| Index: webrtc/modules/audio_processing/logging/apm_data_dumper.cc
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| diff --git a/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..97233693a66c5f56e6b0cefbf4cee6e2536c3e81
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| --- /dev/null
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| +++ b/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
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| @@ -0,0 +1,85 @@
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| +/*
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| + *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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| + *
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| + *  Use of this source code is governed by a BSD-style license
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| + *  that can be found in the LICENSE file in the root of the source
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| + *  tree. An additional intellectual property rights grant can be found
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| + *  in the file PATENTS.  All contributing project authors may
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| + *  be found in the AUTHORS file in the root of the source tree.
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| + */
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| +
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| +#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
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| +
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| +#include "webrtc/base/stringutils.h"
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| +
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| +// Check to verify that the define is properly set.
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| +#if !defined(WEBRTC_AEC_DEBUG_DUMP) || \
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| +    (WEBRTC_AEC_DEBUG_DUMP != 0 && WEBRTC_AEC_DEBUG_DUMP != 1)
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| +#error "Set WEBRTC_AEC_DEBUG_DUMP to either 0 or 1"
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| +#endif
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| +
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| +namespace webrtc {
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| +
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| +namespace {
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| +
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| +#if WEBRTC_AEC_DEBUG_DUMP == 1
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| +std::string FormFileName(const std::string& name,
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| +                         int instance_index,
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| +                         int reinit_index,
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| +                         const std::string& suffix) {
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| +  char instance_index_string[10];
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| +  rtc::sprintfn(instance_index_string, sizeof(instance_index_string), "%d",
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| +                instance_index);
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| +  char reinit_index_string[10];
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| +  rtc::sprintfn(reinit_index_string, sizeof(reinit_index_string), "%d",
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| +                reinit_index);
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| +  return name + "_" + instance_index_string + "-" +
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| +         std::to_string(reinit_index) + suffix;
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| +}
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| +#endif
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| +
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| +}  // namespace
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| +
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| +ApmDataDumper::~ApmDataDumper() {
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| +#if WEBRTC_AEC_DEBUG_DUMP == 1
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| +  for (auto& raw_files_element : raw_files_) {
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| +    fclose(raw_files_element.second);
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| +  }
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| +
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| +  // Deleting the wav files implicitly causes the files to be closed.
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| +  for (auto& wav_files_element : wav_files_) {
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| +    delete wav_files_element.second;
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| +  }
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| +#endif
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| +}
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| +
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| +#if WEBRTC_AEC_DEBUG_DUMP == 1
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| +FILE* ApmDataDumper::GetRawFile(const std::string& name) {
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| +  std::string filename =
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| +      FormFileName(name, instance_index_, recording_set_index_, ".dat");
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| +  auto search = raw_files_.find(filename);
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| +  if (search != raw_files_.end()) {
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| +    return search->second;
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| +  }
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| +  FILE* file = fopen(filename.c_str(), "wb");
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| +  raw_files_[filename] = file;
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| +  return file;
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| +}
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| +
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| +WavWriter* ApmDataDumper::GetWavFile(const std::string& name,
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| +                                     int sample_rate_hz,
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| +                                     int num_channels) {
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| +  std::string filename =
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| +      FormFileName(name, instance_index_, recording_set_index_, ".wav");
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| +  auto search = wav_files_.find(filename);
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| +  if (search != wav_files_.end()) {
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| +    return search->second;
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| +  }
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| +  WavWriter* file =
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| +      new WavWriter(filename.c_str(), sample_rate_hz, num_channels);
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| +  wav_files_[filename] = file;
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| +  return file;
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| +}
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| +#endif
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| +
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| +}  // namespace webrtc
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| 
 |