diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 47bfa7d..85c7280 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -587,14 +587,8 @@
     return false;
   }
 
-  auto aec_dump = webrtc::AecDumpFactory::Create(
-      std::move(file), max_size_bytes, low_priority_worker_queue_.get());
-  if (!aec_dump) {
-    return false;
-  }
-
-  ap->AttachAecDump(std::move(aec_dump));
-  return true;
+  return ap->CreateAndAttachAecDump(file.Release(), max_size_bytes,
+                                    low_priority_worker_queue_.get());
 }
 
 void WebRtcVoiceEngine::StopAecDump() {
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 86ecbff..7ca78e2 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -50,6 +50,7 @@
     "../../rtc_base:macromagic",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:arch",
+    "../../rtc_base/system:file_wrapper",
     "../../rtc_base/system:rtc_export",
     "agc:gain_control_interface",
     "//third_party/abseil-cpp/absl/types:optional",
@@ -112,6 +113,20 @@
   ]
 }
 
+rtc_source_set("aec_dump_interface") {
+  visibility = [ "*" ]
+  sources = [
+    "include/aec_dump.cc",
+    "include/aec_dump.h",
+  ]
+
+  deps = [
+    ":api",
+    ":audio_frame_view",
+    "../../rtc_base:deprecation",
+  ]
+}
+
 rtc_library("audio_processing") {
   visibility = [ "*" ]
   configs += [ ":apm_debug_dump" ]
@@ -134,8 +149,6 @@
     "gain_control_impl.h",
     "gain_controller2.cc",
     "gain_controller2.h",
-    "include/aec_dump.cc",
-    "include/aec_dump.h",
     "level_estimator.cc",
     "level_estimator.h",
     "render_queue_item_verifier.h",
@@ -147,6 +160,7 @@
 
   defines = []
   deps = [
+    ":aec_dump_interface",
     ":api",
     ":apm_logging",
     ":audio_buffer",
@@ -178,6 +192,7 @@
     "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
     "aec3",
+    "aec_dump:aec_dump",
     "aecm:aecm_core",
     "agc",
     "agc:gain_control_interface",
@@ -198,6 +213,12 @@
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers",
   ]
+
+  if (rtc_enable_protobuf) {
+    deps += [ "aec_dump:aec_dump_impl" ]
+  } else {
+    deps += [ "aec_dump:null_aec_dump_factory" ]
+  }
 }
 
 rtc_library("voice_detection") {
@@ -283,6 +304,7 @@
     testonly = true
     sources = [ "include/mock_audio_processing.h" ]
     deps = [
+      ":aec_dump_interface",
       ":api",
       ":audio_buffer",
       ":audio_processing",
diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn
index 7ba3bc0..9887f7d 100644
--- a/modules/audio_processing/aec_dump/BUILD.gn
+++ b/modules/audio_processing/aec_dump/BUILD.gn
@@ -13,7 +13,7 @@
   sources = [ "aec_dump_factory.h" ]
 
   deps = [
-    "../",
+    "..:aec_dump_interface",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:file_wrapper",
     "../../../rtc_base/system:rtc_export",
@@ -29,6 +29,7 @@
     ]
 
     deps = [
+      "..:aec_dump_interface",
       "..:audioproc_test_utils",
       "../",
       "../../../test:test_support",
@@ -64,7 +65,7 @@
 
     deps = [
       ":aec_dump",
-      "../",
+      "..:aec_dump_interface",
       "../../../api/audio:audio_frame_api",
       "../../../api/task_queue",
       "../../../rtc_base:checks",
@@ -104,6 +105,6 @@
 
   deps = [
     ":aec_dump",
-    "../",
+    "..:aec_dump_interface",
   ]
 }
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 7751bac..6abebd2 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -22,6 +22,7 @@
 #include "api/audio/audio_frame.h"
 #include "common_audio/audio_converter.h"
 #include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "modules/audio_processing/agc2/gain_applier.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/common.h"
@@ -1532,6 +1533,32 @@
   }
 }
 
+bool AudioProcessingImpl::CreateAndAttachAecDump(const std::string& file_name,
+                                                 int64_t max_log_size_bytes,
+                                                 rtc::TaskQueue* worker_queue) {
+  std::unique_ptr<AecDump> aec_dump =
+      AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue);
+  if (!aec_dump) {
+    return false;
+  }
+
+  AttachAecDump(std::move(aec_dump));
+  return true;
+}
+
+bool AudioProcessingImpl::CreateAndAttachAecDump(FILE* handle,
+                                                 int64_t max_log_size_bytes,
+                                                 rtc::TaskQueue* worker_queue) {
+  std::unique_ptr<AecDump> aec_dump =
+      AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue);
+  if (!aec_dump) {
+    return false;
+  }
+
+  AttachAecDump(std::move(aec_dump));
+  return true;
+}
+
 void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
   RTC_DCHECK(aec_dump);
   rtc::CritScope cs_render(&crit_render_);
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 188777e..3aa86ac 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -11,8 +11,11 @@
 #ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 #define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
+#include <stdio.h>
+
 #include <list>
 #include <memory>
+#include <string>
 #include <vector>
 
 #include "api/function_view.h"
@@ -70,6 +73,13 @@
   int Initialize(const ProcessingConfig& processing_config) override;
   void ApplyConfig(const AudioProcessing::Config& config) override;
   void SetExtraOptions(const webrtc::Config& config) override;
+  bool CreateAndAttachAecDump(const std::string& file_name,
+                              int64_t max_log_size_bytes,
+                              rtc::TaskQueue* worker_queue) override;
+  bool CreateAndAttachAecDump(FILE* handle,
+                              int64_t max_log_size_bytes,
+                              rtc::TaskQueue* worker_queue) override;
+  // TODO(webrtc:5298) Deprecated variant.
   void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
   void DetachAecDump() override;
   void SetRuntimeSetting(RuntimeSetting setting) override;
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 25b08c6..d84318f 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -33,8 +33,13 @@
 #include "rtc_base/arraysize.h"
 #include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
+#include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/system/rtc_export.h"
 
+namespace rtc {
+class TaskQueue;
+}  // namespace rtc
+
 namespace webrtc {
 
 class AecDump;
@@ -600,6 +605,23 @@
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
 
+  // Creates and attaches an webrtc::AecDump for recording debugging
+  // information.
+  // The |worker_queue| may not be null and must outlive the created
+  // AecDump instance. |max_log_size_bytes == -1| means the log size
+  // will be unlimited. |handle| may not be null. The AecDump takes
+  // responsibility for |handle| and closes it in the destructor. A
+  // return value of true indicates that the file has been
+  // sucessfully opened, while a value of false indicates that
+  // opening the file failed.
+  virtual bool CreateAndAttachAecDump(const std::string& file_name,
+                                      int64_t max_log_size_bytes,
+                                      rtc::TaskQueue* worker_queue) = 0;
+  virtual bool CreateAndAttachAecDump(FILE* handle,
+                                      int64_t max_log_size_bytes,
+                                      rtc::TaskQueue* worker_queue) = 0;
+
+  // TODO(webrtc:5298) Deprecated variant.
   // Attaches provided webrtc::AecDump for recording debugging
   // information. Log file and maximum file size logic is supposed to
   // be handled by implementing instance of AecDump. Calling this
diff --git a/modules/audio_processing/include/mock_audio_processing.h b/modules/audio_processing/include/mock_audio_processing.h
index 9492a38..bdae99a 100644
--- a/modules/audio_processing/include/mock_audio_processing.h
+++ b/modules/audio_processing/include/mock_audio_processing.h
@@ -13,7 +13,6 @@
 
 #include <memory>
 
-#include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/include/aec_dump.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
@@ -128,8 +127,15 @@
   MOCK_CONST_METHOD0(delay_offset_ms, int());
   MOCK_METHOD1(set_stream_analog_level, void(int));
   MOCK_CONST_METHOD0(recommended_stream_analog_level, int());
-
-  virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) {}
+  MOCK_METHOD3(CreateAndAttachAecDump,
+               bool(const std::string& file_name,
+                    int64_t max_log_size_bytes,
+                    rtc::TaskQueue* worker_queue));
+  MOCK_METHOD3(CreateAndAttachAecDump,
+               bool(FILE* handle,
+                    int64_t max_log_size_bytes,
+                    rtc::TaskQueue* worker_queue));
+  MOCK_METHOD1(AttachAecDump, void(std::unique_ptr<AecDump>));
   MOCK_METHOD0(DetachAecDump, void());
 
   MOCK_METHOD0(GetStatistics, AudioProcessingStats());
diff --git a/test/pc/e2e/test_peer_factory.cc b/test/pc/e2e/test_peer_factory.cc
index 2b01d8d..0d08f8e 100644
--- a/test/pc/e2e/test_peer_factory.cc
+++ b/test/pc/e2e/test_peer_factory.cc
@@ -292,8 +292,8 @@
   rtc::scoped_refptr<AudioProcessing> audio_processing =
       webrtc::AudioProcessingBuilder().Create();
   if (params->aec_dump_path && audio_processing) {
-    audio_processing->AttachAecDump(
-        AecDumpFactory::Create(*params->aec_dump_path, -1, task_queue));
+    audio_processing->CreateAndAttachAecDump(*params->aec_dump_path, -1,
+                                             task_queue);
   }
   rtc::scoped_refptr<AudioDeviceModule> audio_device_module =
       CreateAudioDeviceModule(