Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

2 files changed, 365 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/third_party/libwebrtc/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
new file mode 100644
index 0000000000..fb25b74fff
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
@@ -0,0 +1,88 @@
+/*
+ *  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 <sstream>
+
+#include "rtc_base/stringutils.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
+std::string FormFileName(const char* name,
+                         int instance_index,
+                         int reinit_index,
+                         const std::string& 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() {}
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+FILE* ApmDataDumper::GetRawFile(const char* name) {
+  std::string filename =
+      FormFileName(name, instance_index_, recording_set_index_, ".dat");
+  auto& f = raw_files_[filename];
+  if (!f) {
+    f.reset(fopen(filename.c_str(), "wb"));
+  }
+  return f.get();
+}
+
+WavWriter* ApmDataDumper::GetWavFile(const char* name,
+                                     int sample_rate_hz,
+                                     int num_channels) {
+  std::string filename =
+      FormFileName(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));
+  }
+  return f.get();
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/webrtc/modules/audio_processing/logging/apm_data_dumper.h b/third_party/libwebrtc/webrtc/modules/audio_processing/logging/apm_data_dumper.h
new file mode 100644
index 0000000000..83ff8b19ad
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/modules/audio_processing/logging/apm_data_dumper.h
@@ -0,0 +1,277 @@
+/*
+ *  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 <stdio.h>
+
+#include <memory>
+#include <string>
+#include <unordered_map>
+
+#include "api/array_view.h"
+#include "common_audio/wav_file.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/logging.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 {
+
+#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();
+
+  // 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(const char* name, double v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, size_t v_length, const double* v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const double> v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpRaw(const char* name, float v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, size_t v_length, const float* v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const float> v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpRaw(const char* name, bool v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    DumpRaw(name, static_cast<int16_t>(v));
+#endif
+  }
+
+  void DumpRaw(const char* name, size_t v_length, const bool* v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        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_APM_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpRaw(const char* name, int16_t v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, size_t v_length, const int16_t* v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpRaw(const char* name, int32_t v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, size_t v_length, const int32_t* v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, size_t v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, size_t v_length, const size_t* v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const int32_t> v) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpWav(const char* name,
+               size_t v_length,
+               const float* v,
+               int sample_rate_hz,
+               int num_channels) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (rtc::LogMessage::aec_debug()) {
+      WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
+      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(const char* name,
+               rtc::ArrayView<const float> v,
+               int sample_rate_hz,
+               int num_channels) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
+#endif
+  }
+
+ private:
+#if WEBRTC_APM_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);
+
+  uint32_t debug_written_;
+
+  void updateDebugWritten(uint32_t amount) {
+    debug_written_ += amount;
+    // Limit largest files to a specific (rough) size, to avoid filling up disk.
+    if (debug_written_ >= rtc::LogMessage::aec_debug_size()) {
+      rtc::LogMessage::set_aec_debug(false);
+    }
+  }
+
+#endif
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ApmDataDumper);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_