
 }  // namespace
 
 class AudioEncoderOpus::PacketLossFractionSmoother {
  public:
   explicit PacketLossFractionSmoother(const Clock* clock)
       : clock_(clock),
         last_sample_time_ms_(clock_->TimeInMilliseconds()),
@@ skipping 131 matching lines @@
   }
   return RecreateEncoderInstance(conf);
 }
 
 void AudioEncoderOpus::SetMaxPlaybackRate(int frequency_hz) {
   auto conf = config_;
   conf.max_playback_rate_hz = 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) {
   audio_network_adaptor_ = audio_network_adaptor_creator_(config_string, clock);
   return audio_network_adaptor_.get() != nullptr;
 }
 
 void AudioEncoderOpus::DisableAudioNetworkAdaptor() {
   audio_network_adaptor_.reset(nullptr);
 }
@@ skipping 158 matching lines @@
   RTC_DCHECK_GT(num_channels_to_encode, 0u);
   RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
 
   if (num_channels_to_encode_ == num_channels_to_encode)
     return;
 
   RTC_CHECK_EQ(0, WebRtcOpus_SetForceChannels(inst_, num_channels_to_encode));
   num_channels_to_encode_ = num_channels_to_encode;
 }
 
+void AudioEncoderOpus::SetProjectedPacketLossRate(float fraction) {
+  float 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()));
+}
+
 void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
   auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
   // |audio_network_adaptor_| is supposed to be configured to output all
   // following parameters.
   RTC_DCHECK(config.bitrate_bps);
   RTC_DCHECK(config.frame_length_ms);
   RTC_DCHECK(config.uplink_packet_loss_fraction);
   RTC_DCHECK(config.enable_fec);
   RTC_DCHECK(config.enable_dtx);
   RTC_DCHECK(config.num_channels);
@@ skipping 15 matching lines @@
   AudioNetworkAdaptorImpl::Config config;
   config.clock = clock;
   return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
       config, ControllerManagerImpl::Create(
                   config_string, NumChannels(), supported_frame_lengths_ms(),
                   num_channels_to_encode_, next_frame_length_ms_,
                   GetTargetBitrate(), config_.fec_enabled, GetDtx(), clock)));
 }
 
 }  // namespace webrtc