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* 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.
// MSVC++ requires this to be set before any other includes to get M_PI.
#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 "modules/audio_processing/include/gain_control.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
struct AecCore;
class AecDump;
class AudioBuffer;
class AudioFrame;
class StreamConfig;
class ProcessingConfig;
class EchoDetector;
class GainControl;
class LevelEstimator;
class NoiseSuppression;
class CustomAudioAnalyzer;
class CustomProcessing;
class VoiceDetection;
// Use to enable the extended filter mode in the AEC, along with robustness
// measures around the reported system delays. It comes with a significant
// increase in AEC complexity, but is much more robust to unreliable reported
// delays.
// Detailed changes to the algorithm:
// - The filter length is changed from 48 to 128 ms. This comes with tuning of
// several parameters: i) filter adaptation stepsize and error threshold;
// ii) non-linear processing smoothing and overdrive.
// - Option to ignore the reported delays on platforms which we deem
// sufficiently unreliable. See WEBRTC_UNTRUSTED_DELAY in echo_cancellation.c.
// - Faster startup times by removing the excessive "startup phase" processing
// of reported delays.
// - Much more conservative adjustments to the far-end read pointer. We smooth
// the delay difference more heavily, and back off from the difference more.
// Adjustments force a readaptation of the filter, so they should be avoided
// except when really necessary.
struct ExtendedFilter {
ExtendedFilter() : enabled(false) {}
explicit ExtendedFilter(bool enabled) : enabled(enabled) {}
static const ConfigOptionID identifier = ConfigOptionID::kExtendedFilter;
bool enabled;
// Enables the refined linear filter adaptation in the echo canceller.
// This configuration only applies to non-mobile echo cancellation.
// It can be set in the constructor or using AudioProcessing::SetExtraOptions().
struct RefinedAdaptiveFilter {
RefinedAdaptiveFilter() : enabled(false) {}
explicit RefinedAdaptiveFilter(bool enabled) : enabled(enabled) {}
static const ConfigOptionID identifier =
bool enabled;
// Enables delay-agnostic echo cancellation. This feature relies on internally
// estimated delays between the process and reverse streams, thus not relying
// on reported system delays. This configuration only applies to non-mobile echo
// cancellation. It can be set in the constructor or using
// AudioProcessing::SetExtraOptions().
struct DelayAgnostic {
DelayAgnostic() : enabled(false) {}
explicit DelayAgnostic(bool enabled) : enabled(enabled) {}
static const ConfigOptionID identifier = ConfigOptionID::kDelayAgnostic;
bool enabled;
// 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).
static const int kAgcStartupMinVolume = 85;
static const int kAgcStartupMinVolume = 0;
#endif // defined(WEBRTC_CHROMIUM_BUILD)
static constexpr int kClippedLevelMin = 70;
struct ExperimentalAgc {
ExperimentalAgc() = default;
explicit ExperimentalAgc(bool enabled) : enabled(enabled) {}
ExperimentalAgc(bool enabled,
bool enabled_agc2_level_estimator,
bool digital_adaptive_disabled,
bool analyze_before_aec)
: enabled(enabled),
analyze_before_aec(analyze_before_aec) {}
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),
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;
// 'analyze_before_aec' is an experimental flag. It is intended to be removed
// at some point.
bool analyze_before_aec = false;
// Use to enable experimental noise suppression. It can be set in the
// constructor or using AudioProcessing::SetExtraOptions().
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.high_pass_filter.enabled = true;
// config.gain_controller2.enabled = true;
// apm->ApplyConfig(config)
// apm->noise_reduction()->set_level(kHighSuppression);
// apm->noise_reduction()->Enable(true);
// apm->gain_control()->set_analog_level_limits(0, 255);
// apm->gain_control()->set_mode(kAdaptiveAnalog);
// apm->gain_control()->Enable(true);
// apm->voice_detection()->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->gain_control()->set_stream_analog_level(analog_level);
// apm->ProcessStream(capture_frame);
// // Call required stream_ functions.
// analog_level = apm->gain_control()->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 AudioProcessing : public rtc::RefCountInterface {
// 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.
struct Config {
// 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;
} high_pass_filter;
struct EchoCanceller {
bool enabled = false;
bool mobile_mode = false;
// Recommended not to use. Will be removed in the future.
// APM components are not fine-tuned for legacy suppression levels.
bool legacy_moderate_suppression_level = false;
// Recommended not to use. Will be removed in the future.
bool use_legacy_aec = false;
} echo_canceller;
// Enables background noise suppression.
struct NoiseSuppression {
bool enabled = false;
enum Level { kLow, kModerate, kHigh, kVeryHigh };
Level level = kModerate;
} noise_suppression;
// Enables reporting of |has_voice| in webrtc::AudioProcessingStats.
struct VoiceDetection {
bool enabled = false;
} voice_detection;
// 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;
// Explicit copy assignment implementation to avoid issues with memory
// sanitizer complaints in case of self-assignment.
// TODO(peah): Add buildflag to ensure that this is only included for memory
// sanitizer builds.
Config& operator=(const Config& config) {
if (this != &config) {
memcpy(this, &config, sizeof(*this));
return *this;
// TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
enum ChannelLayout {
// Left, right.
// Mono, keyboard, and mic.
// Left, right, keyboard, and mic.
// Specifies the properties of a setting to be passed to AudioProcessing at
// runtime.
class RuntimeSetting {
enum class Type {
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};
static RuntimeSetting CreateCustomRenderSetting(float payload) {
return {Type::kCustomRenderProcessingRuntimeSetting, payload};
Type type() const { return type_; }
void GetFloat(float* value) const {
*value = value_;
RuntimeSetting(Type id, float value) : type_(id), value_(value) {}
Type type_;
float 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 deinterleaved float audio with the range [-1, 1]. Each element
// of |src| points to a channel buffer, arranged according to
// |input_layout|. At output, the channels will be arranged according to
// |output_layout| at |output_sample_rate_hz| in |dest|.
// The output layout must have one channel or as many channels as the input.
// |src| and |dest| may use the same memory, if desired.
// TODO(mgraczyk): Remove once clients are updated to use the new interface.
virtual int ProcessStream(const float* const* src,
size_t samples_per_channel,
int input_sample_rate_hz,
ChannelLayout input_layout,
int output_sample_rate_hz,
ChannelLayout output_layout,
float* 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;
// 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 deinterleaved float audio with the range [-1, 1]. Each element
// of |data| points to a channel buffer, arranged according to |layout|.
// TODO(mgraczyk): Remove once clients are updated to use the new interface.
virtual int AnalyzeReverseStream(const float* const* data,
size_t samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) = 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;
// 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;
virtual bool was_stream_delay_set() 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;
// Sets a delay |offset| in ms to add to the values passed in through
// set_stream_delay_ms(). May be positive or negative.
// Note that this could cause an otherwise valid value passed to
// set_stream_delay_ms() to return an error.
virtual void set_delay_offset_ms(int offset) = 0;
virtual int delay_offset_ms() const = 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.
virtual void UpdateHistogramsOnCallEnd() = 0;
// Get audio processing statistics. 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) const = 0;
// These provide access to the component interfaces and should never return
// NULL. The pointers will be valid for the lifetime of the APM instance.
// The memory for these objects is entirely managed internally.
virtual GainControl* gain_control() const = 0;
virtual LevelEstimator* level_estimator() const = 0;
virtual NoiseSuppression* noise_suppression() const = 0;
virtual VoiceDetection* voice_detection() const = 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
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 =
static constexpr int kMaxNativeSampleRateHz =
kNativeSampleRatesHz[kNumNativeSampleRates - 1];
static const int kChunkSizeMs = 10;
class RTC_EXPORT 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);
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 {
// 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_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); }
static size_t calculate_frames(int sample_rate_hz) {
return static_cast<size_t>(AudioProcessing::kChunkSizeMs * sample_rate_hz /
int sample_rate_hz_;
size_t num_channels_;
bool has_keyboard_;
size_t num_frames_;
class ProcessingConfig {
enum StreamName {
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];
// An estimation component used to retrieve level metrics.
class LevelEstimator {
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// Returns the root mean square (RMS) level in dBFs (decibels from digital
// full-scale), or alternately dBov. It is computed over all primary stream
// frames since the last call to RMS(). The returned value is positive but
// should be interpreted as negative. It is constrained to [0, 127].
// The computation follows:
// with the intent that it can provide the RTP audio level indication.
// Frames passed to ProcessStream() with an |_energy| of zero are considered
// to have been muted. The RMS of the frame will be interpreted as -127.
virtual int RMS() = 0;
virtual ~LevelEstimator() {}
// The noise suppression (NS) component attempts to remove noise while
// retaining speech. Recommended to be enabled on the client-side.
// Recommended to be enabled on the client-side.
class NoiseSuppression {
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// Determines the aggressiveness of the suppression. Increasing the level
// will reduce the noise level at the expense of a higher speech distortion.
enum Level { kLow, kModerate, kHigh, kVeryHigh };
virtual int set_level(Level level) = 0;
virtual Level level() const = 0;
// Returns the internally computed prior speech probability of current frame
// averaged over output channels. This is not supported in fixed point, for
// which |kUnsupportedFunctionError| is returned.
virtual float speech_probability() const = 0;
// Returns the noise estimate per frequency bin averaged over all channels.
virtual std::vector<float> NoiseEstimate() = 0;
virtual ~NoiseSuppression() {}
// Experimental interface for a custom analysis submodule.
class CustomAudioAnalyzer {
// (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 {
// (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 {
// (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;
// The voice activity detection (VAD) component analyzes the stream to
// determine if voice is present. A facility is also provided to pass in an
// external VAD decision.
// In addition to |stream_has_voice()| the VAD decision is provided through the
// |AudioFrame| passed to |ProcessStream()|. The |vad_activity_| member will be
// modified to reflect the current decision.
class VoiceDetection {
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// Returns true if voice is detected in the current frame. Should be called
// after |ProcessStream()|.
virtual bool stream_has_voice() const = 0;
// Some of the APM functionality requires a VAD decision. In the case that
// a decision is externally available for the current frame, it can be passed
// in here, before |ProcessStream()| is called.
// VoiceDetection does _not_ need to be enabled to use this. If it happens to
// be enabled, detection will be skipped for any frame in which an external
// VAD decision is provided.
virtual int set_stream_has_voice(bool has_voice) = 0;
// Specifies the likelihood that a frame will be declared to contain voice.
// A higher value makes it more likely that speech will not be clipped, at
// the expense of more noise being detected as voice.
enum Likelihood {
virtual int set_likelihood(Likelihood likelihood) = 0;
virtual Likelihood likelihood() const = 0;
// Sets the |size| of the frames in ms on which the VAD will operate. Larger
// frames will improve detection accuracy, but reduce the frequency of
// updates.
// This does not impact the size of frames passed to |ProcessStream()|.
virtual int set_frame_size_ms(int size) = 0;
virtual int frame_size_ms() const = 0;
virtual ~VoiceDetection() {}
} // namespace webrtc