diff --git a/src/modules/audio_processing/aec/aec_core.c b/src/modules/audio_processing/aec/aec_core.c
index 1637e6fdbc..93ec26b0d0 100644
--- a/src/modules/audio_processing/aec/aec_core.c
+++ b/src/modules/audio_processing/aec/aec_core.c
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -543,19 +543,13 @@ void WebRtcAec_InitMetrics(aec_t *aec)
     WebRtcAec_InitStats(&aec->rerl);
 }
 
-
 void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend) {
   float fft[PART_LEN2];
   float xf[2][PART_LEN1];
 
   // Check if the buffer is full, and in that case flush the oldest data.
   if (WebRtc_available_write(aec->far_buf) < 1) {
-    WebRtc_MoveReadPtr(aec->far_buf, 1);
-    WebRtc_MoveReadPtr(aec->far_buf_windowed, 1);
-    aec->system_delay -= PART_LEN;
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-    WebRtc_MoveReadPtr(aec->far_time_buf, 1);
-#endif
+    WebRtcAec_MoveFarReadPtr(aec, 1);
   }
   // Convert far-end partition to the frequency domain without windowing.
   memcpy(fft, farend, sizeof(float) * PART_LEN2);
@@ -568,6 +562,16 @@ void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend) {
   WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
 }
 
+int WebRtcAec_MoveFarReadPtr(aec_t *aec, int elements) {
+  int elements_moved = WebRtc_MoveReadPtr(aec->far_buf_windowed, elements);
+  WebRtc_MoveReadPtr(aec->far_buf, elements);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+  WebRtc_MoveReadPtr(aec->far_time_buf, elements);
+#endif
+  aec->system_delay -= elements_moved * PART_LEN;
+  return elements_moved;
+}
+
 void WebRtcAec_ProcessFrame(aec_t *aec,
                             const short *nearend,
                             const short *nearendH,
@@ -608,16 +612,10 @@ void WebRtcAec_ProcessFrame(aec_t *aec,
     // |system_delay| indicates others.
     if (aec->system_delay < FRAME_LEN) {
       // We don't have enough data so we rewind 10 ms.
-      WebRtc_MoveReadPtr(aec->far_buf_windowed, -(aec->mult + 1));
-      aec->system_delay -= WebRtc_MoveReadPtr(aec->far_buf, -(aec->mult + 1)) *
-          PART_LEN;
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-      WebRtc_MoveReadPtr(aec->far_time_buf, -(aec->mult + 1));
-#endif
+      WebRtcAec_MoveFarReadPtr(aec, -(aec->mult + 1));
     }
 
     // 2) Compensate for a possible change in the system delay.
-
     WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
     moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
     aec->knownDelay -= moved_elements * PART_LEN;
diff --git a/src/modules/audio_processing/aec/aec_core.h b/src/modules/audio_processing/aec/aec_core.h
index d326a6842d..c07528d644 100644
--- a/src/modules/audio_processing/aec/aec_core.h
+++ b/src/modules/audio_processing/aec/aec_core.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -177,4 +177,9 @@ void WebRtcAec_ProcessFrame(aec_t* aec,
                             const short *nearendH,
                             int knownDelay);
 
+// A helper function to call WebRtc_MoveReadPtr() for all far-end buffers.
+// Returns the number of elements moved, and adjusts |system_delay| by the
+// corresponding amount in ms.
+int WebRtcAec_MoveFarReadPtr(aec_t* aec, int elements);
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
diff --git a/src/modules/audio_processing/aec/echo_cancellation.c b/src/modules/audio_processing/aec/echo_cancellation.c
index bde9c87c5b..5d690b9e62 100644
--- a/src/modules/audio_processing/aec/echo_cancellation.c
+++ b/src/modules/audio_processing/aec/echo_cancellation.c
@@ -526,18 +526,12 @@ WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
                 // Enable the AEC
                 aecpc->ECstartup = 0;
             } else if (overhead_elements > 0) {
-                WebRtc_MoveReadPtr(aecpc->aec->far_buf_windowed,
-                                   overhead_elements);
-                WebRtc_MoveReadPtr(aecpc->aec->far_buf, overhead_elements);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-                WebRtc_MoveReadPtr(aecpc->aec->far_time_buf, overhead_elements);
-#endif
                 // TODO(bjornv): Do we need a check on how much we actually
                 // moved the read pointer? It should always be possible to move
                 // the pointer |overhead_elements| since we have only added data
                 // to the buffer and no delay compensation nor AEC processing
                 // has been done.
-                aecpc->aec->system_delay -= overhead_elements * PART_LEN;
+                WebRtcAec_MoveFarReadPtr(aecpc->aec, overhead_elements);
 
                 // Enable the AEC
                 aecpc->ECstartup = 0;
@@ -895,6 +889,10 @@ static int EstBufDelay(aecpc_t* aecpc) {
     current_delay -= kResamplingDelay;
   }
 
+  if (current_delay < PART_LEN) {
+    current_delay += WebRtcAec_MoveFarReadPtr(aecpc->aec, 1) * PART_LEN;
+  }
+
   aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.8 * aecpc->filtDelay +
           0.2 * current_delay));
 
diff --git a/test/data/audio_processing/output_data_float.pb b/test/data/audio_processing/output_data_float.pb
index d9a51bbabf..b2b934dd86 100644
Binary files a/test/data/audio_processing/output_data_float.pb and b/test/data/audio_processing/output_data_float.pb differ