webrtc/modules/audio_processing/agc/gain_control.h
Per Åhgren 0695df1a59 Reland "Replace the ExperimentalAgc config with the new config format"
This is a reland of f3aa6326b8

Original change's description:
> Replace the ExperimentalAgc config with the new config format
> 
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
> 
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
> 
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}

Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
2020-01-17 10:09:09 +00:00

105 lines
4.5 KiB
C++

/*
* Copyright (c) 2018 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_AGC_GAIN_CONTROL_H_
#define MODULES_AUDIO_PROCESSING_AGC_GAIN_CONTROL_H_
namespace webrtc {
// 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:
// 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() const = 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() {}
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC_GAIN_CONTROL_H_