tg2sip/webrtc_dsp/modules/audio_processing/include/gain_control.h

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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
/*
* 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_INCLUDE_GAIN_CONTROL_H_
#define MODULES_AUDIO_PROCESSING_INCLUDE_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:
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() {}
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_