Correction of the AEC3 underrun behavior and minor other corrections

webrtc/modules/audio_processing/aec3/render_delay_buffer.cc

 /*
  *  Copyright (c) 2017 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 106 matching lines @@
 RenderDelayBufferImpl::~RenderDelayBufferImpl() = default;
 
 void RenderDelayBufferImpl::Reset() {
   // Empty all data in the buffers.
   delay_ = 0;
   last_insert_index_ = 0;
   downsampled_render_buffer_.position = 0;
   downsampled_render_buffer_.buffer.fill(0.f);
   fft_buffer_.Clear();
   api_call_jitter_buffer_.Reset();
+  for (auto& c : buffer_) {
+    for (auto& b : c) {
+      std::fill(b.begin(), b.end(), 0.f);
+    }
+  }
 }
 
 bool RenderDelayBufferImpl::Insert(
     const std::vector<std::vector<float>>& block) {
   RTC_DCHECK_EQ(block.size(), buffer_[0].size());
   RTC_DCHECK_EQ(block[0].size(), buffer_[0][0].size());
 
   if (api_call_jitter_buffer_.Full()) {
     // Report buffer overrun and let the caller handle the overrun.
     return false;
@@ skipping 57 matching lines @@
   }
 }
 
 }  // namespace
 
 RenderDelayBuffer* RenderDelayBuffer::Create(size_t num_bands) {
   return new RenderDelayBufferImpl(num_bands);
 }
 
 }  // namespace webrtc