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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc2/clipping_predictor.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/gtest_prod_util.h"
namespace webrtc {
class MonoInputVolumeController;
// The input volume controller recommends what volume to use, handles volume
// changes and clipping detection and prediction. In particular, it handles
// changes triggered by the user (e.g., volume set to zero by a HW mute button).
// This class is not thread-safe.
// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
// convention.
class InputVolumeController final {
public:
// Config for the constructor.
struct Config {
// Minimum input volume that can be recommended. Not enforced when the
// applied input volume is zero outside startup.
int min_input_volume = 20;
// Lowest input volume level that will be applied in response to clipping.
int clipped_level_min = 70;
// Amount input volume level is lowered with every clipping event. Limited
// to (0, 255].
int clipped_level_step = 15;
// Proportion of clipped samples required to declare a clipping event.
// Limited to (0.0f, 1.0f).
float clipped_ratio_threshold = 0.1f;
// Time in frames to wait after a clipping event before checking again.
// Limited to values higher than 0.
int clipped_wait_frames = 300;
// Enables clipping prediction functionality.
bool enable_clipping_predictor = true;
// Speech level target range (dBFS). If the speech level is in the range
// [`target_range_min_dbfs`, `target_range_max_dbfs`], no input volume
// adjustments are done based on the speech level. For speech levels below
// and above the range, the targets `target_range_min_dbfs` and
// `target_range_max_dbfs` are used, respectively.
int target_range_max_dbfs = -30;
int target_range_min_dbfs = -50;
// Number of wait frames between the recommended input volume updates.
int update_input_volume_wait_frames = 100;
// Speech probability threshold: speech probabilities below the threshold
// are considered silence. Limited to [0.0f, 1.0f].
float speech_probability_threshold = 0.7f;
// Minimum speech frame ratio for volume updates to be allowed. Limited to
// [0.0f, 1.0f].
float speech_ratio_threshold = 0.6f;
};
// Ctor. `num_capture_channels` specifies the number of channels for the audio
// passed to `AnalyzePreProcess()` and `Process()`. Clamps
// `config.startup_min_level` in the [12, 255] range.
InputVolumeController(int num_capture_channels, const Config& config);
~InputVolumeController();
InputVolumeController(const InputVolumeController&) = delete;
InputVolumeController& operator=(const InputVolumeController&) = delete;
// TODO(webrtc:7494): Integrate initialization into ctor and remove.
void Initialize();
// Analyzes `audio_buffer` before `RecommendInputVolume()` is called so tha
// the analysis can be performed before digital processing operations take
// place (e.g., echo cancellation). The analysis consists of input clipping
// detection and prediction (if enabled).
void AnalyzeInputAudio(int applied_input_volume,
const AudioBuffer& audio_buffer);
// Adjusts the recommended input volume upwards/downwards based on the result
// of `AnalyzeInputAudio()` and on `speech_level_dbfs` (if specified). Must
// be called after `AnalyzeInputAudio()`. The value of `speech_probability`
// is expected to be in the range [0, 1] and `speech_level_dbfs` in the range
// [-90, 30] and both should be estimated after echo cancellation and noise
// suppression are applied. Returns a non-empty input volume recommendation if
// available. If `capture_output_used_` is true, returns the applied input
// volume.
absl::optional<int> RecommendInputVolume(
float speech_probability,
absl::optional<float> speech_level_dbfs);
// Stores whether the capture output will be used or not. Call when the
// capture stream output has been flagged to be used/not-used. If unused, the
// controller disregards all incoming audio.
void HandleCaptureOutputUsedChange(bool capture_output_used);
// Returns true if clipping prediction is enabled.
// TODO(bugs.webrtc.org/7494): Deprecate this method.
bool clipping_predictor_enabled() const { return !!clipping_predictor_; }
// Returns true if clipping prediction is used to adjust the input volume.
// TODO(bugs.webrtc.org/7494): Deprecate this method.
bool use_clipping_predictor_step() const {
return use_clipping_predictor_step_;
}
// Only use for testing: Use `RecommendInputVolume()` elsewhere.
// Returns the value of a member variable, needed for testing
// `AnalyzeInputAudio()`.
int recommended_input_volume() const { return recommended_input_volume_; }
// Only use for testing.
bool capture_output_used() const { return capture_output_used_; }
private:
friend class InputVolumeControllerTestHelper;
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest, MinInputVolumeDefault);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest, MinInputVolumeDisabled);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
MinInputVolumeOutOfRangeAbove);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
MinInputVolumeOutOfRangeBelow);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest, MinInputVolumeEnabled50);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
ClippingParametersVerified);
// Sets the applied input volume and resets the recommended input volume.
void SetAppliedInputVolume(int level);
void AggregateChannelLevels();
const int num_capture_channels_;
// Minimum input volume that can be recommended.
const int min_input_volume_;
// TODO(bugs.webrtc.org/7494): Once
// `AudioProcessingImpl::recommended_stream_analog_level()` becomes a trivial
// getter, leave uninitialized.
// Recommended input volume. After `SetAppliedInputVolume()` is called it
// holds holds the observed input volume. Possibly updated by
// `AnalyzePreProcess()` and `Process()`; after these calls, holds the
// recommended input volume.
int recommended_input_volume_ = 0;
// Applied input volume. After `SetAppliedInputVolume()` is called it holds
// the current applied volume.
absl::optional<int> applied_input_volume_;
bool capture_output_used_;
// Clipping detection and prediction.
const int clipped_level_step_;
const float clipped_ratio_threshold_;
const int clipped_wait_frames_;
const std::unique_ptr<ClippingPredictor> clipping_predictor_;
const bool use_clipping_predictor_step_;
int frames_since_clipped_;
int clipping_rate_log_counter_;
float clipping_rate_log_;
// Target range minimum and maximum. If the seech level is in the range
// [`target_range_min_dbfs`, `target_range_max_dbfs`], no volume adjustments
// take place. Instead, the digital gain controller is assumed to adapt to
// compensate for the speech level RMS error.
const int target_range_max_dbfs_;
const int target_range_min_dbfs_;
// Channel controllers updating the gain upwards/downwards.
std::vector<std::unique_ptr<MonoInputVolumeController>> channel_controllers_;
int channel_controlling_gain_ = 0;
};
// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
// convention.
class MonoInputVolumeController {
public:
MonoInputVolumeController(int min_input_volume_after_clipping,
int min_input_volume,
int update_input_volume_wait_frames,
float speech_probability_threshold,
float speech_ratio_threshold);
~MonoInputVolumeController();
MonoInputVolumeController(const MonoInputVolumeController&) = delete;
MonoInputVolumeController& operator=(const MonoInputVolumeController&) =
delete;
void Initialize();
void HandleCaptureOutputUsedChange(bool capture_output_used);
// Sets the current input volume.
void set_stream_analog_level(int input_volume) {
recommended_input_volume_ = input_volume;
}
// Lowers the recommended input volume in response to clipping based on the
// suggested reduction `clipped_level_step`. Must be called after
// `set_stream_analog_level()`.
void HandleClipping(int clipped_level_step);
// TODO(bugs.webrtc.org/7494): Rename, audio not passed to the method anymore.
// Adjusts the recommended input volume upwards/downwards depending on the
// result of `HandleClipping()` and on `rms_error_dbfs`. Updates are only
// allowed for active speech segments and when `rms_error_dbfs` is not empty.
// Must be called after `HandleClipping()`.
void Process(absl::optional<int> rms_error_dbfs, float speech_probability);
// Returns the recommended input volume. Must be called after `Process()`.
int recommended_analog_level() const { return recommended_input_volume_; }
void ActivateLogging() { log_to_histograms_ = true; }
int min_input_volume_after_clipping() const {
return min_input_volume_after_clipping_;
}
// Only used for testing.
int min_input_volume() const { return min_input_volume_; }
private:
// Sets a new input volume, after first checking that it hasn't been updated
// by the user, in which case no action is taken.
void SetInputVolume(int new_volume);
// Sets the maximum input volume that the input volume controller is allowed
// to apply. The volume must be at least `kClippedLevelMin`.
void SetMaxLevel(int level);
int CheckVolumeAndReset();
// Updates the recommended input volume. If the volume slider needs to be
// moved, we check first if the user has adjusted it, in which case we take no
// action and cache the updated level.
void UpdateInputVolume(int rms_error_dbfs);
const int min_input_volume_;
const int min_input_volume_after_clipping_;
int max_input_volume_;
int last_recommended_input_volume_ = 0;
bool capture_output_used_ = true;
bool check_volume_on_next_process_ = true;
bool startup_ = true;
// TODO(bugs.webrtc.org/7494): Create a separate member for the applied
// input volume.
// Recommended input volume. After `set_stream_analog_level()` is
// called, it holds the observed applied input volume. Possibly updated by
// `HandleClipping()` and `Process()`; after these calls, holds the
// recommended input volume.
int recommended_input_volume_ = 0;
bool log_to_histograms_ = false;
// Counters for frames and speech frames since the last update in the
// recommended input volume.
const int update_input_volume_wait_frames_;
int frames_since_update_input_volume_ = 0;
int speech_frames_since_update_input_volume_ = 0;
bool is_first_frame_ = true;
// Speech probability threshold for a frame to be considered speech (instead
// of silence). Limited to [0.0f, 1.0f].
const float speech_probability_threshold_;
// Minimum ratio of speech frames. Limited to [0.0f, 1.0f].
const float speech_ratio_threshold_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_