diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index bfbb7ea571..425c213c39 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -302,7 +302,7 @@ ControllerManagerImpl::ControllerManagerImpl(const Config& config)
 
 ControllerManagerImpl::ControllerManagerImpl(
     const Config& config,
-    std::vector<std::unique_ptr<Controller>>&& controllers,
+    std::vector<std::unique_ptr<Controller>> controllers,
     const std::map<const Controller*, std::pair<int, float>>& scoring_points)
     : config_(config),
       controllers_(std::move(controllers)),
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
index 78e962c3db..600fe65c10 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -74,7 +74,7 @@ class ControllerManagerImpl final : public ControllerManager {
   // Dependency injection for testing.
   ControllerManagerImpl(
       const Config& config,
-      std::vector<std::unique_ptr<Controller>>&& controllers,
+      std::vector<std::unique_ptr<Controller>> controllers,
       const std::map<const Controller*, std::pair<int, float>>&
           chracteristic_points);
 
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 e94ea6572e..5756e2af68 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -368,31 +368,31 @@ class AudioEncoderOpus::PacketLossFractionSmoother {
 AudioEncoderOpus::AudioEncoderOpus(const AudioEncoderOpusConfig& config)
     : AudioEncoderOpus(config, config.payload_type) {}
 
+AudioEncoderOpus::AudioEncoderOpus(const AudioEncoderOpusConfig& config,
+                                   int payload_type)
+    : AudioEncoderOpus(
+          config,
+          payload_type,
+          [this](const ProtoString& config_string, RtcEventLog* event_log) {
+            return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
+          },
+          // We choose 5sec as initial time constant due to empirical data.
+          rtc::MakeUnique<SmoothingFilterImpl>(5000)) {}
+
 AudioEncoderOpus::AudioEncoderOpus(
     const AudioEncoderOpusConfig& config,
     int payload_type,
-    AudioNetworkAdaptorCreator&& audio_network_adaptor_creator,
+    const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
     std::unique_ptr<SmoothingFilter> bitrate_smoother)
     : payload_type_(payload_type),
-      send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
-          "WebRTC-SendSideBwe-WithOverhead")),
+      send_side_bwe_with_overhead_(
+          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
       packet_loss_rate_(0.0),
       inst_(nullptr),
       packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
-      audio_network_adaptor_creator_(
-          audio_network_adaptor_creator
-              ? std::move(audio_network_adaptor_creator)
-              : [this](const ProtoString& config_string,
-                       RtcEventLog* event_log) {
-                  return DefaultAudioNetworkAdaptorCreator(config_string,
-                                                           event_log);
-              }),
-      bitrate_smoother_(bitrate_smoother
-          ? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
-              // We choose 5sec as initial time constant due to empirical data.
-              new SmoothingFilterImpl(5000))) {
-  RTC_DCHECK_GE(payload_type, 0);
-  RTC_DCHECK_LE(payload_type, 127);
+      audio_network_adaptor_creator_(audio_network_adaptor_creator),
+      bitrate_smoother_(std::move(bitrate_smoother)) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
 
   // Sanity check of the redundant payload type field that we want to get rid
   // of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 473a65d29d..2382700fab 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -67,11 +67,14 @@ class AudioEncoderOpus final : public AudioEncoder {
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
   AudioEncoderOpus(const AudioEncoderOpusConfig& config);
 
+  AudioEncoderOpus(const AudioEncoderOpusConfig& config, int payload_type);
+
+  // Dependency injection for testing.
   AudioEncoderOpus(
       const AudioEncoderOpusConfig& config,
       int payload_type,
-      AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
-      std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
+      const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+      std::unique_ptr<SmoothingFilter> bitrate_smoother);
 
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
   AudioEncoderOpus(int payload_type, const SdpAudioFormat& format);
@@ -164,7 +167,7 @@ class AudioEncoderOpus final : public AudioEncoder {
   int next_frame_length_ms_;
   int complexity_;
   std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
-  AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
+  const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
   std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
   rtc::Optional<size_t> overhead_bytes_per_packet_;
   const std::unique_ptr<SmoothingFilter> bitrate_smoother_;