diff --git a/api/peer_connection_factory_proxy.h b/api/peer_connection_factory_proxy.h
index 620f1ca172..b61a78291c 100644
--- a/api/peer_connection_factory_proxy.h
+++ b/api/peer_connection_factory_proxy.h
@@ -56,6 +56,7 @@ PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
               CreateAudioTrack,
               const std::string&,
               AudioSourceInterface*)
+PROXY_METHOD2(bool, StartAecDump, FILE*, int64_t)
 PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
 PROXY_METHOD0(void, StopAecDump)
 END_PROXY_MAP()
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index ceec13ab0c..0b2bfa473e 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -67,6 +67,8 @@
 #ifndef API_PEER_CONNECTION_INTERFACE_H_
 #define API_PEER_CONNECTION_INTERFACE_H_
 
+#include <stdio.h>
+
 #include <memory>
 #include <string>
 #include <vector>
@@ -1405,6 +1407,12 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
   // reached, logging is stopped automatically. If max_size_bytes is set to a
   // value <= 0, no limit will be used, and logging will continue until the
   // StopAecDump function is called.
+  // TODO(webrtc:6463): Delete default implementation when downstream mocks
+  // classes are updated.
+  virtual bool StartAecDump(FILE* file, int64_t max_size_bytes) {
+    return false;
+  }
+  // TODO(webrtc:6463): Deprecated; PlatformFile will soon be deleted.
   virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
 
   // Stops logging the AEC dump.
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 6627eea13a..6dee916f3d 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -95,6 +95,7 @@ rtc_static_library("rtc_media_base") {
     "../rtc_base:sanitizer",
     "../rtc_base:stringutils",
     "../rtc_base/synchronization:sequence_checker",
+    "../rtc_base/system:file_wrapper",
     "../rtc_base/system:rtc_export",
     "../rtc_base/third_party/sigslot",
     "//third_party/abseil-cpp/absl/algorithm:container",
diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc
index 358b51c7de..ab398ce4a9 100644
--- a/media/base/fake_media_engine.cc
+++ b/media/base/fake_media_engine.cc
@@ -542,7 +542,7 @@ void FakeVoiceEngine::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
 int FakeVoiceEngine::GetInputLevel() {
   return 0;
 }
-bool FakeVoiceEngine::StartAecDump(rtc::PlatformFile file,
+bool FakeVoiceEngine::StartAecDump(webrtc::FileWrapper file,
                                    int64_t max_size_bytes) {
   return false;
 }
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index b42f896b80..f3c77411d7 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -526,7 +526,7 @@ class FakeVoiceEngine : public VoiceEngineInterface {
   void SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   void SetSendCodecs(const std::vector<AudioCodec>& codecs);
   int GetInputLevel();
-  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
+  bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
   void StopAecDump() override;
   bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
   void StopRtcEventLog();
diff --git a/media/base/media_engine.h b/media/base/media_engine.h
index 9bc49f95f1..bf4fd3cc25 100644
--- a/media/base/media_engine.h
+++ b/media/base/media_engine.h
@@ -28,7 +28,7 @@
 #include "media/base/codec.h"
 #include "media/base/media_channel.h"
 #include "media/base/video_common.h"
-#include "rtc_base/platform_file.h"
+#include "rtc_base/system/file_wrapper.h"
 
 namespace webrtc {
 class AudioDeviceModule;
@@ -80,7 +80,8 @@ class VoiceEngineInterface {
   // Starts AEC dump using existing file, a maximum file size in bytes can be
   // specified. Logging is stopped just before the size limit is exceeded.
   // If max_size_bytes is set to a value <= 0, no limit will be used.
-  virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
+  virtual bool StartAecDump(webrtc::FileWrapper file,
+                            int64_t max_size_bytes) = 0;
 
   // Stops recording AEC dump.
   virtual void StopAecDump() = 0;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 0635c9f3db..46e219f1e9 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -574,11 +574,11 @@ void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
   channels_.erase(it);
 }
 
-bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
+bool WebRtcVoiceEngine::StartAecDump(webrtc::FileWrapper file,
                                      int64_t max_size_bytes) {
   RTC_DCHECK(worker_thread_checker_.IsCurrent());
   auto aec_dump = webrtc::AecDumpFactory::Create(
-      file, max_size_bytes, low_priority_worker_queue_.get());
+      std::move(file), max_size_bytes, low_priority_worker_queue_.get());
   if (!aec_dump) {
     return false;
   }
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index aaa2778a21..a4c8baa37f 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -78,7 +78,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   // specified. When the maximum file size is reached, logging is stopped and
   // the file is closed. If max_size_bytes is set to <= 0, no limit will be
   // used.
-  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
+  bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
 
   // Stops AEC dump.
   void StopAecDump() override;
diff --git a/modules/audio_processing/aec_dump/aec_dump_factory.h b/modules/audio_processing/aec_dump/aec_dump_factory.h
index 2a800a9418..429a8a5881 100644
--- a/modules/audio_processing/aec_dump/aec_dump_factory.h
+++ b/modules/audio_processing/aec_dump/aec_dump_factory.h
@@ -15,7 +15,6 @@
 #include <string>
 
 #include "modules/audio_processing/include/aec_dump.h"
-#include "rtc_base/platform_file.h"
 #include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -33,11 +32,7 @@ class RTC_EXPORT AecDumpFactory {
   // responsibility for |handle| and closes it in the destructor. A
   // non-null return value indicates that the file has been
   // sucessfully opened.
-  static std::unique_ptr<AecDump> Create(webrtc::FileWrapper&& file,
-                                         int64_t max_log_size_bytes,
-                                         rtc::TaskQueue* worker_queue);
-  // TODO(webrtc:6463): Deprecated. Delete, together with the PlatformFile type.
-  static std::unique_ptr<AecDump> Create(rtc::PlatformFile file,
+  static std::unique_ptr<AecDump> Create(webrtc::FileWrapper file,
                                          int64_t max_log_size_bytes,
                                          rtc::TaskQueue* worker_queue);
   static std::unique_ptr<AecDump> Create(std::string file_name,
diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.cc b/modules/audio_processing/aec_dump/aec_dump_impl.cc
index ebd0f575c3..ba15336c08 100644
--- a/modules/audio_processing/aec_dump/aec_dump_impl.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_impl.cc
@@ -216,7 +216,7 @@ std::unique_ptr<WriteToFileTask> AecDumpImpl::CreateWriteToFileTask() {
                                             &num_bytes_left_for_log_);
 }
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper&& file,
+std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
                                                 int64_t max_log_size_bytes,
                                                 rtc::TaskQueue* worker_queue) {
   RTC_DCHECK(worker_queue);
@@ -227,13 +227,6 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper&& file,
                                         worker_queue);
 }
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(rtc::PlatformFile file,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
-  return Create(FileWrapper(rtc::FdopenPlatformFileForWriting(file)),
-                max_log_size_bytes, worker_queue);
-}
-
 std::unique_ptr<AecDump> AecDumpFactory::Create(std::string file_name,
                                                 int64_t max_log_size_bytes,
                                                 rtc::TaskQueue* worker_queue) {
@@ -246,4 +239,5 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
                                                 rtc::TaskQueue* worker_queue) {
   return Create(FileWrapper(handle), max_log_size_bytes, worker_queue);
 }
+
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec_dump/null_aec_dump_factory.cc b/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
index 14e7c037cf..126adeb4eb 100644
--- a/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
+++ b/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
@@ -13,13 +13,7 @@
 
 namespace webrtc {
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper&& file,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
-  return nullptr;
-}
-
-std::unique_ptr<AecDump> AecDumpFactory::Create(rtc::PlatformFile file,
+std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
                                                 int64_t max_log_size_bytes,
                                                 rtc::TaskQueue* worker_queue) {
   return nullptr;
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 1c1d741b3f..2c23cb3b37 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -1581,14 +1581,14 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) {
 
   const std::string filename =
       test::TempFilename(test::OutputPath(), "debug_aec");
-  FILE* fid = fopen(filename.c_str(), "w");
-  ASSERT_TRUE(fid);
+  FileWrapper f = FileWrapper::OpenWriteOnly(filename.c_str());
+  ASSERT_TRUE(f.is_open());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stopping without having started should be OK.
   apm_->DetachAecDump();
 
-  auto aec_dump = AecDumpFactory::Create(fid, -1, &worker_queue);
+  auto aec_dump = AecDumpFactory::Create(std::move(f), -1, &worker_queue);
   EXPECT_TRUE(aec_dump);
   apm_->AttachAecDump(std::move(aec_dump));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(revframe_));
@@ -1596,14 +1596,12 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) {
   apm_->DetachAecDump();
 
   // Verify the file has been written.
-  fid = fopen(filename.c_str(), "r");
+  FILE* fid = fopen(filename.c_str(), "r");
   ASSERT_TRUE(fid != NULL);
 
   // Clean it up.
   ASSERT_EQ(0, fclose(fid));
   ASSERT_EQ(0, remove(filename.c_str()));
-#else
-  ASSERT_EQ(0, fclose(fid));
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 65f968af47..b4573d9a39 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -100,6 +100,7 @@ rtc_static_library("rtc_pc_base") {
     "../rtc_base:deprecation",
     "../rtc_base:rtc_task_queue",
     "../rtc_base:stringutils",
+    "../rtc_base/system:file_wrapper",
     "../rtc_base/third_party/base64",
     "../rtc_base/third_party/sigslot",
     "../system_wrappers:metrics",
@@ -229,6 +230,7 @@ rtc_static_library("peerconnection") {
     "../rtc_base:checks",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:safe_minmax",
+    "../rtc_base/system:file_wrapper",
     "../rtc_base/system:rtc_export",
     "../rtc_base/third_party/base64",
     "../rtc_base/third_party/sigslot",
diff --git a/pc/channel_manager.cc b/pc/channel_manager.cc
index 55f526800c..0023a2f6e4 100644
--- a/pc/channel_manager.cc
+++ b/pc/channel_manager.cc
@@ -358,10 +358,10 @@ void ChannelManager::DestroyRtpDataChannel(RtpDataChannel* data_channel) {
   data_channels_.erase(it);
 }
 
-bool ChannelManager::StartAecDump(rtc::PlatformFile file,
+bool ChannelManager::StartAecDump(webrtc::FileWrapper file,
                                   int64_t max_size_bytes) {
   return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
-    return media_engine_->voice().StartAecDump(file, max_size_bytes);
+    return media_engine_->voice().StartAecDump(std::move(file), max_size_bytes);
   });
 }
 
diff --git a/pc/channel_manager.h b/pc/channel_manager.h
index 34f9013d73..cae812f3c5 100644
--- a/pc/channel_manager.h
+++ b/pc/channel_manager.h
@@ -28,7 +28,7 @@
 #include "pc/channel.h"
 #include "pc/rtp_transport_internal.h"
 #include "pc/session_description.h"
-#include "rtc_base/platform_file.h"
+#include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/thread.h"
 
 namespace cricket {
@@ -153,7 +153,7 @@ class ChannelManager final {
   // Starts AEC dump using existing file, with a specified maximum file size in
   // bytes. When the limit is reached, logging will stop and the file will be
   // closed. If max_size_bytes is set to <= 0, no limit will be used.
-  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
+  bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes);
 
   // Stops recording AEC dump.
   void StopAecDump();
diff --git a/pc/peer_connection_factory.cc b/pc/peer_connection_factory.cc
index b2b12b0dbd..e3da4cfce6 100644
--- a/pc/peer_connection_factory.cc
+++ b/pc/peer_connection_factory.cc
@@ -37,6 +37,7 @@
 #include "pc/video_track.h"
 #include "rtc_base/bind.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/system/file_wrapper.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
@@ -208,10 +209,20 @@ PeerConnectionFactory::CreateAudioSource(const cricket::AudioOptions& options) {
   return source;
 }
 
+bool PeerConnectionFactory::StartAecDump(FILE* file, int64_t max_size_bytes) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  return channel_manager_->StartAecDump(FileWrapper(file), max_size_bytes);
+}
+
 bool PeerConnectionFactory::StartAecDump(rtc::PlatformFile file,
                                          int64_t max_size_bytes) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
-  return channel_manager_->StartAecDump(file, max_size_bytes);
+  FILE* f = rtc::FdopenPlatformFileForWriting(file);
+  if (!f) {
+    rtc::ClosePlatformFile(file);
+    return false;
+  }
+  return StartAecDump(f, max_size_bytes);
 }
 
 void PeerConnectionFactory::StopAecDump() {
diff --git a/pc/peer_connection_factory.h b/pc/peer_connection_factory.h
index 9b2ce67e39..6f68a26c77 100644
--- a/pc/peer_connection_factory.h
+++ b/pc/peer_connection_factory.h
@@ -69,6 +69,7 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
       const std::string& id,
       AudioSourceInterface* audio_source) override;
 
+  bool StartAecDump(FILE* file, int64_t max_size_bytes) override;
   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
   void StopAecDump() override;