Replaced the data logging functionality in the AEC with a generic logging functionality

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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+
+#include <stdio.h>
+
+#include <map>
+#include <string>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_audio/wav_file.h"
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_AEC_DEBUG_DUMP) || \
+    (WEBRTC_AEC_DEBUG_DUMP != 0 && WEBRTC_AEC_DEBUG_DUMP != 1)
+#error "Set WEBRTC_AEC_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+
+// Class that handles dumping of variables into files.
+class ApmDataDumper {
+ public:
+// Constructor that takes an instance index that may
+// be used to distinguish data dumped from different
+// instances of the code.
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+  explicit ApmDataDumper(int instance_index)
+      : instance_index_(instance_index) {}
+#else
+  explicit ApmDataDumper(int instance_index) {}
+#endif
+
+  ~ApmDataDumper();
+
+  // Reitializes the data dumping such that new versions

[hlundin-webrtc, 2016/04/25 11:29:59]

Reinitializes

[peah-webrtc, 2016/04/25 11:46:40]

Thanks!
Done.

+  // of all files being dumped to are created.
+  void InitiateNewSetOfRecordings() {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    ++recording_set_index_;
+#endif
+  }
+
+  // Methods for performing dumping of data of various types into
+  // various formats.
+  void DumpRaw(const std::string& name, int v_length, const float* v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    FILE* file = GetRawFile(name);
+    fwrite(v, sizeof(v[0]), v_length, file);
+#endif
+  }
+
+  void DumpRaw(const std::string& name, rtc::ArrayView<const float> v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpWav(const std::string& name,
+               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 DumpMonoWav(const std::string& 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::map<std::string, FILE*> raw_files_;
+  std::map<std::string, WavWriter*> wav_files_;
+
+  FILE* GetRawFile(const std::string& name);
+  WavWriter* GetWavFile(const std::string& 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_