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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/agc/gain_control.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc/gain_control.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/agc/gain_control.h | 105 |
1 files changed, 105 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/gain_control.h b/third_party/libwebrtc/modules/audio_processing/agc/gain_control.h new file mode 100644 index 0000000000..389b2114af --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc/gain_control.h @@ -0,0 +1,105 @@ +/* + * 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_ |