diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
index bd4a2c0292..8892ee5a0b 100644
--- a/api/media_stream_interface.h
+++ b/api/media_stream_interface.h
@@ -216,6 +216,11 @@ class AudioTrackSinkInterface {
                   number_of_frames);
   }
 
+  // Returns the number of channels encoded by the sink. This can be less than
+  // the number_of_channels if down-mixing occur. A value of -1 means an unknown
+  // number.
+  virtual int NumPreferredChannels() const { return -1; }
+
  protected:
   virtual ~AudioTrackSinkInterface() {}
 };
diff --git a/media/base/audio_source.h b/media/base/audio_source.h
index 8a8796800b..51fe0e13e1 100644
--- a/media/base/audio_source.h
+++ b/media/base/audio_source.h
@@ -36,6 +36,11 @@ class AudioSource {
     // Called when the AudioSource is going away.
     virtual void OnClose() = 0;
 
+    // Returns the number of channels encoded by the sink. This can be less than
+    // the number_of_channels if down-mixing occur. A value of -1 means an
+    // unknown number.
+    virtual int NumPreferredChannels() const = 0;
+
    protected:
     virtual ~Sink() {}
   };
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 1751dd8bfe..42940bf1b4 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -371,6 +371,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
                 size_t number_of_frames,
                 absl::optional<int64_t> absolute_capture_timestamp_ms) override;
     void OnClose() override;
+    int NumPreferredChannels() const override { return -1; }
     AudioSource* source() const;
 
    private:
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index bc382d03e3..2ed78b429b 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1114,6 +1114,14 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
         *audio_codec_spec_);
 
     UpdateAllowedBitrateRange();
+
+    // Encoder will only use two channels if the stereo parameter is set.
+    const auto& it = send_codec_spec.format.parameters.find("stereo");
+    if (it != send_codec_spec.format.parameters.end() && it->second == "1") {
+      num_encoded_channels_ = 2;
+    } else {
+      num_encoded_channels_ = 1;
+    }
   }
 
   void UpdateAudioNetworkAdaptorConfig() {
@@ -1133,6 +1141,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     stream_->Reconfigure(config_);
   }
 
+  int NumPreferredChannels() const override { return num_encoded_channels_; }
+
   const AdaptivePtimeConfig adaptive_ptime_config_;
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
@@ -1154,6 +1164,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // TODO(webrtc:11717): Remove this once audio_network_adaptor in AudioOptions
   // has been removed.
   absl::optional<std::string> audio_network_adaptor_config_from_options_;
+  int num_encoded_channels_ = -1;
 };
 
 class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 0da6dfca80..5a7e237c90 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -405,6 +405,7 @@ void LocalAudioSinkAdapter::OnData(
   if (sink_) {
     sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
                   number_of_frames, absolute_capture_timestamp_ms);
+    num_preferred_channels_ = sink_->NumPreferredChannels();
   }
 }
 
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index c2fe91f01d..51ae1e978b 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -237,12 +237,16 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
            /*absolute_capture_timestamp_ms=*/absl::nullopt);
   }
 
+  // AudioSinkInterface implementation.
+  int NumPreferredChannels() const override { return num_preferred_channels_; }
+
   // cricket::AudioSource implementation.
   void SetSink(cricket::AudioSource::Sink* sink) override;
 
   cricket::AudioSource::Sink* sink_;
   // Critical section protecting |sink_|.
   Mutex lock_;
+  int num_preferred_channels_ = -1;
 };
 
 class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase {