| /* |
| * 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 WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ |
| |
| #include <stddef.h> // size_t |
| #include <stdio.h> // FILE |
| #include <vector> |
| |
| #include "webrtc/base/platform_file.h" |
| #include "webrtc/common.h" |
| #include "webrtc/typedefs.h" |
| |
| struct AecCore; |
| |
| namespace webrtc { |
| |
| class AudioFrame; |
| class EchoCancellation; |
| class EchoControlMobile; |
| class GainControl; |
| class HighPassFilter; |
| class LevelEstimator; |
| class NoiseSuppression; |
| class VoiceDetection; |
| |
| // Use to enable the delay correction feature. This now engages an 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 DelayCorrection { |
| DelayCorrection() : enabled(false) {} |
| explicit DelayCorrection(bool enabled) : enabled(enabled) {} |
| bool enabled; |
| }; |
| |
| // Use to disable the reported system delays. By disabling the reported system |
| // delays the echo cancellation algorithm assumes the process and reverse |
| // streams to be aligned. This configuration only applies to EchoCancellation |
| // and not EchoControlMobile and is set with AudioProcessing::SetExtraOptions(). |
| // Note that by disabling reported system delays the EchoCancellation may |
| // regress in performance. |
| struct ReportedDelay { |
| ReportedDelay() : enabled(true) {} |
| explicit ReportedDelay(bool enabled) : enabled(enabled) {} |
| bool enabled; |
| }; |
| |
| // Must be provided through AudioProcessing::Create(Confg&). It will have no |
| // impact if used with AudioProcessing::SetExtraOptions(). |
| struct ExperimentalAgc { |
| ExperimentalAgc() : enabled(true) {} |
| explicit ExperimentalAgc(bool enabled) : enabled(enabled) {} |
| bool enabled; |
| }; |
| |
| // 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) {} |
| bool enabled; |
| }; |
| |
| // Coordinates in meters. |
| struct Point { |
| Point(float x, float y, float z) { |
| c[0] = x; |
| c[1] = y; |
| c[2] = z; |
| } |
| float c[3]; |
| }; |
| |
| // Use to enable beamforming. Must be provided through the constructor. It will |
| // have no impact if used with AudioProcessing::SetExtraOptions(). |
| struct Beamforming { |
| Beamforming() : enabled(false) {} |
| Beamforming(bool enabled, const std::vector<Point>& array_geometry) |
| : enabled(enabled), |
| array_geometry(array_geometry) {} |
| const bool enabled; |
| const std::vector<Point> array_geometry; |
| }; |
| |
| static const int kAudioProcMaxNativeSampleRateHz = 32000; |
| |
| // 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, which are used for |
| // analysis by some components, are passed to |AnalyzeReverseStream()|. 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->AnalyzeReverseStream(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: |
| enum ChannelLayout { |
| kMono, |
| // Left, right. |
| kStereo, |
| // Mono, keyboard mic. |
| kMonoAndKeyboard, |
| // Left, right, keyboard mic. |
| kStereoAndKeyboard |
| }; |
| |
| // 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); |
| 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 |AnalyzeReverseStream()| 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 |output_layout| matches |input_layout| |
| // |
| // The float interfaces accept arbitrary rates and support differing input |
| // and output layouts, but the output may only remove channels, not add. |
| 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; |
| |
| // DEPRECATED. |
| // TODO(ajm): Remove after Chromium has upgraded to using Initialize(). |
| virtual int set_sample_rate_hz(int rate) = 0; |
| // TODO(ajm): Remove after voice engine no longer requires it to resample |
| // the reverse stream to the forward rate. |
| virtual int input_sample_rate_hz() const = 0; |
| // TODO(ajm): Remove after Chromium no longer depends on it. |
| virtual int sample_rate_hz() const = 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 int num_input_channels() const = 0; |
| virtual int num_output_channels() const = 0; |
| virtual int 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; |
| virtual bool output_will_be_muted() const = 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 may only remove channels, not add. |src| and |dest| |
| // may use the same memory, if desired. |
| virtual int ProcessStream(const float* const* src, |
| int samples_per_channel, |
| int input_sample_rate_hz, |
| ChannelLayout input_layout, |
| int output_sample_rate_hz, |
| ChannelLayout output_layout, |
| float* const* dest) = 0; |
| |
| // Analyzes a 10 ms |frame| of the reverse direction audio stream. The frame |
| // will not be modified. On the client-side, this is the far-end (or to be |
| // rendered) audio. |
| // |
| // It is only 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. |sample_rate_hz_| must correspond to |
| // |input_sample_rate_hz()| |
| // |
| // TODO(ajm): add const to input; requires an implementation fix. |
| virtual int AnalyzeReverseStream(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|. |
| virtual int AnalyzeReverseStream(const float* const* data, |
| int samples_per_channel, |
| int sample_rate_hz, |
| ChannelLayout layout) = 0; |
| |
| // This must be called if and only if echo processing is enabled. |
| // |
| // Sets the |delay| in ms between AnalyzeReverseStream() 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 AnalyzeReverseStream() 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; |
| virtual bool stream_key_pressed() const = 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. |
| static const size_t kMaxFilenameSize = 1024; |
| virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0; |
| |
| // Same as above but uses an existing file handle. Takes ownership |
| // of |handle| and closes it at StopDebugRecording(). |
| virtual int StartDebugRecording(FILE* handle) = 0; |
| |
| // 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) { |
| return -1; |
| } |
| |
| // Stops recording debugging information, and closes the file. Recording |
| // cannot be resumed in the same file (without overwriting it). |
| virtual int StopDebugRecording() = 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 kChunkSizeMs = 10; |
| }; |
| |
| // 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 { |
| public: |
| // 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 { |
| kLowSuppression, |
| kModerateSuppression, |
| kHighSuppression |
| }; |
| |
| // 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 { |
| // RERL = ERL + ERLE |
| 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; |
| }; |
| |
| // 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|. The values are averaged over the time period since the |
| // last call to |GetDelayMetrics()|. |
| virtual int GetDelayMetrics(int* median, int* std) = 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; |
| |
| protected: |
| 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 { |
| public: |
| // 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 { |
| kQuietEarpieceOrHeadset, |
| kEarpiece, |
| kLoudEarpiece, |
| kSpeakerphone, |
| kLoudSpeakerphone |
| }; |
| |
| // 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(); |
| |
| protected: |
| 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 { |
| public: |
| 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. |
| 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 |
| }; |
| |
| 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; |
| |
| protected: |
| virtual ~GainControl() {} |
| }; |
| |
| // A filtering component which removes DC offset and low-frequency noise. |
| // Recommended to be enabled on the client-side. |
| class HighPassFilter { |
| public: |
| virtual int Enable(bool enable) = 0; |
| virtual bool is_enabled() const = 0; |
| |
| protected: |
| virtual ~HighPassFilter() {} |
| }; |
| |
| // An estimation component used to retrieve level metrics. |
| class LevelEstimator { |
| public: |
| 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: https://tools.ietf.org/html/rfc6465 |
| // 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; |
| |
| protected: |
| 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 { |
| public: |
| 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; |
| |
| protected: |
| 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 { |
| public: |
| 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 { |
| kVeryLowLikelihood, |
| kLowLikelihood, |
| kModerateLikelihood, |
| kHighLikelihood |
| }; |
| |
| 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; |
| |
| protected: |
| virtual ~VoiceDetection() {} |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ |