New module for the adaptive level controlling functionality in the audio processing module

webrtc/modules/audio_processing/logging/apm_data_dumper.h

 /*
  *  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 55 matching lines @@
     fwrite(v, sizeof(v[0]), v_length, file);
 #endif
   }
 
   void DumpRaw(const char* name, rtc::ArrayView<const float> v) {
 #if WEBRTC_AEC_DEBUG_DUMP == 1
     DumpRaw(name, v.size(), v.data());
 #endif
   }
 
+  void DumpRaw(const char* name, int v_length, const bool* v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    FILE* file = GetRawFile(name);
+    for (int k = 0; k < v_length; ++k) {
+      int16_t value = static_cast<int16_t>(v[k]);
+      fwrite(&value, sizeof(value), 1, file);
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const bool> v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
   void DumpRaw(const char* name, int v_length, const int16_t* v) {
 #if WEBRTC_AEC_DEBUG_DUMP == 1
     FILE* file = GetRawFile(name);
     fwrite(v, sizeof(v[0]), v_length, file);
 #endif
   }
 
   void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
 #if WEBRTC_AEC_DEBUG_DUMP == 1
     DumpRaw(name, v.size(), v.data());
@@ skipping 17 matching lines @@
                int v_length,
                const float* v,
                int sample_rate_hz,
                int num_channels) {
 #if WEBRTC_AEC_DEBUG_DUMP == 1
     WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
     file->WriteSamples(v, v_length);
 #endif
   }
 
+  void DumpWav(const char* name,
+               rtc::ArrayView<const float> v,
+               int sample_rate_hz,
+               int num_channels) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
+#endif
+  }
+
  private:
 #if WEBRTC_AEC_DEBUG_DUMP == 1
   const int instance_index_;
   int recording_set_index_ = 0;
   std::unordered_map<std::string, std::unique_ptr<FILE, RawFileCloseFunctor>>
       raw_files_;
   std::unordered_map<std::string, std::unique_ptr<WavWriter>> wav_files_;
 
   FILE* GetRawFile(const char* name);
   WavWriter* GetWavFile(const char* name, int sample_rate_hz, int num_channels);
 #endif
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ApmDataDumper);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_