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When FixedGainController::SetGain() is called first on a large value (e.g., 40 dB) and afterwards on a smaller one (e.g., 0 dB), the limiter used by FixedGainController takes time (about 10-20 seconds) to converge. During that period, the audio is not audible and the volume slowly increases. Even if switching from 40 dB to 0 dB is unlikely, this behavior can be corrected by resetting the limiter every time that FixedGainController::SetGain() is called. This eliminates the undesired effect described above even when the transient is short. Bug: webrtc:7494 Change-Id: I419b8986d2181448b4671cdbbd1c256dfb460216 Reviewed-on: https://webrtc-review.googlesource.com/94902 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24451}
63 lines
2.3 KiB
C++
63 lines
2.3 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_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
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#include <array>
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#include <vector>
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class ApmDataDumper;
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// Produces a smooth signal level estimate from an input audio
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// stream. The estimate smoothing is done through exponential
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// filtering.
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class FixedDigitalLevelEstimator {
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public:
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// Sample rates are allowed if the number of samples in a frame
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// (sample_rate_hz * kFrameDurationMs / 1000) is divisible by
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// kSubFramesInSample. For kFrameDurationMs=10 and
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// kSubFramesInSample=20, this means that sample_rate_hz has to be
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// divisible by 2000.
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FixedDigitalLevelEstimator(size_t sample_rate_hz,
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ApmDataDumper* apm_data_dumper);
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// The input is assumed to be in FloatS16 format. Scaled input will
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// produce similarly scaled output. A frame of with kFrameDurationMs
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// ms of audio produces a level estimates in the same scale. The
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// level estimate contains kSubFramesInFrame values.
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std::array<float, kSubFramesInFrame> ComputeLevel(
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const AudioFrameView<const float>& float_frame);
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// Rate may be changed at any time (but not concurrently) from the
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// value passed to the constructor. The class is not thread safe.
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void SetSampleRate(size_t sample_rate_hz);
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// Resets the level estimator internal state.
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void Reset();
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private:
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void CheckParameterCombination();
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ApmDataDumper* const apm_data_dumper_ = nullptr;
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float filter_state_level_;
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size_t samples_in_frame_;
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size_t samples_in_sub_frame_;
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RTC_DISALLOW_COPY_AND_ASSIGN(FixedDigitalLevelEstimator);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
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