mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
109 lines
4.6 KiB
C++
109 lines
4.6 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_
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#define MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_
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namespace webrtc {
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// The automatic gain control (AGC) component brings the signal to an
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// appropriate range. This is done by applying a digital gain directly and, in
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// the analog mode, prescribing an analog gain to be applied at the audio HAL.
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//
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// Recommended to be enabled on the client-side.
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class GainControl {
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public:
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virtual int Enable(bool enable) = 0;
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virtual bool is_enabled() const = 0;
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// When an analog mode is set, this must be called prior to |ProcessStream()|
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// to pass the current analog level from the audio HAL. Must be within the
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// range provided to |set_analog_level_limits()|.
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virtual int set_stream_analog_level(int level) = 0;
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// When an analog mode is set, this should be called after |ProcessStream()|
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// to obtain the recommended new analog level for the audio HAL. It is the
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// users responsibility to apply this level.
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virtual int stream_analog_level() = 0;
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enum Mode {
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// Adaptive mode intended for use if an analog volume control is available
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// on the capture device. It will require the user to provide coupling
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// between the OS mixer controls and AGC through the |stream_analog_level()|
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// functions.
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//
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// It consists of an analog gain prescription for the audio device and a
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// digital compression stage.
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kAdaptiveAnalog,
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// Adaptive mode intended for situations in which an analog volume control
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// is unavailable. It operates in a similar fashion to the adaptive analog
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// mode, but with scaling instead applied in the digital domain. As with
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// the analog mode, it additionally uses a digital compression stage.
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kAdaptiveDigital,
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// Fixed mode which enables only the digital compression stage also used by
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// the two adaptive modes.
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//
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// It is distinguished from the adaptive modes by considering only a
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// short time-window of the input signal. It applies a fixed gain through
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// most of the input level range, and compresses (gradually reduces gain
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// with increasing level) the input signal at higher levels. This mode is
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// preferred on embedded devices where the capture signal level is
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// predictable, so that a known gain can be applied.
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kFixedDigital
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};
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virtual int set_mode(Mode mode) = 0;
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virtual Mode mode() const = 0;
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// Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
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// from digital full-scale). The convention is to use positive values. For
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// instance, passing in a value of 3 corresponds to -3 dBFs, or a target
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// level 3 dB below full-scale. Limited to [0, 31].
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//
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// TODO(ajm): use a negative value here instead, if/when VoE will similarly
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// update its interface.
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virtual int set_target_level_dbfs(int level) = 0;
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virtual int target_level_dbfs() const = 0;
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// Sets the maximum |gain| the digital compression stage may apply, in dB. A
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// higher number corresponds to greater compression, while a value of 0 will
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// leave the signal uncompressed. Limited to [0, 90].
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virtual int set_compression_gain_db(int gain) = 0;
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virtual int compression_gain_db() const = 0;
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// When enabled, the compression stage will hard limit the signal to the
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// target level. Otherwise, the signal will be compressed but not limited
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// above the target level.
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virtual int enable_limiter(bool enable) = 0;
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virtual bool is_limiter_enabled() const = 0;
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// Sets the |minimum| and |maximum| analog levels of the audio capture device.
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// Must be set if and only if an analog mode is used. Limited to [0, 65535].
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virtual int set_analog_level_limits(int minimum, int maximum) = 0;
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virtual int analog_level_minimum() const = 0;
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virtual int analog_level_maximum() const = 0;
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// Returns true if the AGC has detected a saturation event (period where the
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// signal reaches digital full-scale) in the current frame and the analog
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// level cannot be reduced.
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//
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// This could be used as an indicator to reduce or disable analog mic gain at
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// the audio HAL.
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virtual bool stream_is_saturated() const = 0;
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protected:
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virtual ~GainControl() {}
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_
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