diff --git a/api/mediastreaminterface.cc b/api/mediastreaminterface.cc
index e36d5cb83b..955e7e4c60 100644
--- a/api/mediastreaminterface.cc
+++ b/api/mediastreaminterface.cc
@@ -30,4 +30,8 @@ AudioTrackInterface::GetAudioProcessor() {
   return nullptr;
 }
 
+const cricket::AudioOptions AudioSourceInterface::options() const {
+  return {};
+}
+
 }  // namespace webrtc
diff --git a/api/mediastreaminterface.h b/api/mediastreaminterface.h
index 30f8f71602..6d967660c5 100644
--- a/api/mediastreaminterface.h
+++ b/api/mediastreaminterface.h
@@ -23,6 +23,7 @@
 #include <vector>
 
 #include "absl/types/optional.h"
+#include "api/audio_options.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
@@ -207,6 +208,11 @@ class AudioSourceInterface : public MediaSourceInterface {
   // TODO(tommi): Make pure virtual.
   virtual void AddSink(AudioTrackSinkInterface* sink) {}
   virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
+
+  // Returns options for the AudioSource.
+  // (for some of the settings this approach is broken, e.g. setting
+  // audio network adaptation on the source is the wrong layer of abstraction).
+  virtual const cricket::AudioOptions options() const;
 };
 
 // Interface of the audio processor used by the audio track to collect
diff --git a/pc/localaudiosource.h b/pc/localaudiosource.h
index c5f65304d0..c48f5407d4 100644
--- a/pc/localaudiosource.h
+++ b/pc/localaudiosource.h
@@ -29,7 +29,7 @@ class LocalAudioSource : public Notifier<AudioSourceInterface> {
   SourceState state() const override { return kLive; }
   bool remote() const override { return false; }
 
-  virtual const cricket::AudioOptions& options() const { return options_; }
+  const cricket::AudioOptions options() const override { return options_; }
 
   void AddSink(AudioTrackSinkInterface* sink) override {}
   void RemoveSink(AudioTrackSinkInterface* sink) override {}
diff --git a/pc/rtpsender.cc b/pc/rtpsender.cc
index 3196641166..abd1748cea 100644
--- a/pc/rtpsender.cc
+++ b/pc/rtpsender.cc
@@ -405,9 +405,7 @@ void AudioRtpSender::SetAudioSend() {
   // options since it is also applied to all streams/channels, local or remote.
   if (track_->enabled() && track_->GetSource() &&
       !track_->GetSource()->remote()) {
-    // TODO(xians): Remove this static_cast since we should be able to connect
-    // a remote audio track to a peer connection.
-    options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
+    options = track_->GetSource()->options();
   }
 #endif