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Hypothetical scenario: short weak speech at start of call, then high noise. The digital adaptive AGC2 would pick a high gain, and then continue to apply it on the noise. Unless the noise is detected by the noise estimator, the gain would never be reduced. This CL addresses the issue by sending limiter gain info to the adaptive digital AGC2. Bug: webrtc:7494 Change-Id: Idf5c2686af0f5e5bad981d39a95b8efc9ffb9d64 Reviewed-on: https://webrtc-review.googlesource.com/102641 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24922}
65 lines
2.3 KiB
C++
65 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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float LastAudioLevel() const { return filter_state_level_; }
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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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