diff --git a/media/BUILD.gn b/media/BUILD.gn
index ab2f8f2c9c..dd19656bfa 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -291,7 +291,6 @@ rtc_static_library("rtc_audio_video") {
     "../rtc_base:checks",
     "../rtc_base:rtc_task_queue",
     "../rtc_base:stringutils",
-    "../rtc_base/experiments:audio_allocation_settings",
     "../rtc_base/experiments:experimental_screenshare_settings",
     "../rtc_base/experiments:field_trial_parser",
     "../rtc_base/experiments:normalize_simulcast_size_experiment",
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 9fe6f79056..189d7a68e8 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1024,7 +1024,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
-  const webrtc::AudioAllocationSettings allocation_settings_;
   webrtc::Call* call_ = nullptr;
   webrtc::AudioSendStream::Config config_;
   // The stream is owned by WebRtcAudioSendStream and may be reallocated if
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index a4c8baa37f..5ef2fde84e 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -27,7 +27,6 @@
 #include "media/engine/apm_helpers.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/experiments/audio_allocation_settings.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"
@@ -104,8 +103,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
-  const webrtc::AudioAllocationSettings allocation_settings_;
-
   // The audio device module.
   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
   rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;