diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index bbf6bdd0d9..054e090ba6 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -152,6 +152,7 @@ if (rtc_include_tests) {
       "../api/audio_codecs:audio_codecs_api",
       "../api/audio_codecs/opus:audio_decoder_opus",
       "../api/audio_codecs/opus:audio_encoder_opus",
+      "../api/crypto:frame_decryptor_interface",
       "../api/rtc_event_log",
       "../api/task_queue:default_task_queue_factory",
       "../api/units:time_delta",
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index cc53a746ff..7476e08914 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -246,6 +246,14 @@ void AudioReceiveStream::SetUseTransportCcAndNackHistory(bool use_transport_cc,
   }
 }
 
+void AudioReceiveStream::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  // TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream,
+  // expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetFrameDecryptor(std::move(frame_decryptor));
+}
+
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats(
     bool get_and_clear_legacy_stats) const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 4f63155377..108794cd92 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -91,6 +91,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
   void SetUseTransportCcAndNackHistory(bool use_transport_cc,
                                        int history_ms) override;
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
 
   webrtc::AudioReceiveStream::Stats GetStats(
       bool get_and_clear_legacy_stats) const override;
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 0582171b62..28568b17c4 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -177,6 +177,9 @@ class ChannelReceive : public ChannelReceiveInterface {
       rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
       override;
 
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+
  private:
   void ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
@@ -275,10 +278,12 @@ class ChannelReceive : public ChannelReceiveInterface {
   SequenceChecker construction_thread_;
 
   // E2EE Audio Frame Decryption
-  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
+  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
+      RTC_GUARDED_BY(worker_thread_checker_);
   webrtc::CryptoOptions crypto_options_;
 
-  webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_;
+  webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_
+      RTC_GUARDED_BY(worker_thread_checker_);
 
   webrtc::CaptureClockOffsetUpdater capture_clock_offset_updater_;
 
@@ -889,6 +894,13 @@ void ChannelReceive::SetDepacketizerToDecoderFrameTransformer(
   InitFrameTransformerDelegate(std::move(frame_transformer));
 }
 
+void ChannelReceive::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  // TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  frame_decryptor_ = std::move(frame_decryptor);
+}
+
 NetworkStatistics ChannelReceive::GetNetworkStatistics(
     bool get_and_clear_legacy_stats) const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index c55968b55f..0a51be6649 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -159,6 +159,9 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
   virtual void SetDepacketizerToDecoderFrameTransformer(
       rtc::scoped_refptr<webrtc::FrameTransformerInterface>
           frame_transformer) = 0;
+
+  virtual void SetFrameDecryptor(
+      rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
 };
 
 std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index 7f140d400d..deef5ae068 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -17,6 +17,7 @@
 #include <utility>
 #include <vector>
 
+#include "api/crypto/frame_decryptor_interface.h"
 #include "api/test/mock_frame_encryptor.h"
 #include "audio/channel_receive.h"
 #include "audio/channel_send.h"
@@ -98,6 +99,11 @@ class MockChannelReceive : public voe::ChannelReceiveInterface {
       SetDepacketizerToDecoderFrameTransformer,
       (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
       (override));
+  MOCK_METHOD(
+      void,
+      SetFrameDecryptor,
+      (rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
+      (override));
 };
 
 class MockChannelSend : public voe::ChannelSendInterface {
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 45c318c404..2f67f7cc14 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -177,6 +177,8 @@ class AudioReceiveStream {
   virtual void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) = 0;
   virtual void SetUseTransportCcAndNackHistory(bool use_transport_cc,
                                                int history_ms) = 0;
+  virtual void SetFrameDecryptor(
+      rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index 5f484285a5..0190d88e0e 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -113,6 +113,11 @@ void FakeAudioReceiveStream::SetUseTransportCcAndNackHistory(
   config_.rtp.nack.rtp_history_ms = history_ms;
 }
 
+void FakeAudioReceiveStream::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  config_.frame_decryptor = std::move(frame_decryptor);
+}
+
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats(
     bool get_and_clear_legacy_stats) const {
   return stats_;
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 79f155cd86..8df85d7564 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -112,6 +112,8 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
       std::map<int, webrtc::SdpAudioFormat> decoder_map) override;
   void SetUseTransportCcAndNackHistory(bool use_transport_cc,
                                        int history_ms) override;
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
 
   webrtc::AudioReceiveStream::Stats GetStats(
       bool get_and_clear_legacy_stats) const override;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 602d23cf68..575e2325f3 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1218,7 +1218,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
       rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
     RTC_DCHECK_RUN_ON(&worker_thread_checker_);
     config_.frame_decryptor = frame_decryptor;
-    RecreateAudioReceiveStream();
+    stream_->SetFrameDecryptor(std::move(frame_decryptor));
   }
 
   void SetLocalSsrc(uint32_t local_ssrc) {