@recallai/desktop-sdk 2024.10.16-348bb0a1cdf9c002fa25ef93373a09667bd4e463 → 2024.10.16-453fae05e0e4b90869a62e06a4c266ed53f681b6

package/desktop_sdk_macos.framework/Versions/A/Frameworks/GStreamer.framework/Versions/1.0/include/webrtc-audio-processing-1/modules/audio_processing/include/audio_processing.h

@@ -1,924 +0,0 @@
-/*
- *  Copyright (c) 2012 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_H_
-#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
-
-// MSVC++ requires this to be set before any other includes to get M_PI.
-#ifndef _USE_MATH_DEFINES
-#define _USE_MATH_DEFINES
-#endif
-
-#include <math.h>
-#include <stddef.h>  // size_t
-#include <stdio.h>   // FILE
-#include <string.h>
-
-#include <vector>
-
-#include "absl/types/optional.h"
-#include "api/array_view.h"
-#include "api/audio/echo_canceller3_config.h"
-#include "api/audio/echo_control.h"
-#include "api/scoped_refptr.h"
-#include "modules/audio_processing/include/audio_processing_statistics.h"
-#include "modules/audio_processing/include/config.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/constructor_magic.h"
-#include "rtc_base/deprecation.h"
-#include "rtc_base/ref_count.h"
-#include "rtc_base/system/file_wrapper.h"
-#include "rtc_base/system/rtc_export.h"
-
-namespace rtc {
-class TaskQueue;
-}  // namespace rtc
-
-namespace webrtc {
-
-class AecDump;
-class AudioBuffer;
-
-class StreamConfig;
-class ProcessingConfig;
-
-class EchoDetector;
-class CustomAudioAnalyzer;
-class CustomProcessing;
-
-// Use to enable experimental gain control (AGC). At startup the experimental
-// AGC moves the microphone volume up to |startup_min_volume| if the current
-// microphone volume is set too low. The value is clamped to its operating range
-// [12, 255]. Here, 255 maps to 100%.
-//
-// Must be provided through AudioProcessingBuilder().Create(config).
-#if defined(WEBRTC_CHROMIUM_BUILD)
-static const int kAgcStartupMinVolume = 85;
-#else
-static const int kAgcStartupMinVolume = 0;
-#endif  // defined(WEBRTC_CHROMIUM_BUILD)
-static constexpr int kClippedLevelMin = 70;
-
-// To be deprecated: Please instead use the flag in the
-// AudioProcessing::Config::AnalogGainController.
-// TODO(webrtc:5298): Remove.
-struct ExperimentalAgc {
-  ExperimentalAgc() = default;
-  explicit ExperimentalAgc(bool enabled) : enabled(enabled) {}
-  ExperimentalAgc(bool enabled,
-                  bool enabled_agc2_level_estimator,
-                  bool digital_adaptive_disabled)
-      : enabled(enabled),
-        enabled_agc2_level_estimator(enabled_agc2_level_estimator),
-        digital_adaptive_disabled(digital_adaptive_disabled) {}
-  // Deprecated constructor: will be removed.
-  ExperimentalAgc(bool enabled,
-                  bool enabled_agc2_level_estimator,
-                  bool digital_adaptive_disabled,
-                  bool analyze_before_aec)
-      : enabled(enabled),
-        enabled_agc2_level_estimator(enabled_agc2_level_estimator),
-        digital_adaptive_disabled(digital_adaptive_disabled) {}
-  ExperimentalAgc(bool enabled, int startup_min_volume)
-      : enabled(enabled), startup_min_volume(startup_min_volume) {}
-  ExperimentalAgc(bool enabled, int startup_min_volume, int clipped_level_min)
-      : enabled(enabled),
-        startup_min_volume(startup_min_volume),
-        clipped_level_min(clipped_level_min) {}
-  static const ConfigOptionID identifier = ConfigOptionID::kExperimentalAgc;
-  bool enabled = true;
-  int startup_min_volume = kAgcStartupMinVolume;
-  // Lowest microphone level that will be applied in response to clipping.
-  int clipped_level_min = kClippedLevelMin;
-  bool enabled_agc2_level_estimator = false;
-  bool digital_adaptive_disabled = false;
-};
-
-// To be deprecated: Please instead use the flag in the
-// AudioProcessing::Config::TransientSuppression.
-//
-// Use to enable experimental noise suppression. It can be set in the
-// constructor.
-// TODO(webrtc:5298): Remove.
-struct ExperimentalNs {
-  ExperimentalNs() : enabled(false) {}
-  explicit ExperimentalNs(bool enabled) : enabled(enabled) {}
-  static const ConfigOptionID identifier = ConfigOptionID::kExperimentalNs;
-  bool enabled;
-};
-
-// The Audio Processing Module (APM) provides a collection of voice processing
-// components designed for real-time communications software.
-//
-// APM operates on two audio streams on a frame-by-frame basis. Frames of the
-// primary stream, on which all processing is applied, are passed to
-// |ProcessStream()|. Frames of the reverse direction stream are passed to
-// |ProcessReverseStream()|. On the client-side, this will typically be the
-// near-end (capture) and far-end (render) streams, respectively. APM should be
-// placed in the signal chain as close to the audio hardware abstraction layer
-// (HAL) as possible.
-//
-// On the server-side, the reverse stream will normally not be used, with
-// processing occurring on each incoming stream.
-//
-// Component interfaces follow a similar pattern and are accessed through
-// corresponding getters in APM. All components are disabled at create-time,
-// with default settings that are recommended for most situations. New settings
-// can be applied without enabling a component. Enabling a component triggers
-// memory allocation and initialization to allow it to start processing the
-// streams.
-//
-// Thread safety is provided with the following assumptions to reduce locking
-// overhead:
-//   1. The stream getters and setters are called from the same thread as
-//      ProcessStream(). More precisely, stream functions are never called
-//      concurrently with ProcessStream().
-//   2. Parameter getters are never called concurrently with the corresponding
-//      setter.
-//
-// APM accepts only linear PCM audio data in chunks of 10 ms. The int16
-// interfaces use interleaved data, while the float interfaces use deinterleaved
-// data.
-//
-// Usage example, omitting error checking:
-// AudioProcessing* apm = AudioProcessingBuilder().Create();
-//
-// AudioProcessing::Config config;
-// config.echo_canceller.enabled = true;
-// config.echo_canceller.mobile_mode = false;
-//
-// config.gain_controller1.enabled = true;
-// config.gain_controller1.mode =
-// AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-// config.gain_controller1.analog_level_minimum = 0;
-// config.gain_controller1.analog_level_maximum = 255;
-//
-// config.gain_controller2.enabled = true;
-//
-// config.high_pass_filter.enabled = true;
-//
-// config.voice_detection.enabled = true;
-//
-// apm->ApplyConfig(config)
-//
-// apm->noise_reduction()->set_level(kHighSuppression);
-// apm->noise_reduction()->Enable(true);
-//
-// // Start a voice call...
-//
-// // ... Render frame arrives bound for the audio HAL ...
-// apm->ProcessReverseStream(render_frame);
-//
-// // ... Capture frame arrives from the audio HAL ...
-// // Call required set_stream_ functions.
-// apm->set_stream_delay_ms(delay_ms);
-// apm->set_stream_analog_level(analog_level);
-//
-// apm->ProcessStream(capture_frame);
-//
-// // Call required stream_ functions.
-// analog_level = apm->recommended_stream_analog_level();
-// has_voice = apm->stream_has_voice();
-//
-// // Repeate render and capture processing for the duration of the call...
-// // Start a new call...
-// apm->Initialize();
-//
-// // Close the application...
-// delete apm;
-//
-class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
- public:
-  // The struct below constitutes the new parameter scheme for the audio
-  // processing. It is being introduced gradually and until it is fully
-  // introduced, it is prone to change.
-  // TODO(peah): Remove this comment once the new config scheme is fully rolled
-  // out.
-  //
-  // The parameters and behavior of the audio processing module are controlled
-  // by changing the default values in the AudioProcessing::Config struct.
-  // The config is applied by passing the struct to the ApplyConfig method.
-  //
-  // This config is intended to be used during setup, and to enable/disable
-  // top-level processing effects. Use during processing may cause undesired
-  // submodule resets, affecting the audio quality. Use the RuntimeSetting
-  // construct for runtime configuration.
-  struct RTC_EXPORT Config {
-
-    // Sets the properties of the audio processing pipeline.
-    struct RTC_EXPORT Pipeline {
-      Pipeline();
-
-      // Maximum allowed processing rate used internally. May only be set to
-      // 32000 or 48000 and any differing values will be treated as 48000. The
-      // default rate is currently selected based on the CPU architecture, but
-      // that logic may change.
-      int maximum_internal_processing_rate;
-      // Allow multi-channel processing of render audio.
-      bool multi_channel_render = false;
-      // Allow multi-channel processing of capture audio when AEC3 is active
-      // or a custom AEC is injected..
-      bool multi_channel_capture = false;
-    } pipeline;
-
-    // Enabled the pre-amplifier. It amplifies the capture signal
-    // before any other processing is done.
-    struct PreAmplifier {
-      bool enabled = false;
-      float fixed_gain_factor = 1.f;
-    } pre_amplifier;
-
-    struct HighPassFilter {
-      bool enabled = false;
-      bool apply_in_full_band = true;
-    } high_pass_filter;
-
-    struct EchoCanceller {
-      bool enabled = false;
-      bool mobile_mode = false;
-      bool export_linear_aec_output = false;
-      // Enforce the highpass filter to be on (has no effect for the mobile
-      // mode).
-      bool enforce_high_pass_filtering = true;
-    } echo_canceller;
-
-    // Enables background noise suppression.
-    struct NoiseSuppression {
-      bool enabled = false;
-      enum Level { kLow, kModerate, kHigh, kVeryHigh };
-      Level level = kModerate;
-      bool analyze_linear_aec_output_when_available = false;
-    } noise_suppression;
-
-    // Enables transient suppression.
-    struct TransientSuppression {
-      bool enabled = false;
-    } transient_suppression;
-
-    // Enables reporting of |voice_detected| in webrtc::AudioProcessingStats.
-    struct VoiceDetection {
-      bool enabled = false;
-    } voice_detection;
-
-    // Enables automatic gain control (AGC) functionality.
-    // The automatic gain control (AGC) component brings the signal to an
-    // appropriate range. This is done by applying a digital gain directly and,
-    // in the analog mode, prescribing an analog gain to be applied at the audio
-    // HAL.
-    // Recommended to be enabled on the client-side.
-    struct GainController1 {
-      bool enabled = false;
-      enum Mode {
-        // Adaptive mode intended for use if an analog volume control is
-        // available on the capture device. It will require the user to provide
-        // coupling between the OS mixer controls and AGC through the
-        // stream_analog_level() functions.
-        // It consists of an analog gain prescription for the audio device and a
-        // digital compression stage.
-        kAdaptiveAnalog,
-        // Adaptive mode intended for situations in which an analog volume
-        // control is unavailable. It operates in a similar fashion to the
-        // adaptive analog mode, but with scaling instead applied in the digital
-        // domain. As with the analog mode, it additionally uses a digital
-        // compression stage.
-        kAdaptiveDigital,
-        // Fixed mode which enables only the digital compression stage also used
-        // by the two adaptive modes.
-        // It is distinguished from the adaptive modes by considering only a
-        // short time-window of the input signal. It applies a fixed gain
-        // through most of the input level range, and compresses (gradually
-        // reduces gain with increasing level) the input signal at higher
-        // levels. This mode is preferred on embedded devices where the capture
-        // signal level is predictable, so that a known gain can be applied.
-        kFixedDigital
-      };
-      Mode mode = kAdaptiveAnalog;
-      // Sets the target peak level (or envelope) of the AGC in dBFs (decibels
-      // from digital full-scale). The convention is to use positive values. For
-      // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
-      // level 3 dB below full-scale. Limited to [0, 31].
-      int target_level_dbfs = 3;
-      // Sets the maximum gain the digital compression stage may apply, in dB. A
-      // higher number corresponds to greater compression, while a value of 0
-      // will leave the signal uncompressed. Limited to [0, 90].
-      // For updates after APM setup, use a RuntimeSetting instead.
-      int compression_gain_db = 9;
-      // When enabled, the compression stage will hard limit the signal to the
-      // target level. Otherwise, the signal will be compressed but not limited
-      // above the target level.
-      bool enable_limiter = true;
-      // Sets the minimum and maximum analog levels of the audio capture device.
-      // Must be set if an analog mode is used. Limited to [0, 65535].
-      int analog_level_minimum = 0;
-      int analog_level_maximum = 255;
-
-      // Enables the analog gain controller functionality.
-      struct AnalogGainController {
-        bool enabled = true;
-        int startup_min_volume = kAgcStartupMinVolume;
-        // Lowest analog microphone level that will be applied in response to
-        // clipping.
-        int clipped_level_min = kClippedLevelMin;
-        bool enable_agc2_level_estimator = false;
-        bool enable_digital_adaptive = true;
-      } analog_gain_controller;
-    } gain_controller1;
-
-    // Enables the next generation AGC functionality. This feature replaces the
-    // standard methods of gain control in the previous AGC. Enabling this
-    // submodule enables an adaptive digital AGC followed by a limiter. By
-    // setting |fixed_gain_db|, the limiter can be turned into a compressor that
-    // first applies a fixed gain. The adaptive digital AGC can be turned off by
-    // setting |adaptive_digital_mode=false|.
-    struct GainController2 {
-      enum LevelEstimator { kRms, kPeak };
-      bool enabled = false;
-      struct {
-        float gain_db = 0.f;
-      } fixed_digital;
-      struct {
-        bool enabled = false;
-        float vad_probability_attack = 1.f;
-        LevelEstimator level_estimator = kRms;
-        int level_estimator_adjacent_speech_frames_threshold = 1;
-        // TODO(crbug.com/webrtc/7494): Remove `use_saturation_protector`.
-        bool use_saturation_protector = true;
-        float initial_saturation_margin_db = 20.f;
-        float extra_saturation_margin_db = 2.f;
-        int gain_applier_adjacent_speech_frames_threshold = 1;
-        float max_gain_change_db_per_second = 3.f;
-        float max_output_noise_level_dbfs = -50.f;
-      } adaptive_digital;
-    } gain_controller2;
-
-    struct ResidualEchoDetector {
-      bool enabled = true;
-    } residual_echo_detector;
-
-    // Enables reporting of |output_rms_dbfs| in webrtc::AudioProcessingStats.
-    struct LevelEstimation {
-      bool enabled = false;
-    } level_estimation;
-
-    std::string ToString() const;
-  };
-
-  // TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
-  enum ChannelLayout {
-    kMono,
-    // Left, right.
-    kStereo,
-    // Mono, keyboard, and mic.
-    kMonoAndKeyboard,
-    // Left, right, keyboard, and mic.
-    kStereoAndKeyboard
-  };
-
-  // Specifies the properties of a setting to be passed to AudioProcessing at
-  // runtime.
-  class RuntimeSetting {
-   public:
-    enum class Type {
-      kNotSpecified,
-      kCapturePreGain,
-      kCaptureCompressionGain,
-      kCaptureFixedPostGain,
-      kPlayoutVolumeChange,
-      kCustomRenderProcessingRuntimeSetting,
-      kPlayoutAudioDeviceChange,
-      kCaptureOutputUsed
-    };
-
-    // Play-out audio device properties.
-    struct PlayoutAudioDeviceInfo {
-      int id;          // Identifies the audio device.
-      int max_volume;  // Maximum play-out volume.
-    };
-
-    RuntimeSetting() : type_(Type::kNotSpecified), value_(0.f) {}
-    ~RuntimeSetting() = default;
-
-    static RuntimeSetting CreateCapturePreGain(float gain) {
-      RTC_DCHECK_GE(gain, 1.f) << "Attenuation is not allowed.";
-      return {Type::kCapturePreGain, gain};
-    }
-
-    // Corresponds to Config::GainController1::compression_gain_db, but for
-    // runtime configuration.
-    static RuntimeSetting CreateCompressionGainDb(int gain_db) {
-      RTC_DCHECK_GE(gain_db, 0);
-      RTC_DCHECK_LE(gain_db, 90);
-      return {Type::kCaptureCompressionGain, static_cast<float>(gain_db)};
-    }
-
-    // Corresponds to Config::GainController2::fixed_digital::gain_db, but for
-    // runtime configuration.
-    static RuntimeSetting CreateCaptureFixedPostGain(float gain_db) {
-      RTC_DCHECK_GE(gain_db, 0.f);
-      RTC_DCHECK_LE(gain_db, 90.f);
-      return {Type::kCaptureFixedPostGain, gain_db};
-    }
-
-    // Creates a runtime setting to notify play-out (aka render) audio device
-    // changes.
-    static RuntimeSetting CreatePlayoutAudioDeviceChange(
-        PlayoutAudioDeviceInfo audio_device) {
-      return {Type::kPlayoutAudioDeviceChange, audio_device};
-    }
-
-    // Creates a runtime setting to notify play-out (aka render) volume changes.
-    // |volume| is the unnormalized volume, the maximum of which
-    static RuntimeSetting CreatePlayoutVolumeChange(int volume) {
-      return {Type::kPlayoutVolumeChange, volume};
-    }
-
-    static RuntimeSetting CreateCustomRenderSetting(float payload) {
-      return {Type::kCustomRenderProcessingRuntimeSetting, payload};
-    }
-
-    static RuntimeSetting CreateCaptureOutputUsedSetting(bool payload) {
-      return {Type::kCaptureOutputUsed, payload};
-    }
-
-    Type type() const { return type_; }
-    // Getters do not return a value but instead modify the argument to protect
-    // from implicit casting.
-    void GetFloat(float* value) const {
-      RTC_DCHECK(value);
-      *value = value_.float_value;
-    }
-    void GetInt(int* value) const {
-      RTC_DCHECK(value);
-      *value = value_.int_value;
-    }
-    void GetBool(bool* value) const {
-      RTC_DCHECK(value);
-      *value = value_.bool_value;
-    }
-    void GetPlayoutAudioDeviceInfo(PlayoutAudioDeviceInfo* value) const {
-      RTC_DCHECK(value);
-      *value = value_.playout_audio_device_info;
-    }
-
-   private:
-    RuntimeSetting(Type id, float value) : type_(id), value_(value) {}
-    RuntimeSetting(Type id, int value) : type_(id), value_(value) {}
-    RuntimeSetting(Type id, PlayoutAudioDeviceInfo value)
-        : type_(id), value_(value) {}
-    Type type_;
-    union U {
-      U() {}
-      U(int value) : int_value(value) {}
-      U(float value) : float_value(value) {}
-      U(PlayoutAudioDeviceInfo value) : playout_audio_device_info(value) {}
-      float float_value;
-      int int_value;
-      bool bool_value;
-      PlayoutAudioDeviceInfo playout_audio_device_info;
-    } value_;
-  };
-
-  ~AudioProcessing() override {}
-
-  // Initializes internal states, while retaining all user settings. This
-  // should be called before beginning to process a new audio stream. However,
-  // it is not necessary to call before processing the first stream after
-  // creation.
-  //
-  // It is also not necessary to call if the audio parameters (sample
-  // rate and number of channels) have changed. Passing updated parameters
-  // directly to |ProcessStream()| and |ProcessReverseStream()| is permissible.
-  // If the parameters are known at init-time though, they may be provided.
-  // TODO(webrtc:5298): Change to return void.
-  virtual int Initialize() = 0;
-
-  // The int16 interfaces require:
-  //   - only |NativeRate|s be used
-  //   - that the input, output and reverse rates must match
-  //   - that |processing_config.output_stream()| matches
-  //     |processing_config.input_stream()|.
-  //
-  // The float interfaces accept arbitrary rates and support differing input and
-  // output layouts, but the output must have either one channel or the same
-  // number of channels as the input.
-  virtual int Initialize(const ProcessingConfig& processing_config) = 0;
-
-  // Initialize with unpacked parameters. See Initialize() above for details.
-  //
-  // TODO(mgraczyk): Remove once clients are updated to use the new interface.
-  virtual int Initialize(int capture_input_sample_rate_hz,
-                         int capture_output_sample_rate_hz,
-                         int render_sample_rate_hz,
-                         ChannelLayout capture_input_layout,
-                         ChannelLayout capture_output_layout,
-                         ChannelLayout render_input_layout) = 0;
-
-  // TODO(peah): This method is a temporary solution used to take control
-  // over the parameters in the audio processing module and is likely to change.
-  virtual void ApplyConfig(const Config& config) = 0;
-
-  // TODO(ajm): Only intended for internal use. Make private and friend the
-  // necessary classes?
-  virtual int proc_sample_rate_hz() const = 0;
-  virtual int proc_split_sample_rate_hz() const = 0;
-  virtual size_t num_input_channels() const = 0;
-  virtual size_t num_proc_channels() const = 0;
-  virtual size_t num_output_channels() const = 0;
-  virtual size_t num_reverse_channels() const = 0;
-
-  // Set to true when the output of AudioProcessing will be muted or in some
-  // other way not used. Ideally, the captured audio would still be processed,
-  // but some components may change behavior based on this information.
-  // Default false.
-  virtual void set_output_will_be_muted(bool muted) = 0;
-
-  // Enqueue a runtime setting.
-  virtual void SetRuntimeSetting(RuntimeSetting setting) = 0;
-
-  // Accepts and produces a 10 ms frame interleaved 16 bit integer audio as
-  // specified in |input_config| and |output_config|. |src| and |dest| may use
-  // the same memory, if desired.
-  virtual int ProcessStream(const int16_t* const src,
-                            const StreamConfig& input_config,
-                            const StreamConfig& output_config,
-                            int16_t* const dest) = 0;
-
-  // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
-  // |src| points to a channel buffer, arranged according to |input_stream|. At
-  // output, the channels will be arranged according to |output_stream| in
-  // |dest|.
-  //
-  // The output must have one channel or as many channels as the input. |src|
-  // and |dest| may use the same memory, if desired.
-  virtual int ProcessStream(const float* const* src,
-                            const StreamConfig& input_config,
-                            const StreamConfig& output_config,
-                            float* const* dest) = 0;
-
-  // Accepts and produces a 10 ms frame of interleaved 16 bit integer audio for
-  // the reverse direction audio stream as specified in |input_config| and
-  // |output_config|. |src| and |dest| may use the same memory, if desired.
-  virtual int ProcessReverseStream(const int16_t* const src,
-                                   const StreamConfig& input_config,
-                                   const StreamConfig& output_config,
-                                   int16_t* const dest) = 0;
-
-  // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
-  // |data| points to a channel buffer, arranged according to |reverse_config|.
-  virtual int ProcessReverseStream(const float* const* src,
-                                   const StreamConfig& input_config,
-                                   const StreamConfig& output_config,
-                                   float* const* dest) = 0;
-
-  // Accepts deinterleaved float audio with the range [-1, 1]. Each element
-  // of |data| points to a channel buffer, arranged according to
-  // |reverse_config|.
-  virtual int AnalyzeReverseStream(const float* const* data,
-                                   const StreamConfig& reverse_config) = 0;
-
-  // Returns the most recently produced 10 ms of the linear AEC output at a rate
-  // of 16 kHz. If there is more than one capture channel, a mono representation
-  // of the input is returned. Returns true/false to indicate whether an output
-  // returned.
-  virtual bool GetLinearAecOutput(
-      rtc::ArrayView<std::array<float, 160>> linear_output) const = 0;
-
-  // This must be called prior to ProcessStream() if and only if adaptive analog
-  // gain control is enabled, to pass the current analog level from the audio
-  // HAL. Must be within the range provided in Config::GainController1.
-  virtual void set_stream_analog_level(int level) = 0;
-
-  // When an analog mode is set, this should be called after ProcessStream()
-  // to obtain the recommended new analog level for the audio HAL. It is the
-  // user's responsibility to apply this level.
-  virtual int recommended_stream_analog_level() const = 0;
-
-  // This must be called if and only if echo processing is enabled.
-  //
-  // Sets the |delay| in ms between ProcessReverseStream() receiving a far-end
-  // frame and ProcessStream() receiving a near-end frame containing the
-  // corresponding echo. On the client-side this can be expressed as
-  //   delay = (t_render - t_analyze) + (t_process - t_capture)
-  // where,
-  //   - t_analyze is the time a frame is passed to ProcessReverseStream() and
-  //     t_render is the time the first sample of the same frame is rendered by
-  //     the audio hardware.
-  //   - t_capture is the time the first sample of a frame is captured by the
-  //     audio hardware and t_process is the time the same frame is passed to
-  //     ProcessStream().
-  virtual int set_stream_delay_ms(int delay) = 0;
-  virtual int stream_delay_ms() const = 0;
-
-  // Call to signal that a key press occurred (true) or did not occur (false)
-  // with this chunk of audio.
-  virtual void set_stream_key_pressed(bool key_pressed) = 0;
-
-  // Creates and attaches an webrtc::AecDump for recording debugging
-  // information.
-  // The |worker_queue| may not be null and must outlive the created
-  // AecDump instance. |max_log_size_bytes == -1| means the log size
-  // will be unlimited. |handle| may not be null. The AecDump takes
-  // responsibility for |handle| and closes it in the destructor. A
-  // return value of true indicates that the file has been
-  // sucessfully opened, while a value of false indicates that
-  // opening the file failed.
-  virtual bool CreateAndAttachAecDump(const std::string& file_name,
-                                      int64_t max_log_size_bytes,
-                                      rtc::TaskQueue* worker_queue) = 0;
-  virtual bool CreateAndAttachAecDump(FILE* handle,
-                                      int64_t max_log_size_bytes,
-                                      rtc::TaskQueue* worker_queue) = 0;
-
-  // TODO(webrtc:5298) Deprecated variant.
-  // Attaches provided webrtc::AecDump for recording debugging
-  // information. Log file and maximum file size logic is supposed to
-  // be handled by implementing instance of AecDump. Calling this
-  // method when another AecDump is attached resets the active AecDump
-  // with a new one. This causes the d-tor of the earlier AecDump to
-  // be called. The d-tor call may block until all pending logging
-  // tasks are completed.
-  virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) = 0;
-
-  // If no AecDump is attached, this has no effect. If an AecDump is
-  // attached, it's destructor is called. The d-tor may block until
-  // all pending logging tasks are completed.
-  virtual void DetachAecDump() = 0;
-
-  // Get audio processing statistics.
-  virtual AudioProcessingStats GetStatistics() = 0;
-  // TODO(webrtc:5298) Deprecated variant. The |has_remote_tracks| argument
-  // should be set if there are active remote tracks (this would usually be true
-  // during a call). If there are no remote tracks some of the stats will not be
-  // set by AudioProcessing, because they only make sense if there is at least
-  // one remote track.
-  virtual AudioProcessingStats GetStatistics(bool has_remote_tracks) = 0;
-
-  // Returns the last applied configuration.
-  virtual AudioProcessing::Config GetConfig() const = 0;
-
-  enum Error {
-    // Fatal errors.
-    kNoError = 0,
-    kUnspecifiedError = -1,
-    kCreationFailedError = -2,
-    kUnsupportedComponentError = -3,
-    kUnsupportedFunctionError = -4,
-    kNullPointerError = -5,
-    kBadParameterError = -6,
-    kBadSampleRateError = -7,
-    kBadDataLengthError = -8,
-    kBadNumberChannelsError = -9,
-    kFileError = -10,
-    kStreamParameterNotSetError = -11,
-    kNotEnabledError = -12,
-
-    // Warnings are non-fatal.
-    // This results when a set_stream_ parameter is out of range. Processing
-    // will continue, but the parameter may have been truncated.
-    kBadStreamParameterWarning = -13
-  };
-
-  // Native rates supported by the integer interfaces.
-  enum NativeRate {
-    kSampleRate8kHz = 8000,
-    kSampleRate16kHz = 16000,
-    kSampleRate32kHz = 32000,
-    kSampleRate48kHz = 48000
-  };
-
-  // TODO(kwiberg): We currently need to support a compiler (Visual C++) that
-  // complains if we don't explicitly state the size of the array here. Remove
-  // the size when that's no longer the case.
-  static constexpr int kNativeSampleRatesHz[4] = {
-      kSampleRate8kHz, kSampleRate16kHz, kSampleRate32kHz, kSampleRate48kHz};
-  static constexpr size_t kNumNativeSampleRates =
-      arraysize(kNativeSampleRatesHz);
-  static constexpr int kMaxNativeSampleRateHz =
-      kNativeSampleRatesHz[kNumNativeSampleRates - 1];
-
-  static const int kChunkSizeMs = 10;
-};
-
-class RTC_EXPORT AudioProcessingBuilder {
- public:
-  AudioProcessingBuilder();
-  ~AudioProcessingBuilder();
-  // The AudioProcessingBuilder takes ownership of the echo_control_factory.
-  AudioProcessingBuilder& SetEchoControlFactory(
-      std::unique_ptr<EchoControlFactory> echo_control_factory) {
-    echo_control_factory_ = std::move(echo_control_factory);
-    return *this;
-  }
-  // The AudioProcessingBuilder takes ownership of the capture_post_processing.
-  AudioProcessingBuilder& SetCapturePostProcessing(
-      std::unique_ptr<CustomProcessing> capture_post_processing) {
-    capture_post_processing_ = std::move(capture_post_processing);
-    return *this;
-  }
-  // The AudioProcessingBuilder takes ownership of the render_pre_processing.
-  AudioProcessingBuilder& SetRenderPreProcessing(
-      std::unique_ptr<CustomProcessing> render_pre_processing) {
-    render_pre_processing_ = std::move(render_pre_processing);
-    return *this;
-  }
-  // The AudioProcessingBuilder takes ownership of the echo_detector.
-  AudioProcessingBuilder& SetEchoDetector(
-      rtc::scoped_refptr<EchoDetector> echo_detector) {
-    echo_detector_ = std::move(echo_detector);
-    return *this;
-  }
-  // The AudioProcessingBuilder takes ownership of the capture_analyzer.
-  AudioProcessingBuilder& SetCaptureAnalyzer(
-      std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) {
-    capture_analyzer_ = std::move(capture_analyzer);
-    return *this;
-  }
-  // This creates an APM instance using the previously set components. Calling
-  // the Create function resets the AudioProcessingBuilder to its initial state.
-  AudioProcessing* Create();
-  AudioProcessing* Create(const webrtc::Config& config);
-
- private:
-  std::unique_ptr<EchoControlFactory> echo_control_factory_;
-  std::unique_ptr<CustomProcessing> capture_post_processing_;
-  std::unique_ptr<CustomProcessing> render_pre_processing_;
-  rtc::scoped_refptr<EchoDetector> echo_detector_;
-  std::unique_ptr<CustomAudioAnalyzer> capture_analyzer_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioProcessingBuilder);
-};
-
-class StreamConfig {
- public:
-  // sample_rate_hz: The sampling rate of the stream.
-  //
-  // num_channels: The number of audio channels in the stream, excluding the
-  //               keyboard channel if it is present. When passing a
-  //               StreamConfig with an array of arrays T*[N],
-  //
-  //                N == {num_channels + 1  if  has_keyboard
-  //                     {num_channels      if  !has_keyboard
-  //
-  // has_keyboard: True if the stream has a keyboard channel. When has_keyboard
-  //               is true, the last channel in any corresponding list of
-  //               channels is the keyboard channel.
-  StreamConfig(int sample_rate_hz = 0,
-               size_t num_channels = 0,
-               bool has_keyboard = false)
-      : sample_rate_hz_(sample_rate_hz),
-        num_channels_(num_channels),
-        has_keyboard_(has_keyboard),
-        num_frames_(calculate_frames(sample_rate_hz)) {}
-
-  void set_sample_rate_hz(int value) {
-    sample_rate_hz_ = value;
-    num_frames_ = calculate_frames(value);
-  }
-  void set_num_channels(size_t value) { num_channels_ = value; }
-  void set_has_keyboard(bool value) { has_keyboard_ = value; }
-
-  int sample_rate_hz() const { return sample_rate_hz_; }
-
-  // The number of channels in the stream, not including the keyboard channel if
-  // present.
-  size_t num_channels() const { return num_channels_; }
-
-  bool has_keyboard() const { return has_keyboard_; }
-  size_t num_frames() const { return num_frames_; }
-  size_t num_samples() const { return num_channels_ * num_frames_; }
-
-  bool operator==(const StreamConfig& other) const {
-    return sample_rate_hz_ == other.sample_rate_hz_ &&
-           num_channels_ == other.num_channels_ &&
-           has_keyboard_ == other.has_keyboard_;
-  }
-
-  bool operator!=(const StreamConfig& other) const { return !(*this == other); }
-
- private:
-  static size_t calculate_frames(int sample_rate_hz) {
-    return static_cast<size_t>(AudioProcessing::kChunkSizeMs * sample_rate_hz /
-                               1000);
-  }
-
-  int sample_rate_hz_;
-  size_t num_channels_;
-  bool has_keyboard_;
-  size_t num_frames_;
-};
-
-class ProcessingConfig {
- public:
-  enum StreamName {
-    kInputStream,
-    kOutputStream,
-    kReverseInputStream,
-    kReverseOutputStream,
-    kNumStreamNames,
-  };
-
-  const StreamConfig& input_stream() const {
-    return streams[StreamName::kInputStream];
-  }
-  const StreamConfig& output_stream() const {
-    return streams[StreamName::kOutputStream];
-  }
-  const StreamConfig& reverse_input_stream() const {
-    return streams[StreamName::kReverseInputStream];
-  }
-  const StreamConfig& reverse_output_stream() const {
-    return streams[StreamName::kReverseOutputStream];
-  }
-
-  StreamConfig& input_stream() { return streams[StreamName::kInputStream]; }
-  StreamConfig& output_stream() { return streams[StreamName::kOutputStream]; }
-  StreamConfig& reverse_input_stream() {
-    return streams[StreamName::kReverseInputStream];
-  }
-  StreamConfig& reverse_output_stream() {
-    return streams[StreamName::kReverseOutputStream];
-  }
-
-  bool operator==(const ProcessingConfig& other) const {
-    for (int i = 0; i < StreamName::kNumStreamNames; ++i) {
-      if (this->streams[i] != other.streams[i]) {
-        return false;
-      }
-    }
-    return true;
-  }
-
-  bool operator!=(const ProcessingConfig& other) const {
-    return !(*this == other);
-  }
-
-  StreamConfig streams[StreamName::kNumStreamNames];
-};
-
-// Experimental interface for a custom analysis submodule.
-class CustomAudioAnalyzer {
- public:
-  // (Re-) Initializes the submodule.
-  virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
-  // Analyzes the given capture or render signal.
-  virtual void Analyze(const AudioBuffer* audio) = 0;
-  // Returns a string representation of the module state.
-  virtual std::string ToString() const = 0;
-
-  virtual ~CustomAudioAnalyzer() {}
-};
-
-// Interface for a custom processing submodule.
-class CustomProcessing {
- public:
-  // (Re-)Initializes the submodule.
-  virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
-  // Processes the given capture or render signal.
-  virtual void Process(AudioBuffer* audio) = 0;
-  // Returns a string representation of the module state.
-  virtual std::string ToString() const = 0;
-  // Handles RuntimeSettings. TODO(webrtc:9262): make pure virtual
-  // after updating dependencies.
-  virtual void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting);
-
-  virtual ~CustomProcessing() {}
-};
-
-// Interface for an echo detector submodule.
-class EchoDetector : public rtc::RefCountInterface {
- public:
-  // (Re-)Initializes the submodule.
-  virtual void Initialize(int capture_sample_rate_hz,
-                          int num_capture_channels,
-                          int render_sample_rate_hz,
-                          int num_render_channels) = 0;
-
-  // Analysis (not changing) of the render signal.
-  virtual void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) = 0;
-
-  // Analysis (not changing) of the capture signal.
-  virtual void AnalyzeCaptureAudio(
-      rtc::ArrayView<const float> capture_audio) = 0;
-
-  // Pack an AudioBuffer into a vector<float>.
-  static void PackRenderAudioBuffer(AudioBuffer* audio,
-                                    std::vector<float>* packed_buffer);
-
-  struct Metrics {
-    absl::optional<double> echo_likelihood;
-    absl::optional<double> echo_likelihood_recent_max;
-  };
-
-  // Collect current metrics from the echo detector.
-  virtual Metrics GetMetrics() const = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_