Major AEC3 render pipeline changes

webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc

 /*
  *  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 75 matching lines @@
     // Float interface processing.
     // Verify interface invariance.
     RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
               interface_used_ == InterfaceType::kNotSpecified);
     interface_used_ = InterfaceType::kFloatInterface;
 
     RTC_CHECK_EQ(in_buf_->num_channels(),
                  static_cast<size_t>(msg.input_channel_size()));
 
     // Populate input buffer.
-    for (int i = 0; i < msg.input_channel_size(); ++i) {
+    for (size_t i = 0; i < in_buf_->num_channels(); ++i) {
       RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
                    msg.input_channel(i).size());
       std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
                   msg.input_channel(i).size());
     }
   }
 
   if (artificial_nearend_buffer_reader_) {
     if (artificial_nearend_buffer_reader_->Read(
             artificial_nearend_buf_.get())) {
@@ skipping 467 matching lines @@
 }
 
 void AecDumpBasedSimulator::HandleMessage(
     const webrtc::audioproc::ReverseStream& msg) {
   PrepareReverseProcessStreamCall(msg);
   ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
 }
 
 }  // namespace test
 }  // namespace webrtc