diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index b7a7262af7..bf22a290b8 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -74,6 +74,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
     experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
   }
 
+  WEBRTC_STUB_CONST(input_sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
   WEBRTC_STUB_CONST(num_input_channels, ());
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index e99720ddb4..67ad266d50 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -501,6 +501,12 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) {
   }
 }
 
+int AudioProcessingImpl::input_sample_rate_hz() const {
+  // Accessed from outside APM, hence a lock is needed.
+  rtc::CritScope cs(&crit_capture_);
+  return formats_.api_format.input_stream().sample_rate_hz();
+}
+
 int AudioProcessingImpl::proc_sample_rate_hz() const {
   // Used as callback from submodules, hence locking is not allowed.
   return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index eb349769d0..87a145ab3c 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -81,6 +81,7 @@ class AudioProcessingImpl : public AudioProcessing {
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
+  int input_sample_rate_hz() const override;
 
   // Render-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the render lock.
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index c8ddc6a483..5fcc4d4672 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -279,6 +279,10 @@ class AudioProcessing {
   // ensures the options are applied immediately.
   virtual void SetExtraOptions(const Config& config) = 0;
 
+  // TODO(peah): Remove after voice engine no longer requires it to resample
+  // the reverse stream to the forward rate.
+  virtual int input_sample_rate_hz() const = 0;
+
   // TODO(ajm): Only intended for internal use. Make private and friend the
   // necessary classes?
   virtual int proc_sample_rate_hz() const = 0;
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index d46ee6296d..d3a4f7fc8f 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -517,8 +517,20 @@ OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
     }
 
     // --- Far-end Voice Quality Enhancement (AudioProcessing Module)
-    if (feed_data_to_apm)
-      APMProcessReverseStream();
+    if (feed_data_to_apm) {
+      // Convert from mixing to AudioProcessing sample rate, similarly to how it
+      // is done on the send side. Downmix to mono.
+      AudioFrame frame;
+      frame.num_channels_ = 1;
+      frame.sample_rate_hz_ = _audioProcessingModulePtr->input_sample_rate_hz();
+      RemixAndResample(_audioFrame, &audioproc_resampler_, &frame);
+
+      if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) != 0) {
+        WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+                     "AudioProcessingModule::AnalyzeReverseStream() => error");
+        RTC_DCHECK(false);
+      }
+    }
 
     // --- External media processing
     {
@@ -549,13 +561,6 @@ OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
 //                             Private methods
 // ----------------------------------------------------------------------------
 
-void OutputMixer::APMProcessReverseStream() {
-  if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
-                 "AudioProcessingModule::ProcessReverseStream() => error");
-  }
-}
-
 int
 OutputMixer::InsertInbandDtmfTone()
 {
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index 8c0a2de53b..c042a038dd 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -102,7 +102,6 @@ public:
 
 private:
     OutputMixer(uint32_t instanceId);
-    void APMProcessReverseStream();
     int InsertInbandDtmfTone();
 
     // uses