diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index ff5b6d98a1..456f267daf 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1854,14 +1854,13 @@ rtc_library("remote_audio_source") {
     "../api:media_stream_interface",
     "../api:scoped_refptr",
     "../api:sequence_checker",
+    "../api/task_queue",
     "../media:rtc_media_base",
     "../rtc_base",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:safe_conversions",
     "../rtc_base:stringutils",
-    "../rtc_base:threading",
     "../rtc_base/synchronization:mutex",
   ]
   absl_deps = [
diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index 597d19977b..667312c38c 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -14,15 +14,15 @@
 
 #include <memory>
 #include <string>
+#include <utility>
 
 #include "absl/algorithm/container.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_format.h"
-#include "rtc_base/thread.h"
 
 namespace webrtc {
 
@@ -51,9 +51,9 @@ class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
 };
 
 RemoteAudioSource::RemoteAudioSource(
-    rtc::Thread* worker_thread,
+    TaskQueueBase* worker_thread,
     OnAudioChannelGoneAction on_audio_channel_gone_action)
-    : main_thread_(rtc::Thread::Current()),
+    : main_thread_(TaskQueueBase::Current()),
       worker_thread_(worker_thread),
       on_audio_channel_gone_action_(on_audio_channel_gone_action),
       state_(MediaSourceInterface::kInitializing) {
@@ -163,24 +163,18 @@ void RemoteAudioSource::OnAudioChannelGone() {
   if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) {
     return;
   }
-  // Called when the audio channel is deleted.  It may be the worker thread
-  // in libjingle or may be a different worker thread.
-  // This object needs to live long enough for the cleanup logic in OnMessage to
-  // run, so take a reference to it as the data. Sometimes the message may not
-  // be processed (because the thread was destroyed shortly after this call),
-  // but that is fine because the thread destructor will take care of destroying
-  // the message data which will release the reference on RemoteAudioSource.
-  main_thread_->Post(RTC_FROM_HERE, this, 0,
-                     new rtc::ScopedRefMessageData<RemoteAudioSource>(this));
-}
-
-void RemoteAudioSource::OnMessage(rtc::Message* msg) {
-  RTC_DCHECK_RUN_ON(main_thread_);
-  sinks_.clear();
-  SetState(MediaSourceInterface::kEnded);
-  // Will possibly delete this RemoteAudioSource since it is reference counted
-  // in the message.
-  delete msg->pdata;
+  // Called when the audio channel is deleted. It may be the worker thread or
+  // may be a different task queue.
+  // This object needs to live long enough for the cleanup logic in the posted
+  // task to run, so take a reference to it. Sometimes the task may not be
+  // processed (because the task queue was destroyed shortly after this call),
+  // but that is fine because the task queue destructor will take care of
+  // destroying task which will release the reference on RemoteAudioSource.
+  rtc::scoped_refptr<RemoteAudioSource> thiz(this);
+  main_thread_->PostTask([thiz = std::move(thiz)] {
+    thiz->sinks_.clear();
+    thiz->SetState(MediaSourceInterface::kEnded);
+  });
 }
 
 }  // namespace webrtc
diff --git a/pc/remote_audio_source.h b/pc/remote_audio_source.h
index 89af4db714..d294a0f0fb 100644
--- a/pc/remote_audio_source.h
+++ b/pc/remote_audio_source.h
@@ -20,24 +20,16 @@
 #include "api/call/audio_sink.h"
 #include "api/media_stream_interface.h"
 #include "api/notifier.h"
+#include "api/task_queue/task_queue_base.h"
 #include "media/base/media_channel.h"
-#include "rtc_base/message_handler.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/thread.h"
-#include "rtc_base/thread_message.h"
-
-namespace rtc {
-struct Message;
-class Thread;
-}  // namespace rtc
 
 namespace webrtc {
 
 // This class implements the audio source used by the remote audio track.
 // This class works by configuring itself as a sink with the underlying media
 // engine, then when receiving data will fan out to all added sinks.
-class RemoteAudioSource : public Notifier<AudioSourceInterface>,
-                          rtc::MessageHandler {
+class RemoteAudioSource : public Notifier<AudioSourceInterface> {
  public:
   // In Unified Plan, receivers map to m= sections and their tracks and sources
   // survive SSRCs being reconfigured. The life cycle of the remote audio source
@@ -52,7 +44,7 @@ class RemoteAudioSource : public Notifier<AudioSourceInterface>,
   };
 
   explicit RemoteAudioSource(
-      rtc::Thread* worker_thread,
+      TaskQueueBase* worker_thread,
       OnAudioChannelGoneAction on_audio_channel_gone_action);
 
   // Register and unregister remote audio source with the underlying media
@@ -85,10 +77,8 @@ class RemoteAudioSource : public Notifier<AudioSourceInterface>,
   void OnData(const AudioSinkInterface::Data& audio);
   void OnAudioChannelGone();
 
-  void OnMessage(rtc::Message* msg) override;
-
-  rtc::Thread* const main_thread_;
-  rtc::Thread* const worker_thread_;
+  TaskQueueBase* const main_thread_;
+  TaskQueueBase* const worker_thread_;
   const OnAudioChannelGoneAction on_audio_channel_gone_action_;
   std::list<AudioObserver*> audio_observers_;
   Mutex sink_lock_;