Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )

Reason for revert:
internal bot failure

Original issue's description:
> Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/84e56d576806635c966093d5421c5d04c9b90746
> Cr-Commit-Position: refs/heads/master@{#15310}

TBR=kwiberg@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2537243004
Cr-Commit-Position: refs/heads/master@{#15312}

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

@@ -407,6 +407,12 @@ class RawAudioEncoderWrapper final : public AudioEncoder {
   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_;
@@ -648,7 +654,7 @@ int AudioCodingModuleImpl::SendFrequency() const {
 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
   rtc::CritScope lock(&acm_crit_sect_);
   if (encoder_stack_) {
-    encoder_stack_->OnReceivedTargetAudioBitrate(bitrate_bps);
+    encoder_stack_->SetTargetBitrate(bitrate_bps);
   }
 }
 
@@ -900,7 +906,7 @@ int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
   rtc::CritScope lock(&acm_crit_sect_);
   if (HaveValidEncoder("SetPacketLossRate")) {
-    encoder_stack_->OnReceivedUplinkPacketLossFraction(loss_rate / 100.0);
+    encoder_stack_->SetProjectedPacketLossRate(loss_rate / 100.0);
   }
   return 0;
 }

webrtc/modules/audio_coding/codecs/audio_encoder.cc

@@ -60,6 +60,10 @@ bool AudioEncoder::SetApplication(Application application) {
 
 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
 
+void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
 rtc::ArrayView<std::unique_ptr<AudioEncoder>>
 AudioEncoder::ReclaimContainedEncoders() { return nullptr; }
 
@@ -73,9 +77,13 @@ void AudioEncoder::DisableAudioNetworkAdaptor() {}
 void AudioEncoder::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {}
 
 void AudioEncoder::OnReceivedUplinkPacketLossFraction(
-    float uplink_packet_loss_fraction) {}
+    float uplink_packet_loss_fraction) {
+  SetProjectedPacketLossRate(uplink_packet_loss_fraction);
+}
 
-void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {}
+void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
+  SetTargetBitrate(target_audio_bitrate_bps);
+}
 
 void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
 

webrtc/modules/audio_coding/codecs/audio_encoder.h

@@ -144,6 +144,17 @@ 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);
+
+  // 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);
+
   // 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
@@ -164,7 +175,6 @@ 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);
 

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc

@@ -179,22 +179,19 @@ void AudioEncoderCng::SetMaxPlaybackRate(int frequency_hz) {
   speech_encoder_->SetMaxPlaybackRate(frequency_hz);
 }
 
+void AudioEncoderCng::SetProjectedPacketLossRate(double fraction) {
+  speech_encoder_->SetProjectedPacketLossRate(fraction);
+}
+
+void AudioEncoderCng::SetTargetBitrate(int bits_per_second) {
+  speech_encoder_->SetTargetBitrate(bits_per_second);
+}
+
 rtc::ArrayView<std::unique_ptr<AudioEncoder>>
 AudioEncoderCng::ReclaimContainedEncoders() {
   return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
 }
 
-void AudioEncoderCng::OnReceivedUplinkPacketLossFraction(
-    float uplink_packet_loss_fraction) {
-  speech_encoder_->OnReceivedUplinkPacketLossFraction(
-      uplink_packet_loss_fraction);
-}
-
-void AudioEncoderCng::OnReceivedTargetAudioBitrate(
-    int target_audio_bitrate_bps) {
-  speech_encoder_->OnReceivedTargetAudioBitrate(target_audio_bitrate_bps);
-}
-
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
     size_t frames_to_encode,
     rtc::Buffer* encoded) {

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h

@@ -61,11 +61,10 @@ class AudioEncoderCng final : public AudioEncoder {
   bool SetDtx(bool enable) override;
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
   rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
       override;
-  void OnReceivedUplinkPacketLossFraction(
-      float uplink_packet_loss_fraction) override;
-  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
 
  private:
   EncodedInfo EncodePassive(size_t frames_to_encode,

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc

@@ -217,16 +217,16 @@ TEST_F(AudioEncoderCngTest, CheckFrameSizePropagation) {
   EXPECT_EQ(17U, cng_->Num10MsFramesInNextPacket());
 }
 
-TEST_F(AudioEncoderCngTest, CheckTargetAudioBitratePropagation) {
+TEST_F(AudioEncoderCngTest, CheckChangeBitratePropagation) {
   CreateCng(MakeCngConfig());
-  EXPECT_CALL(*mock_encoder_, OnReceivedTargetAudioBitrate(4711));
+  EXPECT_CALL(*mock_encoder_, SetTargetBitrate(4711));
-  cng_->OnReceivedTargetAudioBitrate(4711);
+  cng_->SetTargetBitrate(4711);
 }
 
-TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
+TEST_F(AudioEncoderCngTest, CheckProjectedPacketLossRatePropagation) {
   CreateCng(MakeCngConfig());
-  EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
+  EXPECT_CALL(*mock_encoder_, SetProjectedPacketLossRate(0.5));
-  cng_->OnReceivedUplinkPacketLossFraction(0.5);
+  cng_->SetProjectedPacketLossRate(0.5);
 }
 
 TEST_F(AudioEncoderCngTest, EncodeCallsVad) {

webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h

@@ -39,12 +39,10 @@ class MockAudioEncoder : public AudioEncoder {
   MOCK_METHOD1(SetDtx, bool(bool enable));
   MOCK_METHOD1(SetApplication, bool(Application application));
   MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz));
+  MOCK_METHOD1(SetProjectedPacketLossRate, void(double fraction));
+  MOCK_METHOD1(SetTargetBitrate, void(int target_bps));
   MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
   MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
-  MOCK_METHOD1(OnReceivedTargetAudioBitrate,
-               void(int target_audio_bitrate_bps));
-  MOCK_METHOD1(OnReceivedUplinkPacketLossFraction,
-               void(float uplink_packet_loss_fraction));
 
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD3(EncodeImpl,

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

@@ -56,18 +56,18 @@ AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
 // 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.
-float OptimizePacketLossRate(float new_loss_rate, float old_loss_rate) {
+double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) {
-  RTC_DCHECK_GE(new_loss_rate, 0.0f);
+  RTC_DCHECK_GE(new_loss_rate, 0.0);
-  RTC_DCHECK_LE(new_loss_rate, 1.0f);
+  RTC_DCHECK_LE(new_loss_rate, 1.0);
-  RTC_DCHECK_GE(old_loss_rate, 0.0f);
+  RTC_DCHECK_GE(old_loss_rate, 0.0);
-  RTC_DCHECK_LE(old_loss_rate, 1.0f);
+  RTC_DCHECK_LE(old_loss_rate, 1.0);
-  constexpr float kPacketLossRate20 = 0.20f;
+  const double kPacketLossRate20 = 0.20;
-  constexpr float kPacketLossRate10 = 0.10f;
+  const double kPacketLossRate10 = 0.10;
-  constexpr float kPacketLossRate5 = 0.05f;
+  const double kPacketLossRate5 = 0.05;
-  constexpr float kPacketLossRate1 = 0.01f;
+  const double kPacketLossRate1 = 0.01;
-  constexpr float kLossRate20Margin = 0.02f;
+  const double kLossRate20Margin = 0.02;
-  constexpr float kLossRate10Margin = 0.01f;
+  const double kLossRate10Margin = 0.01;
-  constexpr float kLossRate5Margin = 0.01f;
+  const double kLossRate5Margin = 0.01;
   if (new_loss_rate >=
       kPacketLossRate20 +
           kLossRate20Margin *
@@ -86,7 +86,7 @@ float OptimizePacketLossRate(float new_loss_rate, float old_loss_rate) {
   } else if (new_loss_rate >= kPacketLossRate1) {
     return kPacketLossRate1;
   } else {
-    return 0.0f;
+    return 0.0;
   }
 }
 
@@ -259,6 +259,28 @@ 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()));
+  const auto new_complexity = config_.GetNewComplexity();
+  if (new_complexity && complexity_ != *new_complexity) {
+    complexity_ = *new_complexity;
+    RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
+  }
+}
+
 bool AudioEncoderOpus::EnableAudioNetworkAdaptor(
     const std::string& config_string,
     const Clock* clock) {
@@ -440,28 +462,6 @@ void AudioEncoderOpus::SetNumChannelsToEncode(size_t 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()));
-  const auto new_complexity = config_.GetNewComplexity();
-  if (new_complexity && complexity_ != *new_complexity) {
-    complexity_ = *new_complexity;
-    RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
-  }
-}
-
 void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
   auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
   // |audio_network_adaptor_| is supposed to be configured to output all

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

@@ -102,6 +102,9 @@ class AudioEncoderOpus final : public AudioEncoder {
 
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
+
   bool EnableAudioNetworkAdaptor(const std::string& config_string,
                                  const Clock* clock) override;
   void DisableAudioNetworkAdaptor() override;
@@ -117,7 +120,7 @@ class AudioEncoderOpus final : public AudioEncoder {
   }
 
   // Getters for testing.
-  float packet_loss_rate() const { return packet_loss_rate_; }
+  double packet_loss_rate() const { return packet_loss_rate_; }
   ApplicationMode application() const { return config_.application; }
   bool fec_enabled() const { return config_.fec_enabled; }
   size_t num_channels_to_encode() const { return num_channels_to_encode_; }
@@ -137,15 +140,13 @@ class AudioEncoderOpus final : public AudioEncoder {
   bool RecreateEncoderInstance(const Config& config);
   void SetFrameLength(int frame_length_ms);
   void SetNumChannelsToEncode(size_t num_channels_to_encode);
-  void SetProjectedPacketLossRate(float fraction);
-  void SetTargetBitrate(int target_bps);
   void ApplyAudioNetworkAdaptor();
   std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
       const std::string& config_string,
       const Clock* clock) const;
 
   Config config_;
-  float packet_loss_rate_;
+  double packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;

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

@@ -151,27 +151,26 @@ TEST(AudioEncoderOpusTest, ToggleDtx) {
   EXPECT_TRUE(states.encoder->SetDtx(false));
 }
 
-TEST(AudioEncoderOpusTest,
+TEST(AudioEncoderOpusTest, SetBitrate) {
-     OnReceivedTargetAudioBitrateWithoutAudioNetworkAdaptor) {
   auto states = CreateCodec(1);
   // Constants are replicated from audio_states.encoderopus.cc.
   const int kMinBitrateBps = 500;
   const int kMaxBitrateBps = 512000;
   // Set a too low bitrate.
-  states.encoder->OnReceivedTargetAudioBitrate(kMinBitrateBps - 1);
+  states.encoder->SetTargetBitrate(kMinBitrateBps - 1);
   EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
   // Set a too high bitrate.
-  states.encoder->OnReceivedTargetAudioBitrate(kMaxBitrateBps + 1);
+  states.encoder->SetTargetBitrate(kMaxBitrateBps + 1);
   EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
   // Set the minimum rate.
-  states.encoder->OnReceivedTargetAudioBitrate(kMinBitrateBps);
+  states.encoder->SetTargetBitrate(kMinBitrateBps);
   EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
   // Set the maximum rate.
-  states.encoder->OnReceivedTargetAudioBitrate(kMaxBitrateBps);
+  states.encoder->SetTargetBitrate(kMaxBitrateBps);
   EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
   // Set rates from 1000 up to 32000 bps.
   for (int rate = 1000; rate <= 32000; rate += 1000) {
-    states.encoder->OnReceivedTargetAudioBitrate(rate);
+    states.encoder->SetTargetBitrate(rate);
     EXPECT_EQ(rate, states.encoder->GetTargetBitrate());
   }
 }
@@ -180,31 +179,24 @@ namespace {
 
 // Returns a vector with the n evenly-spaced numbers a, a + (b - a)/(n - 1),
 // ..., b.
-std::vector<float> IntervalSteps(float a, float b, size_t n) {
+std::vector<double> IntervalSteps(double a, double b, size_t n) {
-  RTC_DCHECK_GT(n, 1u);
+  RTC_DCHECK_GT(n, 1);
-  const float step = (b - a) / (n - 1);
+  const double step = (b - a) / (n - 1);
-  std::vector<float> points;
+  std::vector<double> points;
-  points.push_back(a);
+  for (size_t i = 0; i < n; ++i)
-  for (size_t i = 1; i < n - 1; ++i)
     points.push_back(a + i * step);
-  points.push_back(b);
   return points;
 }
 
 // Sets the packet loss rate to each number in the vector in turn, and verifies
 // that the loss rate as reported by the encoder is |expected_return| for all
 // of them.
-void TestSetPacketLossRate(AudioEncoderOpusStates* states,
+void TestSetPacketLossRate(AudioEncoderOpus* encoder,
-                           const std::vector<float>& losses,
+                           const std::vector<double>& losses,
-                           float expected_return) {
+                           double expected_return) {
-  // |kSampleIntervalMs| is chosen to ease the calculation since
+  for (double loss : losses) {
-  // 0.9999 ^ 184198 = 1e-8. Which minimizes the effect of
+    encoder->SetProjectedPacketLossRate(loss);
-  // PacketLossFractionSmoother used in AudioEncoderOpus.
+    EXPECT_DOUBLE_EQ(expected_return, encoder->packet_loss_rate());
-  constexpr int64_t kSampleIntervalMs = 184198;
-  for (float loss : losses) {
-    states->encoder->OnReceivedUplinkPacketLossFraction(loss);
-    states->simulated_clock->AdvanceTimeMilliseconds(kSampleIntervalMs);
-    EXPECT_FLOAT_EQ(expected_return, states->encoder->packet_loss_rate());
   }
 }
 
@@ -212,23 +204,23 @@ void TestSetPacketLossRate(AudioEncoderOpusStates* states,
 
 TEST(AudioEncoderOpusTest, PacketLossRateOptimized) {
   auto states = CreateCodec(1);
-  auto I = [](float a, float b) { return IntervalSteps(a, b, 10); };
+  auto I = [](double a, double b) { return IntervalSteps(a, b, 10); };
-  constexpr float eps = 1e-8f;
+  const double eps = 1e-15;
 
   // Note that the order of the following calls is critical.
 
   // clang-format off
-  TestSetPacketLossRate(&states, I(0.00f      , 0.01f - eps), 0.00f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.00      , 0.01 - eps), 0.00);
-  TestSetPacketLossRate(&states, I(0.01f + eps, 0.06f - eps), 0.01f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.01 + eps, 0.06 - eps), 0.01);
-  TestSetPacketLossRate(&states, I(0.06f + eps, 0.11f - eps), 0.05f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.06 + eps, 0.11 - eps), 0.05);
-  TestSetPacketLossRate(&states, I(0.11f + eps, 0.22f - eps), 0.10f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.11 + eps, 0.22 - eps), 0.10);
-  TestSetPacketLossRate(&states, I(0.22f + eps, 1.00f      ), 0.20f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.22 + eps, 1.00      ), 0.20);
 
-  TestSetPacketLossRate(&states, I(1.00f      , 0.18f + eps), 0.20f);
+  TestSetPacketLossRate(states.encoder.get(), I(1.00      , 0.18 + eps), 0.20);
-  TestSetPacketLossRate(&states, I(0.18f - eps, 0.09f + eps), 0.10f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.18 - eps, 0.09 + eps), 0.10);
-  TestSetPacketLossRate(&states, I(0.09f - eps, 0.04f + eps), 0.05f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.09 - eps, 0.04 + eps), 0.05);
-  TestSetPacketLossRate(&states, I(0.04f - eps, 0.01f + eps), 0.01f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.04 - eps, 0.01 + eps), 0.01);
-  TestSetPacketLossRate(&states, I(0.01f - eps, 0.00f      ), 0.00f);
+  TestSetPacketLossRate(states.encoder.get(), I(0.01 - eps, 0.00      ), 0.00);
   // clang-format on
 }
 
@@ -325,13 +317,13 @@ TEST(AudioEncoderOpusTest,
   // will fail.
   constexpr float kPacketLossFraction_1 = 0.02f;
   constexpr float kPacketLossFraction_2 = 0.198f;
-  // |kSecondSampleTimeMs| is chosen to ease the calculation since
+  // |kSecondSampleTimeMs| is chose to ease the calculation since
   // 0.9999 ^ 6931 = 0.5.
-  constexpr int64_t kSecondSampleTimeMs = 6931;
+  constexpr float kSecondSampleTimeMs = 6931;
 
   // First time, no filtering.
   states.encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
-  EXPECT_FLOAT_EQ(0.01f, states.encoder->packet_loss_rate());
+  EXPECT_DOUBLE_EQ(0.01, states.encoder->packet_loss_rate());
 
   states.simulated_clock->AdvanceTimeMilliseconds(kSecondSampleTimeMs);
   states.encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
@@ -340,7 +332,7 @@ TEST(AudioEncoderOpusTest,
   // (0.02 + 0.198) / 2 = 0.109, which reach the threshold for the optimized
   // packet loss rate to increase to 0.05. If no smoothing has been made, the
   // optimized packet loss rate should have been increase to 0.1.
-  EXPECT_FLOAT_EQ(0.05f, states.encoder->packet_loss_rate());
+  EXPECT_DOUBLE_EQ(0.05, states.encoder->packet_loss_rate());
 }
 
 }  // namespace webrtc

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc

@@ -115,20 +115,17 @@ void AudioEncoderCopyRed::SetMaxPlaybackRate(int frequency_hz) {
   speech_encoder_->SetMaxPlaybackRate(frequency_hz);
 }
 
+void AudioEncoderCopyRed::SetProjectedPacketLossRate(double fraction) {
+  speech_encoder_->SetProjectedPacketLossRate(fraction);
+}
+
+void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) {
+  speech_encoder_->SetTargetBitrate(bits_per_second);
+}
+
 rtc::ArrayView<std::unique_ptr<AudioEncoder>>
 AudioEncoderCopyRed::ReclaimContainedEncoders() {
   return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
 }
 
-void AudioEncoderCopyRed::OnReceivedUplinkPacketLossFraction(
-    float uplink_packet_loss_fraction) {
-  speech_encoder_->OnReceivedUplinkPacketLossFraction(
-      uplink_packet_loss_fraction);
-}
-
-void AudioEncoderCopyRed::OnReceivedTargetAudioBitrate(
-    int target_audio_bitrate_bps) {
-  speech_encoder_->OnReceivedTargetAudioBitrate(target_audio_bitrate_bps);
-}
-
 }  // namespace webrtc

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h

@@ -49,11 +49,10 @@ class AudioEncoderCopyRed final : public AudioEncoder {
   bool SetDtx(bool enable) override;
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
   rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
       override;
-  void OnReceivedUplinkPacketLossFraction(
-      float uplink_packet_loss_fraction) override;
-  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc

@@ -97,14 +97,14 @@ TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
   EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
 }
 
-TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
+TEST_F(AudioEncoderCopyRedTest, CheckSetBitratePropagation) {
-  EXPECT_CALL(*mock_encoder_, OnReceivedTargetAudioBitrate(4711));
+  EXPECT_CALL(*mock_encoder_, SetTargetBitrate(4711));
-  red_->OnReceivedTargetAudioBitrate(4711);
+  red_->SetTargetBitrate(4711);
 }
 
-TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
+TEST_F(AudioEncoderCopyRedTest, CheckProjectedPacketLossRatePropagation) {
-  EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
+  EXPECT_CALL(*mock_encoder_, SetProjectedPacketLossRate(0.5));
-  red_->OnReceivedUplinkPacketLossFraction(0.5);
+  red_->SetProjectedPacketLossRate(0.5);
 }
 
 // Checks that the an Encode() call is immediately propagated to the speech

webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

@@ -469,7 +469,7 @@ TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
 
 namespace {
 int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
-  audio_encoder->OnReceivedTargetAudioBitrate(rate);
+  audio_encoder->SetTargetBitrate(rate);
   return audio_encoder->GetTargetBitrate();
 }
 void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,