diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 8f21e89448..8be08a2097 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -41,9 +41,7 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
   memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
 }
 
-AcmReceiver::~AcmReceiver() {
-  delete neteq_;
-}
+AcmReceiver::~AcmReceiver() = default;
 
 int AcmReceiver::SetMinimumDelay(int delay_ms) {
   if (neteq_->SetMinimumDelay(delay_ms))
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 87190c6e67..b374495b3b 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -281,7 +281,7 @@ class AcmReceiver {
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
-  NetEq* neteq_;
+  const std::unique_ptr<NetEq> neteq_;  // NetEq is thread-safe; no lock needed.
   const Clock* const clock_;
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
   rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);