Remove header files for the AEC and the APM test program that are no longer used.

webrtc/modules/audio_processing/test/aec_dump_processor.h

Index: webrtc/modules/audio_processing/test/aec_dump_processor.h
diff --git a/webrtc/modules/audio_processing/test/aec_dump_processor.h b/webrtc/modules/audio_processing/test/aec_dump_processor.h
deleted file mode 100644
index 12b5878aadf2c6695dc2acf219f60adbd6280791..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/test/aec_dump_processor.h
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_
-
-#include <algorithm>
-#include <limits>
-#include <memory>
-#include <string>
-#include <vector>
-
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/optional.h"
-#include "webrtc/common_audio/channel_buffer.h"
-#include "webrtc/common_audio/wav_file.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
-#include "webrtc/modules/audio_processing/test/test_utils.h"
-
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
-#else
-#include "webrtc/modules/audio_processing/debug.pb.h"
-#endif
-
-namespace webrtc {
-namespace test {
-
-// Used to read from an aecdump file and write to a WavWriter.
-class AecDumpFileProcessor final : public AudioFileProcessor {
- public:
-  AecDumpFileProcessor(std::unique_ptr<AudioProcessing> ap,
-                       FILE* dump_file,
-                       std::string out_filename,
-                       std::string reverse_out_filename,
-                       rtc::Optional<int> out_sample_rate_hz,
-                       rtc::Optional<int> out_num_channels,
-                       rtc::Optional<int> reverse_out_sample_rate_hz,
-                       rtc::Optional<int> reverse_out_num_channels,
-                       bool override_config_message);
-
-  virtual ~AecDumpFileProcessor();
-
-  // Processes the messages in the aecdump file and returns
-  // the number of forward stream chunks processed.
-  size_t Process(bool verbose_logging) override;
-
- private:
-  void HandleMessage(const webrtc::audioproc::Init& msg);
-  void HandleMessage(const webrtc::audioproc::Stream& msg);
-  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
-  void HandleMessage(const webrtc::audioproc::Config& msg);
-
-  enum InterfaceType {
-    kIntInterface,
-    kFloatInterface,
-    kNotSpecified,
-  };
-
-  std::unique_ptr<AudioProcessing> ap_;
-  FILE* dump_file_;
-  std::string out_filename_;
-  std::string reverse_out_filename_;
-  rtc::Optional<int> out_sample_rate_hz_;
-  rtc::Optional<int> out_num_channels_;
-  rtc::Optional<int> reverse_out_sample_rate_hz_;
-  rtc::Optional<int> reverse_out_num_channels_;
-  bool override_config_message_;
-
-  std::unique_ptr<ChannelBuffer<float>> in_buf_;
-  std::unique_ptr<ChannelBuffer<float>> reverse_buf_;
-  std::unique_ptr<ChannelBuffer<float>> out_buf_;
-  std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_;
-  std::unique_ptr<WavWriter> out_file_;
-  std::unique_ptr<WavWriter> reverse_out_file_;
-  StreamConfig input_config_;
-  StreamConfig reverse_config_;
-  StreamConfig output_config_;
-  StreamConfig reverse_output_config_;
-  std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
-  std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
-  AudioFrame far_frame_;
-  AudioFrame near_frame_;
-  InterfaceType interface_used_ = InterfaceType::kNotSpecified;
-};
-
-}  // namespace test
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_