AecDump interface

webrtc/modules/audio_processing/test/debug_dump_test.cc

 /*
  *  Copyright (c) 2015 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 169 matching lines @@
   output_config_.set_sample_rate_hz(rate_hz);
   MaybeResetBuffer(&output_, output_config_);
 }
 
 void DebugDumpGenerator::SetOutputChannels(int channels) {
   output_config_.set_num_channels(channels);
   MaybeResetBuffer(&output_, output_config_);
 }
 
 void DebugDumpGenerator::StartRecording() {
-  apm_->StartDebugRecording(dump_file_name_.c_str(), -1);
+  apm_->StartDebugRecording(dump_file_name_.c_str(), -1, nullptr);
 }
 
 void DebugDumpGenerator::Process(size_t num_blocks) {
   for (size_t i = 0; i < num_blocks; ++i) {
     ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_,
                         reverse_config_, reverse_->channels());
     ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
                         input_->channels());
     RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
     apm_->set_stream_key_pressed(i % 10 == 9);
@@ skipping 387 matching lines @@
   config.Set<ExperimentalNs>(new ExperimentalNs(true));
   DebugDumpGenerator generator(config, AudioProcessing::Config());
   generator.StartRecording();
   generator.Process(100);
   generator.StopRecording();
   VerifyDebugDump(generator.dump_file_name());
 }
 
 }  // namespace test
 }  // namespace webrtc