blob: 44ff7327ffeb849184869c7bfadc921438093821 [file] [log] [blame]
* 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 <vector>
#include "webrtc/base/arraysize.h"
#include "webrtc/base/platform_file.h"
#include "webrtc/modules/audio_processing/beamformer/array_util.h"
#include "webrtc/modules/audio_processing/include/config.h"
#include "webrtc/typedefs.h"
namespace webrtc {
struct AecCore;
class AudioFrame;
class NonlinearBeamformer;
class StreamConfig;
class ProcessingConfig;
class EchoCancellation;
class EchoControlMobile;
class GainControl;
class HighPassFilter;
class LevelEstimator;
class NoiseSuppression;
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 next generation AEC functionality. This feature replaces the
// standard methods for echo removal in the AEC. This configuration only applies
// to EchoCancellation and not EchoControlMobile. It can be set in the
// constructor or using AudioProcessing::SetExtraOptions().
struct EchoCanceller3 {
EchoCanceller3() : enabled(false) {}
explicit EchoCanceller3(bool enabled) : enabled(enabled) {}
static const ConfigOptionID identifier = ConfigOptionID::kEchoCanceller3;
bool enabled;
// Enables the refined linear filter adaptation in the echo canceller.
// This configuration only applies to EchoCancellation and not
// EchoControlMobile. 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 the adaptive level controller.
struct LevelControl {
LevelControl() : enabled(false) {}
explicit LevelControl(bool enabled) : enabled(enabled) {}
static const ConfigOptionID identifier = ConfigOptionID::kLevelControl;
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
// EchoCancellation and not EchoControlMobile. 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 AudioProcessing::Create(Confg&).
static const int kAgcStartupMinVolume = 85;
static const int kAgcStartupMinVolume = 0;
#endif // defined(WEBRTC_CHROMIUM_BUILD)
struct ExperimentalAgc {
ExperimentalAgc() : enabled(true), startup_min_volume(kAgcStartupMinVolume) {}
explicit ExperimentalAgc(bool enabled)
: enabled(enabled), startup_min_volume(kAgcStartupMinVolume) {}
ExperimentalAgc(bool enabled, int startup_min_volume)
: enabled(enabled), startup_min_volume(startup_min_volume) {}
static const ConfigOptionID identifier = ConfigOptionID::kExperimentalAgc;
bool enabled;
int startup_min_volume;
// 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;
// Use to enable beamforming. Must be provided through the constructor. It will
// have no impact if used with AudioProcessing::SetExtraOptions().
struct Beamforming {
Beamforming(bool enabled, const std::vector<Point>& array_geometry);
Beamforming(bool enabled,
const std::vector<Point>& array_geometry,
SphericalPointf target_direction);
static const ConfigOptionID identifier = ConfigOptionID::kBeamforming;
const bool enabled;
const std::vector<Point> array_geometry;
const SphericalPointf target_direction;
// Use to enable intelligibility enhancer in audio processing.
// Note: If enabled and the reverse stream has more than one output channel,
// the reverse stream will become an upmixed mono signal.
struct Intelligibility {
Intelligibility() : enabled(false) {}
explicit Intelligibility(bool enabled) : enabled(enabled) {}
static const ConfigOptionID identifier = ConfigOptionID::kIntelligibility;
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 = AudioProcessing::Create(0);
// apm->high_pass_filter()->Enable(true);
// apm->echo_cancellation()->enable_drift_compensation(false);
// apm->echo_cancellation()->Enable(true);
// 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 {
// TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
enum ChannelLayout {
// Left, right.
// Mono, keyboard mic.
// Left, right, keyboard mic.
// Creates an APM instance. Use one instance for every primary audio stream
// requiring processing. On the client-side, this would typically be one
// instance for the near-end stream, and additional instances for each far-end
// stream which requires processing. On the server-side, this would typically
// be one instance for every incoming stream.
static AudioProcessing* Create();
// Allows passing in an optional configuration at create-time.
static AudioProcessing* Create(const Config& config);
// Only for testing.
static AudioProcessing* Create(const Config& config,
NonlinearBeamformer* beamformer);
virtual ~AudioProcessing() {}
// 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 input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout) = 0;
// Pass down additional options which don't have explicit setters. This
// ensures the options are applied immediately.
virtual void SetExtraOptions(const Config& config) = 0;
// TODO(ajm): Only intended for internal use. Make private and friend the
// necessary classes?
virtual int proc_sample_rate_hz() const = 0;
virtual int proc_split_sample_rate_hz() const = 0;
virtual size_t num_input_channels() const = 0;
virtual size_t num_proc_channels() const = 0;
virtual size_t num_output_channels() const = 0;
virtual size_t num_reverse_channels() const = 0;
// Set to true when the output of AudioProcessing will be muted or in some
// other way not used. Ideally, the captured audio would still be processed,
// but some components may change behavior based on this information.
// Default false.
virtual void set_output_will_be_muted(bool muted) = 0;
// 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 rev_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& reverse_input_config,
const StreamConfig& reverse_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_pull 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;
// Starts recording debugging information to a file specified by |filename|,
// a NULL-terminated string. If there is an ongoing recording, the old file
// will be closed, and recording will continue in the newly specified file.
// An already existing file will be overwritten without warning. A maximum
// file size (in bytes) for the log can be specified. The logging is stopped
// once the limit has been reached. If max_log_size_bytes is set to a value
// <= 0, no limit will be used.
static const size_t kMaxFilenameSize = 1024;
virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
int64_t max_log_size_bytes) = 0;
// Same as above but uses an existing file handle. Takes ownership
// of |handle| and closes it at StopDebugRecording().
virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
// TODO(ivoc): Remove this function after Chrome stops using it.
int StartDebugRecording(FILE* handle) {
return StartDebugRecording(handle, -1);
// Same as above but uses an existing PlatformFile handle. Takes ownership
// of |handle| and closes it at StopDebugRecording().
// TODO(xians): Make this interface pure virtual.
virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle);
// Stops recording debugging information, and closes the file. Recording
// cannot be resumed in the same file (without overwriting it).
virtual int StopDebugRecording() = 0;
// Use to send UMA histograms at end of a call. Note that all histogram
// specific member variables are reset.
virtual void UpdateHistogramsOnCallEnd() = 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 EchoCancellation* echo_cancellation() const = 0;
virtual EchoControlMobile* echo_control_mobile() const = 0;
virtual GainControl* gain_control() const = 0;
virtual HighPassFilter* high_pass_filter() const = 0;
virtual LevelEstimator* level_estimator() const = 0;
virtual NoiseSuppression* noise_suppression() const = 0;
virtual VoiceDetection* voice_detection() const = 0;
struct Statistic {
int instant; // Instantaneous value.
int average; // Long-term average.
int maximum; // Long-term maximum.
int minimum; // Long-term minimum.
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
static const int kNativeSampleRatesHz[];
static const size_t kNumNativeSampleRates;
static const int kMaxNativeSampleRateHz;
static const int kChunkSizeMs = 10;
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 / 1000);
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];
// The acoustic echo cancellation (AEC) component provides better performance
// than AECM but also requires more processing power and is dependent on delay
// stability and reporting accuracy. As such it is well-suited and recommended
// for PC and IP phone applications.
// Not recommended to be enabled on the server-side.
class EchoCancellation {
// EchoCancellation and EchoControlMobile may not be enabled simultaneously.
// Enabling one will disable the other.
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// Differences in clock speed on the primary and reverse streams can impact
// the AEC performance. On the client-side, this could be seen when different
// render and capture devices are used, particularly with webcams.
// This enables a compensation mechanism, and requires that
// set_stream_drift_samples() be called.
virtual int enable_drift_compensation(bool enable) = 0;
virtual bool is_drift_compensation_enabled() const = 0;
// Sets the difference between the number of samples rendered and captured by
// the audio devices since the last call to |ProcessStream()|. Must be called
// if drift compensation is enabled, prior to |ProcessStream()|.
virtual void set_stream_drift_samples(int drift) = 0;
virtual int stream_drift_samples() const = 0;
enum SuppressionLevel {
// Sets the aggressiveness of the suppressor. A higher level trades off
// double-talk performance for increased echo suppression.
virtual int set_suppression_level(SuppressionLevel level) = 0;
virtual SuppressionLevel suppression_level() const = 0;
// Returns false if the current frame almost certainly contains no echo
// and true if it _might_ contain echo.
virtual bool stream_has_echo() const = 0;
// Enables the computation of various echo metrics. These are obtained
// through |GetMetrics()|.
virtual int enable_metrics(bool enable) = 0;
virtual bool are_metrics_enabled() const = 0;
// Each statistic is reported in dB.
// P_far: Far-end (render) signal power.
// P_echo: Near-end (capture) echo signal power.
// P_out: Signal power at the output of the AEC.
// P_a: Internal signal power at the point before the AEC's non-linear
// processor.
struct Metrics {
AudioProcessing::Statistic residual_echo_return_loss;
// ERL = 10log_10(P_far / P_echo)
AudioProcessing::Statistic echo_return_loss;
// ERLE = 10log_10(P_echo / P_out)
AudioProcessing::Statistic echo_return_loss_enhancement;
// (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
AudioProcessing::Statistic a_nlp;
// Fraction of time that the AEC linear filter is divergent, in a 1-second
// non-overlapped aggregation window.
float divergent_filter_fraction;
// TODO(ajm): discuss the metrics update period.
virtual int GetMetrics(Metrics* metrics) = 0;
// Enables computation and logging of delay values. Statistics are obtained
// through |GetDelayMetrics()|.
virtual int enable_delay_logging(bool enable) = 0;
virtual bool is_delay_logging_enabled() const = 0;
// The delay metrics consists of the delay |median| and the delay standard
// deviation |std|. It also consists of the fraction of delay estimates
// |fraction_poor_delays| that can make the echo cancellation perform poorly.
// The values are aggregated until the first call to |GetDelayMetrics()| and
// afterwards aggregated and updated every second.
// Note that if there are several clients pulling metrics from
// |GetDelayMetrics()| during a session the first call from any of them will
// change to one second aggregation window for all.
// TODO(bjornv): Deprecated, remove.
virtual int GetDelayMetrics(int* median, int* std) = 0;
virtual int GetDelayMetrics(int* median, int* std,
float* fraction_poor_delays) = 0;
// Returns a pointer to the low level AEC component. In case of multiple
// channels, the pointer to the first one is returned. A NULL pointer is
// returned when the AEC component is disabled or has not been initialized
// successfully.
virtual struct AecCore* aec_core() const = 0;
virtual ~EchoCancellation() {}
// The acoustic echo control for mobile (AECM) component is a low complexity
// robust option intended for use on mobile devices.
// Not recommended to be enabled on the server-side.
class EchoControlMobile {
// EchoCancellation and EchoControlMobile may not be enabled simultaneously.
// Enabling one will disable the other.
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// Recommended settings for particular audio routes. In general, the louder
// the echo is expected to be, the higher this value should be set. The
// preferred setting may vary from device to device.
enum RoutingMode {
// Sets echo control appropriate for the audio routing |mode| on the device.
// It can and should be updated during a call if the audio routing changes.
virtual int set_routing_mode(RoutingMode mode) = 0;
virtual RoutingMode routing_mode() const = 0;
// Comfort noise replaces suppressed background noise to maintain a
// consistent signal level.
virtual int enable_comfort_noise(bool enable) = 0;
virtual bool is_comfort_noise_enabled() const = 0;
// A typical use case is to initialize the component with an echo path from a
// previous call. The echo path is retrieved using |GetEchoPath()|, typically
// at the end of a call. The data can then be stored for later use as an
// initializer before the next call, using |SetEchoPath()|.
// Controlling the echo path this way requires the data |size_bytes| to match
// the internal echo path size. This size can be acquired using
// |echo_path_size_bytes()|. |SetEchoPath()| causes an entire reset, worth
// noting if it is to be called during an ongoing call.
// It is possible that version incompatibilities may result in a stored echo
// path of the incorrect size. In this case, the stored path should be
// discarded.
virtual int SetEchoPath(const void* echo_path, size_t size_bytes) = 0;
virtual int GetEchoPath(void* echo_path, size_t size_bytes) const = 0;
// The returned path size is guaranteed not to change for the lifetime of
// the application.
static size_t echo_path_size_bytes();
virtual ~EchoControlMobile() {}
// 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.
class GainControl {
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// When an analog mode is set, this must be called prior to |ProcessStream()|
// to pass the current analog level from the audio HAL. Must be within the
// range provided to |set_analog_level_limits()|.
virtual int 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
// users responsibility to apply this level.
virtual int stream_analog_level() = 0;
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.
// 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.
// 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.
virtual int set_mode(Mode mode) = 0;
virtual Mode mode() const = 0;
// 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].
// TODO(ajm): use a negative value here instead, if/when VoE will similarly
// update its interface.
virtual int set_target_level_dbfs(int level) = 0;
virtual int target_level_dbfs() const = 0;
// 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].
virtual int set_compression_gain_db(int gain) = 0;
virtual int compression_gain_db() const = 0;
// 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.
virtual int enable_limiter(bool enable) = 0;
virtual bool is_limiter_enabled() const = 0;
// Sets the |minimum| and |maximum| analog levels of the audio capture device.
// Must be set if and only if an analog mode is used. Limited to [0, 65535].
virtual int set_analog_level_limits(int minimum,
int maximum) = 0;
virtual int analog_level_minimum() const = 0;
virtual int analog_level_maximum() const = 0;
// Returns true if the AGC has detected a saturation event (period where the
// signal reaches digital full-scale) in the current frame and the analog
// level cannot be reduced.
// This could be used as an indicator to reduce or disable analog mic gain at
// the audio HAL.
virtual bool stream_is_saturated() const = 0;
virtual ~GainControl() {}
// A filtering component which removes DC offset and low-frequency noise.
// Recommended to be enabled on the client-side.
class HighPassFilter {
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
virtual ~HighPassFilter() {}
// 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 {
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() {}
// 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