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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/capture_levels_adjuster/capture_levels_adjuster.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/capture_levels_adjuster/capture_levels_adjuster.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h | 88 |
1 files changed, 88 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h new file mode 100644 index 0000000000..38b68ad06c --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h @@ -0,0 +1,88 @@ +/* + * Copyright (c) 2021 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_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_ +#define MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_ + +#include <stddef.h> + +#include "modules/audio_processing/audio_buffer.h" +#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h" + +namespace webrtc { + +// Adjusts the level of the capture signal before and after all capture-side +// processing is done using a combination of explicitly specified gains +// and an emulated analog gain functionality where a specified analog level +// results in an additional gain. The pre-adjustment is achieved by combining +// the gain value `pre_gain` and the level `emulated_analog_mic_gain_level` to +// form a combined gain of `pre_gain`*`emulated_analog_mic_gain_level`/255 which +// is multiplied to each sample. The intention of the +// `emulated_analog_mic_gain_level` is to be controlled by the analog AGC +// functionality and to produce an emulated analog mic gain equal to +// `emulated_analog_mic_gain_level`/255. The post level adjustment is achieved +// by multiplying each sample with the value of `post_gain`. Any changes in the +// gains take are done smoothly over one frame and the scaled samples are +// clamped to fit into the allowed S16 sample range. +class CaptureLevelsAdjuster { + public: + // C-tor. The values for the level and the gains must fulfill + // 0 <= emulated_analog_mic_gain_level <= 255. + // 0.f <= pre_gain. + // 0.f <= post_gain. + CaptureLevelsAdjuster(bool emulated_analog_mic_gain_enabled, + int emulated_analog_mic_gain_level, + float pre_gain, + float post_gain); + CaptureLevelsAdjuster(const CaptureLevelsAdjuster&) = delete; + CaptureLevelsAdjuster& operator=(const CaptureLevelsAdjuster&) = delete; + + // Adjusts the level of the signal. This should be called before any of the + // other processing is performed. + void ApplyPreLevelAdjustment(AudioBuffer& audio_buffer); + + // Adjusts the level of the signal. This should be called after all of the + // other processing have been performed. + void ApplyPostLevelAdjustment(AudioBuffer& audio_buffer); + + // Sets the gain to apply to each sample before any of the other processing is + // performed. + void SetPreGain(float pre_gain); + + // Returns the total pre-adjustment gain applied, comprising both the pre_gain + // as well as the gain from the emulated analog mic, to each sample before any + // of the other processing is performed. + float GetPreAdjustmentGain() const { return pre_adjustment_gain_; } + + // Sets the gain to apply to each sample after all of the other processing + // have been performed. + void SetPostGain(float post_gain); + + // Sets the analog gain level to use for the emulated analog gain. + // `level` must be in the range [0...255]. + void SetAnalogMicGainLevel(int level); + + // Returns the current analog gain level used for the emulated analog gain. + int GetAnalogMicGainLevel() const { return emulated_analog_mic_gain_level_; } + + private: + // Updates the value of `pre_adjustment_gain_` based on the supplied values + // for `pre_gain` and `emulated_analog_mic_gain_level_`. + void UpdatePreAdjustmentGain(); + + const bool emulated_analog_mic_gain_enabled_; + int emulated_analog_mic_gain_level_; + float pre_gain_; + float pre_adjustment_gain_; + AudioSamplesScaler pre_scaler_; + AudioSamplesScaler post_scaler_; +}; +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_ |