Refactoring of the buffering of the nearend signal done inside the AEC

webrtc/modules/audio_processing/aec/echo_cancellation.cc

 /*
  *  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
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 665 matching lines @@
 
 static void ProcessExtended(Aec* self,
                             const float* const* near,
                             size_t num_bands,
                             float* const* out,
                             size_t num_samples,
                             int16_t reported_delay_ms,
                             int32_t skew) {
   size_t i;
   const int delay_diff_offset = kDelayDiffOffsetSamples;
+  RTC_DCHECK(num_samples == 80 || num_samples == 160);
 #if defined(WEBRTC_UNTRUSTED_DELAY)
   reported_delay_ms = kFixedDelayMs;
 #else
   // This is the usual mode where we trust the reported system delay values.
   // Due to the longer filter, we no longer add 10 ms to the reported delay
   // to reduce chance of non-causality. Instead we apply a minimum here to avoid
   // issues with the read pointer jumping around needlessly.
   reported_delay_ms = reported_delay_ms < kMinTrustedDelayMs
                           ? kMinTrustedDelayMs
                           : reported_delay_ms;
@@ skipping 153 matching lines @@
   } else {
     self->timeForDelayChange = 0;
   }
   self->lastDelayDiff = delay_difference;
 
   if (self->timeForDelayChange > 25) {
     self->knownDelay = WEBRTC_SPL_MAX((int)self->filtDelay - 256, 0);
   }
 }
 }  // namespace webrtc