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

webrtc/modules/audio_processing/logging/apm_data_dumper.cc

+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+#include "webrtc/base/stringutils.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 {
+
+namespace {
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+std::string FormFileName(const std::string& name,
+                         int instance_index,
+                         int reinit_index,
+                         const std::string& suffix) {
+  char instance_index_string[10];

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

I think the C++ way of doing this (in the absence of std::to_string) is to use a
stringstream; see for instance
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc....

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

Great! Thanks!! With that the code became much nicer!
Done.

+  rtc::sprintfn(instance_index_string, sizeof(instance_index_string), "%d",
+                instance_index);
+  char reinit_index_string[10];
+  rtc::sprintfn(reinit_index_string, sizeof(reinit_index_string), "%d",
+                reinit_index);
+  return name + "_" + instance_index_string + "-" +
+         std::to_string(reinit_index) + suffix;
+}
+#endif
+
+}  // namespace
+
+ApmDataDumper::~ApmDataDumper() {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+  for (auto& raw_files_element : raw_files_) {
+    fclose(raw_files_element.second);
+  }
+
+  // Deleting the wav files implicitly causes the files to be closed.
+  for (auto& wav_files_element : wav_files_) {
+    delete wav_files_element.second;
+  }
+#endif
+}
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+FILE* ApmDataDumper::GetRawFile(const std::string& name) {
+  std::string filename =
+      FormFileName(name, instance_index_, recording_set_index_, ".dat");
+  auto search = raw_files_.find(filename);
+  if (search != raw_files_.end()) {
+    return search->second;
+  }
+  FILE* file = fopen(filename.c_str(), "wb");
+  raw_files_[filename] = file;
+  return file;
+}
+
+WavWriter* ApmDataDumper::GetWavFile(const std::string& name,
+                                     int sample_rate_hz,
+                                     int num_channels) {
+  std::string filename =
+      FormFileName(name, instance_index_, recording_set_index_, ".wav");
+  auto search = wav_files_.find(filename);
+  if (search != wav_files_.end()) {
+    return search->second;
+  }
+  WavWriter* file =
+      new WavWriter(filename.c_str(), sample_rate_hz, num_channels);
+  wav_files_[filename] = file;
+  return file;
+}
+#endif
+
+}  // namespace webrtc