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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/include/audio_processing.h')
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diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h new file mode 100644 index 0000000000..f613a38de1 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h @@ -0,0 +1,941 @@ +/* + * 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/strings/string_view.h" +#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 "rtc_base/arraysize.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; + +// 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 (see +// AudioProcessing::GetFrameSize() for details) and sample rates ranging from +// 8000 Hz to 384000 Hz. The int16 interfaces use interleaved data, while the +// float interfaces use deinterleaved data. +// +// Usage example, omitting error checking: +// rtc::scoped_refptr<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; +// +// apm->ApplyConfig(config) +// +// // 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(); +// +// // Repeat render and capture processing for the duration of the call... +// // Start a new call... +// apm->Initialize(); +// +// // Close the application... +// apm.reset(); +// +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 { + // Ways to downmix a multi-channel track to mono. + enum class DownmixMethod { + kAverageChannels, // Average across channels. + kUseFirstChannel // Use the first channel. + }; + + // Maximum allowed processing rate used internally. May only be set to + // 32000 or 48000 and any differing values will be treated as 48000. + int maximum_internal_processing_rate = 48000; + // 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; + // Indicates how to downmix multi-channel capture audio to mono (when + // needed). + DownmixMethod capture_downmix_method = DownmixMethod::kAverageChannels; + } pipeline; + + // Enabled the pre-amplifier. It amplifies the capture signal + // before any other processing is done. + // TODO(webrtc:5298): Deprecate and use the pre-gain functionality in + // capture_level_adjustment instead. + struct PreAmplifier { + bool enabled = false; + float fixed_gain_factor = 1.0f; + } pre_amplifier; + + // Functionality for general level adjustment in the capture pipeline. This + // should not be used together with the legacy PreAmplifier functionality. + struct CaptureLevelAdjustment { + bool operator==(const CaptureLevelAdjustment& rhs) const; + bool operator!=(const CaptureLevelAdjustment& rhs) const { + return !(*this == rhs); + } + bool enabled = false; + // The `pre_gain_factor` scales the signal before any processing is done. + float pre_gain_factor = 1.0f; + // The `post_gain_factor` scales the signal after all processing is done. + float post_gain_factor = 1.0f; + struct AnalogMicGainEmulation { + bool operator==(const AnalogMicGainEmulation& rhs) const; + bool operator!=(const AnalogMicGainEmulation& rhs) const { + return !(*this == rhs); + } + bool enabled = false; + // Initial analog gain level to use for the emulated analog gain. Must + // be in the range [0...255]. + int initial_level = 255; + } analog_mic_gain_emulation; + } capture_level_adjustment; + + 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 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 RTC_EXPORT GainController1 { + bool operator==(const GainController1& rhs) const; + bool operator!=(const GainController1& rhs) const { + return !(*this == rhs); + } + + 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; + + // Enables the analog gain controller functionality. + struct AnalogGainController { + bool enabled = true; + // TODO(bugs.webrtc.org/7494): Deprecated. Stop using and remove. + int startup_min_volume = 0; + // Lowest analog microphone level that will be applied in response to + // clipping. + int clipped_level_min = 70; + // If true, an adaptive digital gain is applied. + bool enable_digital_adaptive = true; + // Amount the microphone level is lowered with every clipping event. + // Limited to (0, 255]. + int clipped_level_step = 15; + // Proportion of clipped samples required to declare a clipping event. + // Limited to (0.f, 1.f). + float clipped_ratio_threshold = 0.1f; + // Time in frames to wait after a clipping event before checking again. + // Limited to values higher than 0. + int clipped_wait_frames = 300; + + // Enables clipping prediction functionality. + struct ClippingPredictor { + bool enabled = false; + enum Mode { + // Clipping event prediction mode with fixed step estimation. + kClippingEventPrediction, + // Clipped peak estimation mode with adaptive step estimation. + kAdaptiveStepClippingPeakPrediction, + // Clipped peak estimation mode with fixed step estimation. + kFixedStepClippingPeakPrediction, + }; + Mode mode = kClippingEventPrediction; + // Number of frames in the sliding analysis window. + int window_length = 5; + // Number of frames in the sliding reference window. + int reference_window_length = 5; + // Reference window delay (unit: number of frames). + int reference_window_delay = 5; + // Clipping prediction threshold (dBFS). + float clipping_threshold = -1.0f; + // Crest factor drop threshold (dB). + float crest_factor_margin = 3.0f; + // If true, the recommended clipped level step is used to modify the + // analog gain. Otherwise, the predictor runs without affecting the + // analog gain. + bool use_predicted_step = true; + } clipping_predictor; + } analog_gain_controller; + } gain_controller1; + + // Parameters for AGC2, an Automatic Gain Control (AGC) sub-module which + // replaces the AGC sub-module parametrized by `gain_controller1`. + // AGC2 brings the captured audio signal to the desired level by combining + // three different controllers (namely, input volume controller, adapative + // digital controller and fixed digital controller) and a limiter. + // TODO(bugs.webrtc.org:7494): Name `GainController` when AGC1 removed. + struct RTC_EXPORT GainController2 { + bool operator==(const GainController2& rhs) const; + bool operator!=(const GainController2& rhs) const { + return !(*this == rhs); + } + + // AGC2 must be created if and only if `enabled` is true. + bool enabled = false; + + // Parameters for the input volume controller, which adjusts the input + // volume applied when the audio is captured (e.g., microphone volume on + // a soundcard, input volume on HAL). + struct InputVolumeController { + bool operator==(const InputVolumeController& rhs) const; + bool operator!=(const InputVolumeController& rhs) const { + return !(*this == rhs); + } + bool enabled = false; + } input_volume_controller; + + // Parameters for the adaptive digital controller, which adjusts and + // applies a digital gain after echo cancellation and after noise + // suppression. + struct RTC_EXPORT AdaptiveDigital { + bool operator==(const AdaptiveDigital& rhs) const; + bool operator!=(const AdaptiveDigital& rhs) const { + return !(*this == rhs); + } + bool enabled = false; + float headroom_db = 6.0f; + float max_gain_db = 30.0f; + float initial_gain_db = 8.0f; + float max_gain_change_db_per_second = 3.0f; + float max_output_noise_level_dbfs = -50.0f; + } adaptive_digital; + + // Parameters for the fixed digital controller, which applies a fixed + // digital gain after the adaptive digital controller and before the + // limiter. + struct FixedDigital { + // By setting `gain_db` to a value greater than zero, the limiter can be + // turned into a compressor that first applies a fixed gain. + float gain_db = 0.0f; + } fixed_digital; + } gain_controller2; + + std::string ToString() const; + }; + + // 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, + kCapturePostGain, + 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_(0.0f) {} + ~RuntimeSetting() = default; + + static RuntimeSetting CreateCapturePreGain(float gain) { + return {Type::kCapturePreGain, gain}; + } + + static RuntimeSetting CreateCapturePostGain(float gain) { + return {Type::kCapturePostGain, 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, 0.0f); + RTC_DCHECK_LE(gain_db, 90.0f); + 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 capture_output_used) { + return {Type::kCaptureOutputUsed, capture_output_used}; + } + + 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; + + // 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. This method takes a lock. To achieve this in a lock-less + // manner the PostRuntimeSetting can instead be used. + virtual void set_output_will_be_muted(bool muted) = 0; + + // Enqueues a runtime setting. + virtual void SetRuntimeSetting(RuntimeSetting setting) = 0; + + // Enqueues a runtime setting. Returns a bool indicating whether the + // enqueueing was successfull. + virtual bool PostRuntimeSetting(RuntimeSetting setting) = 0; + + // Accepts and produces a ~10 ms frame of 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 [0, 255]. + virtual void set_stream_analog_level(int level) = 0; + + // When an analog mode is set, this should be called after + // `set_stream_analog_level()` and `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(absl::string_view 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]; + + // APM processes audio in chunks of about 10 ms. See GetFrameSize() for + // details. + static constexpr int kChunkSizeMs = 10; + + // Returns floor(sample_rate_hz/100): the number of samples per channel used + // as input and output to the audio processing module in calls to + // ProcessStream, ProcessReverseStream, AnalyzeReverseStream, and + // GetLinearAecOutput. + // + // This is exactly 10 ms for sample rates divisible by 100. For example: + // - 48000 Hz (480 samples per channel), + // - 44100 Hz (441 samples per channel), + // - 16000 Hz (160 samples per channel). + // + // Sample rates not divisible by 100 are received/produced in frames of + // approximately 10 ms. For example: + // - 22050 Hz (220 samples per channel, or ~9.98 ms per frame), + // - 11025 Hz (110 samples per channel, or ~9.98 ms per frame). + // These nondivisible sample rates yield lower audio quality compared to + // multiples of 100. Internal resampling to 10 ms frames causes a simulated + // clock drift effect which impacts the performance of (for example) echo + // cancellation. + static int GetFrameSize(int sample_rate_hz) { return sample_rate_hz / 100; } +}; + +class RTC_EXPORT AudioProcessingBuilder { + public: + AudioProcessingBuilder(); + AudioProcessingBuilder(const AudioProcessingBuilder&) = delete; + AudioProcessingBuilder& operator=(const AudioProcessingBuilder&) = delete; + ~AudioProcessingBuilder(); + + // Sets the APM configuration. + AudioProcessingBuilder& SetConfig(const AudioProcessing::Config& config) { + config_ = config; + return *this; + } + + // Sets the echo controller factory to inject when APM is created. + AudioProcessingBuilder& SetEchoControlFactory( + std::unique_ptr<EchoControlFactory> echo_control_factory) { + echo_control_factory_ = std::move(echo_control_factory); + return *this; + } + + // Sets the capture post-processing sub-module to inject when APM is created. + AudioProcessingBuilder& SetCapturePostProcessing( + std::unique_ptr<CustomProcessing> capture_post_processing) { + capture_post_processing_ = std::move(capture_post_processing); + return *this; + } + + // Sets the render pre-processing sub-module to inject when APM is created. + AudioProcessingBuilder& SetRenderPreProcessing( + std::unique_ptr<CustomProcessing> render_pre_processing) { + render_pre_processing_ = std::move(render_pre_processing); + return *this; + } + + // Sets the echo detector to inject when APM is created. + AudioProcessingBuilder& SetEchoDetector( + rtc::scoped_refptr<EchoDetector> echo_detector) { + echo_detector_ = std::move(echo_detector); + return *this; + } + + // Sets the capture analyzer sub-module to inject when APM is created. + AudioProcessingBuilder& SetCaptureAnalyzer( + std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) { + capture_analyzer_ = std::move(capture_analyzer); + return *this; + } + + // Creates an APM instance with the specified config or the default one if + // unspecified. Injects the specified components transferring the ownership + // to the newly created APM instance - i.e., except for the config, the + // builder is reset to its initial state. + rtc::scoped_refptr<AudioProcessing> Create(); + + private: + AudioProcessing::Config config_; + 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_; +}; + +class StreamConfig { + public: + // sample_rate_hz: The sampling rate of the stream. + // num_channels: The number of audio channels in the stream. + StreamConfig(int sample_rate_hz = 0, size_t num_channels = 0) + : sample_rate_hz_(sample_rate_hz), + num_channels_(num_channels), + 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; } + + int sample_rate_hz() const { return sample_rate_hz_; } + + // The number of channels in the stream. + size_t num_channels() const { return num_channels_; } + + 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_; + } + + 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::GetFrameSize(sample_rate_hz)); + } + + int sample_rate_hz_; + size_t num_channels_; + 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 first channel 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; + + 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_ |