1 /* /android/src/frameworks/base/libs/audioflinger/AudioShelvingFilter.h 2 ** 3 ** Copyright 2009, The Android Open Source Project 4 ** 5 ** Licensed under the Apache License, Version 2.0 (the "License"); 6 ** you may not use this file except in compliance with the License. 7 ** You may obtain a copy of the License at 8 ** 9 ** http://www.apache.org/licenses/LICENSE-2.0 10 ** 11 ** Unless required by applicable law or agreed to in writing, software 12 ** distributed under the License is distributed on an "AS IS" BASIS, 13 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14 ** See the License for the specific language governing permissions and 15 ** limitations under the License. 16 */ 17 18 #ifndef AUDIO_SHELVING_FILTER_H 19 #define AUDIO_SHELVING_FILTER_H 20 21 #include "AudioBiquadFilter.h" 22 #include "AudioCoefInterpolator.h" 23 24 namespace android { 25 26 // A shelving audio filter, with unity skirt gain, and controllable cutoff 27 // frequency and gain. 28 // This filter is able to suppress introduce discontinuities and other artifacts 29 // in the output, even when changing parameters abruptly. 30 // Parameters can be set to any value - this class will make sure to clip them 31 // when they are out of supported range. 32 // 33 // Implementation notes: 34 // This class uses an underlying biquad filter whose parameters are determined 35 // using a linear interpolation from a coefficient table, using a 36 // AudioCoefInterpolator. 37 // All is left for this class to do is mapping between high-level parameters to 38 // fractional indices into the coefficient table. 39 class AudioShelvingFilter { 40 public: 41 // Shelf type 42 enum ShelfType { 43 kLowShelf, 44 kHighShelf 45 }; 46 47 // Constructor. Resets the filter (see reset()). 48 // type Type of the filter (high shelf or low shelf). 49 // nChannels Number of input/output channels (interlaced). 50 // sampleRate The input/output sample rate, in Hz. 51 AudioShelvingFilter(ShelfType type, int nChannels, int sampleRate); 52 53 // Reconfiguration of the filter. Changes input/output format, but does not 54 // alter current parameter values. Clears delay lines. 55 // nChannels Number of input/output channels (interlaced). 56 // sampleRate The input/output sample rate, in Hz. 57 void configure(int nChannels, int sampleRate); 58 59 // Resets the filter parameters to the following values: 60 // frequency: 0 61 // gain: 0 62 // It also disables the filter. Does not clear the delay lines. 63 void reset(); 64 65 // Clears delay lines. Does not alter parameter values. 66 void clear() { mBiquad.clear(); } 67 68 // Sets gain value. Actual change will only take place upon commit(). 69 // This value will be remembered even if the filter is in disabled() state. 70 // millibel Gain value in millibel (1/100 of decibel). 71 void setGain(int32_t millibel); 72 73 // Gets the gain, in millibel, as set. 74 int32_t getGain() const { return mGain - 9600; } 75 76 // Sets cutoff frequency value. Actual change will only take place upon 77 // commit(). 78 // This value will be remembered even if the filter is in disabled() state. 79 // millihertz Frequency value in mHz. 80 void setFrequency(uint32_t millihertz); 81 82 // Gets the frequency, in mHz, as set. 83 uint32_t getFrequency() const { return mNominalFrequency; } 84 85 // Applies all parameter changes done to this point in time. 86 // If the filter is disabled, the new parameters will take place when it is 87 // enabled again. Does not introduce artifacts, unless immediate is set. 88 // immediate Whether to apply change abruptly (ignored if filter is 89 // disabled). 90 void commit(bool immediate = false); 91 92 // Process a buffer of input data. The input and output should contain 93 // frameCount * nChannels interlaced samples. Processing can be done 94 // in-place, by passing the same buffer as both arguments. 95 // in Input buffer. 96 // out Output buffer. 97 // frameCount Number of frames to produce. 98 void process(const audio_sample_t in[], audio_sample_t out[], 99 int frameCount) { mBiquad.process(in, out, frameCount); } 100 101 // Enables the filter, so it would start processing input. Does not 102 // introduce artifacts, unless immediate is set. 103 // immediate Whether to apply change abruptly. 104 void enable(bool immediate = false) { mBiquad.enable(immediate); } 105 106 // Disabled (bypasses) the filter. Does not introduce artifacts, unless 107 // immediate is set. 108 // immediate Whether to apply change abruptly. 109 void disable(bool immediate = false) { mBiquad.disable(immediate); } 110 111 private: 112 // Precision for the mFrequency member. 113 static const int FREQ_PRECISION_BITS = 26; 114 // Precision for the mGain member. 115 static const int GAIN_PRECISION_BITS = 10; 116 117 // Shelf type. 118 ShelfType mType; 119 // Nyquist, in mHz. 120 uint32_t mNiquistFreq; 121 // Fractional index into the gain dimension of the coef table in 122 // GAIN_PRECISION_BITS precision. 123 int32_t mGain; 124 // Fractional index into the frequency dimension of the coef table in 125 // FREQ_PRECISION_BITS precision. 126 uint32_t mFrequency; 127 // Nominal value of frequency, as set. 128 uint32_t mNominalFrequency; 129 // 1/Nyquist[mHz], in 42-bit precision (very small). 130 // Used for scaling the frequency. 131 uint32_t mFrequencyFactor; 132 133 // A biquad filter, used for the actual processing. 134 AudioBiquadFilter mBiquad; 135 // A coefficient interpolator, used for mapping the high level parameters to 136 // the low-level biquad coefficients. This one is used for the high shelf. 137 static AudioCoefInterpolator mHiCoefInterp; 138 // A coefficient interpolator, used for mapping the high level parameters to 139 // the low-level biquad coefficients. This one is used for the low shelf. 140 static AudioCoefInterpolator mLoCoefInterp; 141 }; 142 143 } 144 145 146 #endif // AUDIO_SHELVING_FILTER_H 147
