2 files changed, 552 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc
new file mode 100644
index 0000000000..a15321ad48
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc
@@ -0,0 +1,100 @@
+/*
+ *  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 "modules/audio_processing/logging/apm_data_dumper.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/strings/string_builder.h"
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
+    (WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
+#error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+namespace {
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+
+#if defined(WEBRTC_WIN)
+constexpr char kPathDelimiter = '\\';
+#else
+constexpr char kPathDelimiter = '/';
+#endif
+
+std::string FormFileName(absl::string_view output_dir,
+                         absl::string_view name,
+                         int instance_index,
+                         int reinit_index,
+                         absl::string_view suffix) {
+#ifdef WEBRTC_WIN
+  char sep = '\\';
+#else
+  char sep = '/';
+#endif
+
+  std::stringstream ss;
+  std::string base = rtc::LogMessage::aec_debug_filename();
+  ss << base;
+
+  if (base.length() && base.back() != sep) {
+    ss << sep;
+  }
+
+  ss << name << "_" << instance_index << "-" << reinit_index << suffix;
+  return ss.str();
+}
+#endif
+
+}  // namespace
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+ApmDataDumper::ApmDataDumper(int instance_index)
+    : instance_index_(instance_index)
+    , debug_written_(0) {}
+#else
+ApmDataDumper::ApmDataDumper(int instance_index) {}
+#endif
+
+ApmDataDumper::~ApmDataDumper() = default;
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+bool ApmDataDumper::recording_activated_ = false;
+absl::optional<int> ApmDataDumper::dump_set_to_use_;
+char ApmDataDumper::output_dir_[] = "";
+
+FILE* ApmDataDumper::GetRawFile(absl::string_view name) {
+  std::string filename = FormFileName(output_dir_, name, instance_index_,
+                                      recording_set_index_, ".dat");
+  auto& f = raw_files_[filename];
+  if (!f) {
+    f.reset(fopen(filename.c_str(), "wb"));
+    RTC_CHECK(f.get()) << "Cannot write to " << filename << ".";
+  }
+  return f.get();
+}
+
+WavWriter* ApmDataDumper::GetWavFile(absl::string_view name,
+                                     int sample_rate_hz,
+                                     int num_channels,
+                                     WavFile::SampleFormat format) {
+  std::string filename = FormFileName(output_dir_, name, instance_index_,
+                                      recording_set_index_, ".wav");
+  auto& f = wav_files_[filename];
+  if (!f) {
+    f.reset(
+        new WavWriter(filename.c_str(), sample_rate_hz, num_channels, format));
+  }
+  return f.get();
+}
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h
new file mode 100644
index 0000000000..aa8496819b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h
@@ -0,0 +1,452 @@
+/*
+ *  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 MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+#define MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+
+#include <stdint.h>
+#include <stdio.h>
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+#include <memory>
+#include <string>
+#include <unordered_map>
+#endif
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#if WEBRTC_APM_DEBUG_DUMP == 1
+#include "common_audio/wav_file.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_utils.h"
+#endif
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
+    (WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
+#error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+// Functor used to use as a custom deleter in the map of file pointers to raw
+// files.
+struct RawFileCloseFunctor {
+  void operator()(FILE* f) const { if (f) fclose(f); }
+};
+#endif
+
+// 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.
+  explicit ApmDataDumper(int instance_index);
+
+  ApmDataDumper() = delete;
+  ApmDataDumper(const ApmDataDumper&) = delete;
+  ApmDataDumper& operator=(const ApmDataDumper&) = delete;
+
+  ~ApmDataDumper();
+
+  // Activates or deactivate the dumping functionality.
+  static void SetActivated(bool activated) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    recording_activated_ = activated;
+#endif
+  }
+
+  // Returns whether dumping functionality is enabled/available.
+  static bool IsAvailable() {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    return true;
+#else
+    return false;
+#endif
+  }
+
+  // Default dump set.
+  static constexpr size_t kDefaultDumpSet = 0;
+
+  // Specifies what dump set to use. All dump commands with a different dump set
+  // than the one specified will be discarded. If not specificed, all dump sets
+  // will be used.
+  static void SetDumpSetToUse(int dump_set_to_use) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    dump_set_to_use_ = dump_set_to_use;
+#endif
+  }
+
+  // Set an optional output directory.
+  static void SetOutputDirectory(absl::string_view output_dir) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength);
+    rtc::strcpyn(output_dir_, output_dir.size(), output_dir);
+#endif
+  }
+
+  // Reinitializes the data dumping such that new versions
+  // of all files being dumped to are created.
+  void InitiateNewSetOfRecordings() {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    ++recording_set_index_;
+    debug_written_ = 0;
+#endif
+  }
+
+  // Methods for performing dumping of data of various types into
+  // various formats.
+  void DumpRaw(absl::string_view name,
+               double v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const double* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const double> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               float v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const float* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const float> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name, bool v, int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, static_cast<int16_t>(v));
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const bool* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        for (size_t k = 0; k < v_length; ++k) {
+          int16_t value = static_cast<int16_t>(v[k]);
+          fwrite(&value, sizeof(value), 1, file);
+        }
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const bool> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               int16_t v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const int16_t* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const int16_t> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               int32_t v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const int32_t* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const size_t* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const int32_t> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const size_t> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpWav(absl::string_view name,
+               size_t v_length,
+               const float* v,
+               int sample_rate_hz,
+               int num_channels,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels,
+                                   WavFile::SampleFormat::kFloat);
+      file->WriteSamples(v, v_length);
+      // Cheat and use aec_near as a stand-in for "size of the largest file"
+      // in the dump.  We're looking to limit the total time, and that's a
+      // reasonable stand-in.
+      if (strcmp(name, "aec_near") == 0) {
+       updateDebugWritten(v_length * sizeof(float));
+      }
+    }
+#endif
+  }
+
+  void DumpWav(absl::string_view name,
+               rtc::ArrayView<const float> v,
+               int sample_rate_hz,
+               int num_channels,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
+    }
+#endif
+  }
+
+ private:
+#if WEBRTC_APM_DEBUG_DUMP == 1
+  static bool recording_activated_;
+  static absl::optional<int> dump_set_to_use_;
+  static constexpr size_t kOutputDirMaxLength = 1024;
+  static char output_dir_[kOutputDirMaxLength];
+  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(absl::string_view name);
+  WavWriter* GetWavFile(absl::string_view name,
+                        int sample_rate_hz,
+                        int num_channels,
+                        WavFile::SampleFormat format);
+
+  uint32_t debug_written_ = 0;
+
+  void updateDebugWritten(uint32_t amount) {
+    debug_written_ += amount;
+    if (debug_written_ >= webrtc::Trace::aec_debug_size()) {
+      SetActivated(false);
+    }
+  }
+
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_