diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 0e615ca..3214ce6 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -146,20 +146,22 @@
                           AudioFrame* audio_frame,
                           bool* muted) {
   RTC_DCHECK(muted);
-  // Accessing members, take the lock.
-  MutexLock lock(&mutex_);
 
-  if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
+  int current_sample_rate_hz = 0;
+  if (neteq_->GetAudio(audio_frame, muted, &current_sample_rate_hz) !=
+      NetEq::kOK) {
     RTC_LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     return -1;
   }
 
-  const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
+  RTC_DCHECK_NE(current_sample_rate_hz, 0);
 
   // Update if resampling is required.
   const bool need_resampling =
       (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
 
+  // Accessing members, take the lock.
+  MutexLock lock(&mutex_);
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
@@ -174,8 +176,8 @@
     }
   }
 
-  // TODO(henrik.lundin) Glitches in the output may appear if the output rate
-  // from NetEq changes. See WebRTC issue 3923.
+  // TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
+  // rate from NetEq changes.
   if (need_resampling) {
     // TODO(yujo): handle this more efficiently for muted frames.
     int samples_per_channel_int = resampler_.Resample10Msec(