| /* |
| * 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/audio/echo_canceller3_config.h" |
| #include "api/audio/echo_control.h" |
| #include "api/scoped_refptr.h" |
| #include "modules/audio_processing/include/audio_generator.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/deprecation.h" |
| #include "rtc_base/ref_count.h" |
| #include "rtc_base/system/rtc_export.h" |
| |
| namespace webrtc { |
| |
| class AecDump; |
| class AudioBuffer; |
| class AudioFrame; |
| |
| 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 or using AudioProcessing::SetExtraOptions(). |
| // 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. |
| // In addition to |voice_detected|, VAD decision is provided through the |
| // |AudioFrame| passed to |ProcessStream()|. The |vad_activity_| member will |
| // be modified to reflect the current decision. |
| 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; |
| LevelEstimator level_estimator = kRms; |
| bool use_saturation_protector = true; |
| float extra_saturation_margin_db = 2.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 |
| }; |
| |
| enum class VoiceDetectionResult { kNotAvailable, kDetected, kNotDetected }; |
| |
| // 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 |
| }; |
| |
| // 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}; |
| } |
| |
| 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 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; |
| 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. |
| 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; |
| |
| // Pass down additional options which don't have explicit setters. This |
| // ensures the options are applied immediately. |
| virtual void SetExtraOptions(const webrtc::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; |
| |
| // Processes a 10 ms |frame| of the primary audio stream. On the client-side, |
| // this is the near-end (or captured) audio. |
| // |
| // If needed for enabled functionality, any function with the set_stream_ tag |
| // must be called prior to processing the current frame. Any getter function |
| // with the stream_ tag which is needed should be called after processing. |
| // |
| // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| |
| // members of |frame| must be valid. If changed from the previous call to this |
| // method, it will trigger an initialization. |
| virtual int ProcessStream(AudioFrame* frame) = 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, |
| VoiceDetectionResult* vad_result) = 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; |
| |
| // Processes a 10 ms |frame| of the reverse direction audio stream. The frame |
| // may be modified. On the client-side, this is the far-end (or to be |
| // rendered) audio. |
| // |
| // It is necessary to provide this if echo processing is enabled, as the |
| // reverse stream forms the echo reference signal. It is recommended, but not |
| // necessary, to provide if gain control is enabled. On the server-side this |
| // typically will not be used. If you're not sure what to pass in here, |
| // chances are you don't need to use it. |
| // |
| // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| |
| // members of |frame| must be valid. |
| virtual int ProcessReverseStream(AudioFrame* frame) = 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; |
| |
| // 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; |
| |
| // Attaches provided webrtc::AudioGenerator for modifying playout audio. |
| // Calling this method when another AudioGenerator is attached replaces the |
| // active AudioGenerator with a new one. |
| virtual void AttachPlayoutAudioGenerator( |
| std::unique_ptr<AudioGenerator> audio_generator) = 0; |
| |
| // If no AudioGenerator is attached, this has no effect. If an AecDump is |
| // attached, its destructor is called. |
| virtual void DetachPlayoutAudioGenerator() = 0; |
| |
| // Use to send UMA histograms at end of a call. Note that all histogram |
| // specific member variables are reset. |
| // Deprecated. This method is deprecated and will be removed. |
| // TODO(peah): Remove this method. |
| virtual void UpdateHistogramsOnCallEnd() = 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 AudioFrame 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); |
| // The AudioProcessingBuilder takes ownership of the capture_post_processing. |
| AudioProcessingBuilder& SetCapturePostProcessing( |
| std::unique_ptr<CustomProcessing> capture_post_processing); |
| // The AudioProcessingBuilder takes ownership of the render_pre_processing. |
| AudioProcessingBuilder& SetRenderPreProcessing( |
| std::unique_ptr<CustomProcessing> render_pre_processing); |
| // The AudioProcessingBuilder takes ownership of the echo_detector. |
| AudioProcessingBuilder& SetEchoDetector( |
| rtc::scoped_refptr<EchoDetector> echo_detector); |
| // The AudioProcessingBuilder takes ownership of the capture_analyzer. |
| AudioProcessingBuilder& SetCaptureAnalyzer( |
| std::unique_ptr<CustomAudioAnalyzer> capture_analyzer); |
| // 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 { |
| double echo_likelihood; |
| 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_ |