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 71 matching lines @@
                      fwd_frame_.num_channels_,
                  msg.input_data().size());
     memcpy(fwd_frame_.data_, msg.input_data().data(), msg.input_data().size());
   } else {
     // 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()));
+    // 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());

[aleloi, 2017/04/04 12:18:50]

Here we read from msg.input_channel(i) index i for i = 0 .. num_channels().

With the DCHECK above commented out now, how do we know it's safe? Should it
perhaps change to DCHECK_LE instead?

Also, why has this changed?

[peah-webrtc, 2017/04/04 13:57:12]

That was done by mistake (while analyzing a 4 channel recording. 

Reverted that change.

Done.

     }
   }
 
   if (artificial_nearend_buffer_reader_) {
     if (artificial_nearend_buffer_reader_->Read(
             artificial_nearend_buf_.get())) {
       if (msg.has_input_data()) {
         for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
           fwd_frame_.data_[k] = rtc::saturated_cast<int16_t>(
               fwd_frame_.data_[k] +
@@ skipping 463 matching lines @@
 }
 
 void AecDumpBasedSimulator::HandleMessage(
     const webrtc::audioproc::ReverseStream& msg) {
   PrepareReverseProcessStreamCall(msg);
   ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
 }
 
 }  // namespace test
 }  // namespace webrtc