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 113 matching lines @@
       input_file_channels_(input_channels),
       reverse_audio_(reverse_file_name, reverse_rate_hz, reverse_rate_hz),
       reverse_file_channels_(reverse_channels),
       input_(new ChannelBuffer<float>(input_config_.num_frames(),
                                       input_config_.num_channels())),
       reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(),
                                         reverse_config_.num_channels())),
       output_(new ChannelBuffer<float>(output_config_.num_frames(),
                                        output_config_.num_channels())),
       apm_(AudioProcessing::Create(config)),
-      dump_file_name_(dump_file_name) {
-}
+      dump_file_name_(dump_file_name) {}
 
 DebugDumpGenerator::DebugDumpGenerator(
     const Config& config,
     const AudioProcessing::Config& apm_config)
     : DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"),
                          32000,
                          2,
                          ResourcePath("far32_stereo", "pcm"),
                          32000,
                          2,
@@ skipping 442 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