Make AEC3 recover more quickly for lost capture data

webrtc/modules/audio_processing/aec3/aec3_common.h

 /*
  *  Copyright (c) 2016 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 45 matching lines @@
     kMatchedFilterWindowSizeSubBlocks * 3 / 4;
 constexpr size_t kDownsampledRenderBufferSize =
     kSubBlockSize *
     (kMatchedFilterAlignmentShiftSizeSubBlocks * kNumMatchedFilters +
      kMatchedFilterWindowSizeSubBlocks +
      1);
 
 constexpr size_t kRenderDelayBufferSize =
     (3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize);
 
+constexpr size_t kMinEchoPathDelayBlocks = 5;
 constexpr size_t kMaxApiCallsJitterBlocks = 30;
 constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2;
 static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks,
               "Requirement to ensure buffer overflow detection");
 
 // TODO(peah): Integrate this with how it is done inside audio_processing_impl.
 constexpr size_t NumBandsForRate(int sample_rate_hz) {
   return static_cast<size_t>(sample_rate_hz == 8000 ? 1
                                                     : sample_rate_hz / 16000);
 }
@@ skipping 29 matching lines @@
 static_assert(ValidFullBandRate(32000),
               "Test that 32 kHz is a valid sample rate");
 static_assert(ValidFullBandRate(48000),
               "Test that 48 kHz is a valid sample rate");
 static_assert(!ValidFullBandRate(8001),
               "Test that 8001 Hz is not a valid sample rate");
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_