Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/modules/audio_processing/include/aec_dump.cc

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2017 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/include/aec_dump.h"
+
+namespace webrtc {
+InternalAPMConfig::InternalAPMConfig() = default;
+InternalAPMConfig::InternalAPMConfig(const InternalAPMConfig&) = default;
+InternalAPMConfig::InternalAPMConfig(InternalAPMConfig&&) = default;
+InternalAPMConfig& InternalAPMConfig::operator=(const InternalAPMConfig&) =
+    default;
+
+bool InternalAPMConfig::operator==(const InternalAPMConfig& other) {
+  return aec_enabled == other.aec_enabled &&
+         aec_delay_agnostic_enabled == other.aec_delay_agnostic_enabled &&
+         aec_drift_compensation_enabled ==
+             other.aec_drift_compensation_enabled &&
+         aec_extended_filter_enabled == other.aec_extended_filter_enabled &&
+         aec_suppression_level == other.aec_suppression_level &&
+         aecm_enabled == other.aecm_enabled &&
+         aecm_comfort_noise_enabled == other.aecm_comfort_noise_enabled &&
+         aecm_routing_mode == other.aecm_routing_mode &&
+         agc_enabled == other.agc_enabled && agc_mode == other.agc_mode &&
+         agc_limiter_enabled == other.agc_limiter_enabled &&
+         hpf_enabled == other.hpf_enabled && ns_enabled == other.ns_enabled &&
+         ns_level == other.ns_level &&
+         transient_suppression_enabled == other.transient_suppression_enabled &&
+         noise_robust_agc_enabled == other.noise_robust_agc_enabled &&
+         pre_amplifier_enabled == other.pre_amplifier_enabled &&
+         pre_amplifier_fixed_gain_factor ==
+             other.pre_amplifier_fixed_gain_factor &&
+         experiments_description == other.experiments_description;
+}
+}  // namespace webrtc

webrtc/modules/audio_processing/include/aec_dump.h

@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2017 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_INCLUDE_AEC_DUMP_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/deprecation.h"
+
+namespace webrtc {
+
+// Struct for passing current config from APM without having to
+// include protobuf headers.
+struct InternalAPMConfig {
+  InternalAPMConfig();
+  InternalAPMConfig(const InternalAPMConfig&);
+  InternalAPMConfig(InternalAPMConfig&&);
+
+  InternalAPMConfig& operator=(const InternalAPMConfig&);
+  InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
+
+  bool operator==(const InternalAPMConfig& other);
+
+  bool aec_enabled = false;
+  bool aec_delay_agnostic_enabled = false;
+  bool aec_drift_compensation_enabled = false;
+  bool aec_extended_filter_enabled = false;
+  int aec_suppression_level = 0;
+  bool aecm_enabled = false;
+  bool aecm_comfort_noise_enabled = false;
+  int aecm_routing_mode = 0;
+  bool agc_enabled = false;
+  int agc_mode = 0;
+  bool agc_limiter_enabled = false;
+  bool hpf_enabled = false;
+  bool ns_enabled = false;
+  int ns_level = 0;
+  bool transient_suppression_enabled = false;
+  bool noise_robust_agc_enabled = false;
+  bool pre_amplifier_enabled = false;
+  float pre_amplifier_fixed_gain_factor = 1.f;
+  std::string experiments_description = "";
+};
+
+// An interface for recording configuration and input/output streams
+// of the Audio Processing Module. The recordings are called
+// 'aec-dumps' and are stored in a protobuf format defined in
+// debug.proto.
+// The Write* methods are always safe to call concurrently or
+// otherwise for all implementing subclasses. The intended mode of
+// operation is to create a protobuf object from the input, and send
+// it away to be written to file asynchronously.
+class AecDump {
+ public:
+  struct AudioProcessingState {
+    int delay;
+    int drift;
+    int level;
+    bool keypress;
+  };
+
+  virtual ~AecDump() = default;
+
+  // Logs Event::Type INIT message.
+  virtual void WriteInitMessage(const ProcessingConfig& api_format,
+                                int64_t time_now_ms) = 0;
+  RTC_DEPRECATED void WriteInitMessage(const ProcessingConfig& api_format) {
+    WriteInitMessage(api_format, 0);
+  }
+
+  // Logs Event::Type STREAM message. To log an input/output pair,
+  // call the AddCapture* and AddAudioProcessingState methods followed
+  // by a WriteCaptureStreamMessage call.
+  virtual void AddCaptureStreamInput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamOutput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamInput(const int16_t* const data,
+                                     int num_channels,
+                                     int samples_per_channel) = 0;
+  virtual void AddCaptureStreamOutput(const int16_t* const data,
+                                      int num_channels,
+                                      int samples_per_channel) = 0;
+  virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
+  virtual void WriteCaptureStreamMessage() = 0;
+
+  // Logs Event::Type REVERSE_STREAM message.
+  virtual void WriteRenderStreamMessage(const int16_t* const data,
+                                        int num_channels,
+                                        int samples_per_channel) = 0;
+  virtual void WriteRenderStreamMessage(
+      const AudioFrameView<const float>& src) = 0;
+
+  virtual void WriteRuntimeSetting(
+      const AudioProcessing::RuntimeSetting& runtime_setting) = 0;
+
+  // Logs Event::Type CONFIG message.
+  virtual void WriteConfig(const InternalAPMConfig& config) = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_

webrtc/modules/audio_processing/include/audio_frame_proxies.cc

@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2020 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/include/audio_frame_proxies.h"
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
+  if (!frame || !ap) {
+    return AudioProcessing::Error::kNullPointerError;
+  }
+
+  StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_,
+                            /*has_keyboard=*/false);
+  StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_,
+                             /*has_keyboard=*/false);
+  RTC_DCHECK_EQ(frame->samples_per_channel(), input_config.num_frames());
+
+  int result = ap->ProcessStream(frame->data(), input_config, output_config,
+                                 frame->mutable_data());
+
+  AudioProcessingStats stats = ap->GetStatistics();
+
+  if (stats.voice_detected) {
+    frame->vad_activity_ = *stats.voice_detected
+                               ? AudioFrame::VADActivity::kVadActive
+                               : AudioFrame::VADActivity::kVadPassive;
+  }
+
+  return result;
+}
+
+int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
+  if (!frame || !ap) {
+    return AudioProcessing::Error::kNullPointerError;
+  }
+
+  // Must be a native rate.
+  if (frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate8kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate16kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate32kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate48kHz) {
+    return AudioProcessing::Error::kBadSampleRateError;
+  }
+
+  if (frame->num_channels_ <= 0) {
+    return AudioProcessing::Error::kBadNumberChannelsError;
+  }
+
+  StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_,
+                            /*has_keyboard=*/false);
+  StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_,
+                             /*has_keyboard=*/false);
+
+  int result = ap->ProcessReverseStream(frame->data(), input_config,
+                                        output_config, frame->mutable_data());
+  return result;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/include/audio_frame_proxies.h

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2020 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_INCLUDE_AUDIO_FRAME_PROXIES_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_
+
+namespace webrtc {
+
+class AudioFrame;
+class AudioProcessing;
+
+// Processes a 10 ms |frame| of the primary audio stream using the provided
+// AudioProcessing object. On the client-side, this is the near-end (or
+// captured) audio. The |sample_rate_hz_|, |num_channels_|, and
+// |samples_per_channel_| members of |frame| must be valid. If changed from the
+// previous call to this function, it will trigger an initialization of the
+// provided AudioProcessing object.
+// The function returns any error codes passed from the AudioProcessing
+// ProcessStream method.
+int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame);
+
+// Processes a 10 ms |frame| of the reverse direction audio stream using the
+// provided AudioProcessing object. The frame may be modified. On the
+// client-side, this is the far-end (or to be rendered) audio. The
+// |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| members of
+// |frame| must be valid. If changed from the previous call to this function, it
+// will trigger an initialization of the provided AudioProcessing object.
+// The function returns any error codes passed from the AudioProcessing
+// ProcessReverseStream method.
+int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_

webrtc/modules/audio_processing/include/audio_frame_view.h

@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2018 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_INCLUDE_AUDIO_FRAME_VIEW_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Class to pass audio data in T** format, where T is a numeric type.
+template <class T>
+class AudioFrameView {
+ public:
+  // |num_channels| and |channel_size| describe the T**
+  // |audio_samples|. |audio_samples| is assumed to point to a
+  // two-dimensional |num_channels * channel_size| array of floats.
+  AudioFrameView(T* const* audio_samples,
+                 size_t num_channels,
+                 size_t channel_size)
+      : audio_samples_(audio_samples),
+        num_channels_(num_channels),
+        channel_size_(channel_size) {}
+
+  // Implicit cast to allow converting Frame<float> to
+  // Frame<const float>.
+  template <class U>
+  AudioFrameView(AudioFrameView<U> other)
+      : audio_samples_(other.data()),
+        num_channels_(other.num_channels()),
+        channel_size_(other.samples_per_channel()) {}
+
+  AudioFrameView() = delete;
+
+  size_t num_channels() const { return num_channels_; }
+
+  size_t samples_per_channel() const { return channel_size_; }
+
+  rtc::ArrayView<T> channel(size_t idx) {
+    RTC_DCHECK_LE(0, idx);
+    RTC_DCHECK_LE(idx, num_channels_);
+    return rtc::ArrayView<T>(audio_samples_[idx], channel_size_);
+  }
+
+  rtc::ArrayView<const T> channel(size_t idx) const {
+    RTC_DCHECK_LE(0, idx);
+    RTC_DCHECK_LE(idx, num_channels_);
+    return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
+  }
+
+  T* const* data() { return audio_samples_; }
+
+ private:
+  T* const* audio_samples_;
+  size_t num_channels_;
+  size_t channel_size_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_

webrtc/modules/audio_processing/include/audio_processing.cc

@@ -0,0 +1,124 @@
+/*
+ *  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/include/audio_processing.h"
+
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace {
+
+std::string NoiseSuppressionLevelToString(
+    const AudioProcessing::Config::NoiseSuppression::Level& level) {
+  switch (level) {
+    case AudioProcessing::Config::NoiseSuppression::Level::kLow:
+      return "Low";
+    case AudioProcessing::Config::NoiseSuppression::Level::kModerate:
+      return "Moderate";
+    case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
+      return "High";
+    case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
+      return "VeryHigh";
+  }
+}
+
+std::string GainController1ModeToString(
+    const AudioProcessing::Config::GainController1::Mode& mode) {
+  switch (mode) {
+    case AudioProcessing::Config::GainController1::Mode::kAdaptiveAnalog:
+      return "AdaptiveAnalog";
+    case AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital:
+      return "AdaptiveDigital";
+    case AudioProcessing::Config::GainController1::Mode::kFixedDigital:
+      return "FixedDigital";
+  }
+}
+
+std::string GainController2LevelEstimatorToString(
+    const AudioProcessing::Config::GainController2::LevelEstimator& level) {
+  switch (level) {
+    case AudioProcessing::Config::GainController2::LevelEstimator::kRms:
+      return "Rms";
+    case AudioProcessing::Config::GainController2::LevelEstimator::kPeak:
+      return "Peak";
+  }
+}
+
+int GetDefaultMaxInternalRate() {
+#ifdef WEBRTC_ARCH_ARM_FAMILY
+  return 32000;
+#else
+  return 48000;
+#endif
+}
+
+}  // namespace
+
+constexpr int AudioProcessing::kNativeSampleRatesHz[];
+
+void CustomProcessing::SetRuntimeSetting(
+    AudioProcessing::RuntimeSetting setting) {}
+
+AudioProcessing::Config::Pipeline::Pipeline()
+    : maximum_internal_processing_rate(GetDefaultMaxInternalRate()) {}
+
+std::string AudioProcessing::Config::ToString() const {
+  char buf[1024];
+  rtc::SimpleStringBuilder builder(buf);
+  builder << "AudioProcessing::Config{ "
+             "pipeline: {"
+             "maximum_internal_processing_rate: "
+          << pipeline.maximum_internal_processing_rate
+          << ", multi_channel_render: " << pipeline.multi_channel_render
+          << ", "
+             ", multi_channel_capture: "
+          << pipeline.multi_channel_capture
+          << "}, "
+             "pre_amplifier: { enabled: "
+          << pre_amplifier.enabled
+          << ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
+          << " }, high_pass_filter: { enabled: " << high_pass_filter.enabled
+          << " }, echo_canceller: { enabled: " << echo_canceller.enabled
+          << ", mobile_mode: " << echo_canceller.mobile_mode
+          << ", enforce_high_pass_filtering: "
+          << echo_canceller.enforce_high_pass_filtering
+          << " }, noise_suppression: { enabled: " << noise_suppression.enabled
+          << ", level: "
+          << NoiseSuppressionLevelToString(noise_suppression.level)
+          << " }, transient_suppression: { enabled: "
+          << transient_suppression.enabled
+          << " }, voice_detection: { enabled: " << voice_detection.enabled
+          << " }, gain_controller1: { enabled: " << gain_controller1.enabled
+          << ", mode: " << GainController1ModeToString(gain_controller1.mode)
+          << ", target_level_dbfs: " << gain_controller1.target_level_dbfs
+          << ", compression_gain_db: " << gain_controller1.compression_gain_db
+          << ", enable_limiter: " << gain_controller1.enable_limiter
+          << ", analog_level_minimum: " << gain_controller1.analog_level_minimum
+          << ", analog_level_maximum: " << gain_controller1.analog_level_maximum
+          << " }, gain_controller2: { enabled: " << gain_controller2.enabled
+          << ", fixed_digital: { gain_db: "
+          << gain_controller2.fixed_digital.gain_db
+          << " }, adaptive_digital: { enabled: "
+          << gain_controller2.adaptive_digital.enabled << ", level_estimator: "
+          << GainController2LevelEstimatorToString(
+                 gain_controller2.adaptive_digital.level_estimator)
+          << ", use_saturation_protector: "
+          << gain_controller2.adaptive_digital.use_saturation_protector
+          << ", extra_saturation_margin_db: "
+          << gain_controller2.adaptive_digital.extra_saturation_margin_db
+          << " } }, residual_echo_detector: { enabled: "
+          << residual_echo_detector.enabled
+          << " }, level_estimation: { enabled: " << level_estimation.enabled
+          << " } }";
+  return builder.str();
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/include/audio_processing_statistics.cc

@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2017 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/include/audio_processing_statistics.h"
+
+namespace webrtc {
+
+AudioProcessingStats::AudioProcessingStats() = default;
+
+AudioProcessingStats::AudioProcessingStats(const AudioProcessingStats& other) =
+    default;
+
+AudioProcessingStats::~AudioProcessingStats() = default;
+
+}  // namespace webrtc

webrtc/modules/audio_processing/include/audio_processing_statistics.h

@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2017 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_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+// This version of the stats uses Optionals, it will replace the regular
+// AudioProcessingStatistics struct.
+struct RTC_EXPORT AudioProcessingStats {
+  AudioProcessingStats();
+  AudioProcessingStats(const AudioProcessingStats& other);
+  ~AudioProcessingStats();
+
+  // The root mean square (RMS) level in dBFS (decibels from digital
+  // full-scale) of the last capture frame, after processing. It is
+  // constrained to [-127, 0].
+  // The computation follows: https://tools.ietf.org/html/rfc6465
+  // with the intent that it can provide the RTP audio level indication.
+  // Only reported if level estimation is enabled in AudioProcessing::Config.
+  absl::optional<int> output_rms_dbfs;
+
+  // True if voice is detected in the last capture frame, after processing.
+  // It is conservative in flagging audio as speech, with low likelihood of
+  // incorrectly flagging a frame as voice.
+  // Only reported if voice detection is enabled in AudioProcessing::Config.
+  absl::optional<bool> voice_detected;
+
+  // AEC Statistics.
+  // ERL = 10log_10(P_far / P_echo)
+  absl::optional<double> echo_return_loss;
+  // ERLE = 10log_10(P_echo / P_out)
+  absl::optional<double> echo_return_loss_enhancement;
+  // Fraction of time that the AEC linear filter is divergent, in a 1-second
+  // non-overlapped aggregation window.
+  absl::optional<double> divergent_filter_fraction;
+
+  // The delay metrics consists of the delay median and standard deviation. It
+  // also consists of the fraction of delay estimates that can make the echo
+  // cancellation perform poorly. The values are aggregated until the first
+  // call to |GetStatistics()| and afterwards aggregated and updated every
+  // second. Note that if there are several clients pulling metrics from
+  // |GetStatistics()| during a session the first call from any of them will
+  // change to one second aggregation window for all.
+  absl::optional<int32_t> delay_median_ms;
+  absl::optional<int32_t> delay_standard_deviation_ms;
+
+  // Residual echo detector likelihood.
+  absl::optional<double> residual_echo_likelihood;
+  // Maximum residual echo likelihood from the last time period.
+  absl::optional<double> residual_echo_likelihood_recent_max;
+
+  // The instantaneous delay estimate produced in the AEC. The unit is in
+  // milliseconds and the value is the instantaneous value at the time of the
+  // call to |GetStatistics()|.
+  absl::optional<int32_t> delay_ms;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_

webrtc/modules/audio_processing/include/config.cc

@@ -0,0 +1,23 @@
+/*
+ *  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/include/config.h"
+
+namespace webrtc {
+
+Config::Config() {}
+
+Config::~Config() {
+  for (OptionMap::iterator it = options_.begin(); it != options_.end(); ++it) {
+    delete it->second;
+  }
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/include/config.h

@@ -0,0 +1,131 @@
+/*
+ *  Copyright (c) 2013 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_INCLUDE_CONFIG_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_CONFIG_H_
+
+#include <map>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Only add new values to the end of the enumeration and never remove (only
+// deprecate) to maintain binary compatibility.
+enum class ConfigOptionID {
+  kMyExperimentForTest,
+  kAlgo1CostFunctionForTest,
+  kTemporalLayersFactory,  // Deprecated
+  kNetEqCapacityConfig,    // Deprecated
+  kNetEqFastAccelerate,    // Deprecated
+  kVoicePacing,            // Deprecated
+  kExtendedFilter,         // Deprecated
+  kDelayAgnostic,          // Deprecated
+  kExperimentalAgc,
+  kExperimentalNs,
+  kBeamforming,               // Deprecated
+  kIntelligibility,           // Deprecated
+  kEchoCanceller3,            // Deprecated
+  kAecRefinedAdaptiveFilter,  // Deprecated
+  kLevelControl               // Deprecated
+};
+
+// Class Config is designed to ease passing a set of options across webrtc code.
+// Options are identified by typename in order to avoid incorrect casts.
+//
+// Usage:
+// * declaring an option:
+//    struct Algo1_CostFunction {
+//      virtual float cost(int x) const { return x; }
+//      virtual ~Algo1_CostFunction() {}
+//    };
+//
+// * accessing an option:
+//    config.Get<Algo1_CostFunction>().cost(value);
+//
+// * setting an option:
+//    struct SqrCost : Algo1_CostFunction {
+//      virtual float cost(int x) const { return x*x; }
+//    };
+//    config.Set<Algo1_CostFunction>(new SqrCost());
+//
+// Note: This class is thread-compatible (like STL containers).
+class RTC_EXPORT Config {
+ public:
+  // Returns the option if set or a default constructed one.
+  // Callers that access options too often are encouraged to cache the result.
+  // Returned references are owned by this.
+  //
+  // Requires std::is_default_constructible<T>
+  template <typename T>
+  const T& Get() const;
+
+  // Set the option, deleting any previous instance of the same.
+  // This instance gets ownership of the newly set value.
+  template <typename T>
+  void Set(T* value);
+
+  Config();
+  ~Config();
+
+ private:
+  struct BaseOption {
+    virtual ~BaseOption() {}
+  };
+
+  template <typename T>
+  struct Option : BaseOption {
+    explicit Option(T* v) : value(v) {}
+    ~Option() { delete value; }
+    T* value;
+  };
+
+  template <typename T>
+  static ConfigOptionID identifier() {
+    return T::identifier;
+  }
+
+  // Used to instantiate a default constructed object that doesn't needs to be
+  // owned. This allows Get<T> to be implemented without requiring explicitly
+  // locks.
+  template <typename T>
+  static const T& default_value() {
+    static const T* const def = new T();
+    return *def;
+  }
+
+  typedef std::map<ConfigOptionID, BaseOption*> OptionMap;
+  OptionMap options_;
+
+  Config(const Config&);
+  void operator=(const Config&);
+};
+
+template <typename T>
+const T& Config::Get() const {
+  OptionMap::const_iterator it = options_.find(identifier<T>());
+  if (it != options_.end()) {
+    const T* t = static_cast<Option<T>*>(it->second)->value;
+    if (t) {
+      return *t;
+    }
+  }
+  return default_value<T>();
+}
+
+template <typename T>
+void Config::Set(T* value) {
+  BaseOption*& it = options_[identifier<T>()];
+  delete it;
+  it = new Option<T>(value);
+}
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_CONFIG_H_