Extension and refactoring of the audioproc_f tool to be a fully fledged tool for audio processing m…

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
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index 0000000000000000000000000000000000000000..12b5878aadf2c6695dc2acf219f60adbd6280791
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+/*
+ *  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_