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Diffstat (limited to 'media/libeffects/testlibs/AudioBiquadFilter.h')
| -rw-r--r-- | media/libeffects/testlibs/AudioBiquadFilter.h | 180 |
1 files changed, 180 insertions, 0 deletions
diff --git a/media/libeffects/testlibs/AudioBiquadFilter.h b/media/libeffects/testlibs/AudioBiquadFilter.h new file mode 100644 index 0000000..2b0e2d6 --- /dev/null +++ b/media/libeffects/testlibs/AudioBiquadFilter.h @@ -0,0 +1,180 @@ +/* //device/include/server/AudioFlinger/AudioBiquadFilter.h +** +** Copyright 2007, The Android Open Source Project +** +** Licensed under the Apache License, Version 2.0 (the "License"); +** you may not use this file except in compliance with the License. +** You may obtain a copy of the License at +** +** http://www.apache.org/licenses/LICENSE-2.0 +** +** Unless required by applicable law or agreed to in writing, software +** distributed under the License is distributed on an "AS IS" BASIS, +** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +** See the License for the specific language governing permissions and +** limitations under the License. +*/ + +#ifndef ANDROID_AUDIO_BIQUAD_FILTER_H +#define ANDROID_AUDIO_BIQUAD_FILTER_H + +#include "AudioCommon.h" + +namespace android { +// A biquad filter. +// Implements the recursion y[n]=a0*y[n-1]+a1*y[n-2]+b0*x[n]+b1*x[n-1]+b2*x[n-2] +// (the a0 and a1 coefficients have an opposite sign to the common convention) +// The filter works on fixed sized blocks of data (frameCount multi-channel +// samples, as defined during construction). An arbitrary number of interlaced +// channels is supported. +// Filter can operate in an enabled (active) or disabled (bypassed) states. +// A mechanism for suppression of artifacts caused by abrupt coefficient changes +// is implemented: normally, when the enable(), disable() and setCoefs() methods +// are called without the immediate flag set, the filter smoothly transitions +// from its current state to the desired state. +class AudioBiquadFilter { +public: + // Max number of channels (can be changed here, and everything should work). + static const int MAX_CHANNELS = 2; + // Number of coefficients. + static const int NUM_COEFS = 5; + + // Constructor. + // nChannels Number of input/output channels. + // sampleRate Sample rate, in Hz. + AudioBiquadFilter(int nChannels, int sampleRate); + + // Reconfiguration of the filter. Implies clear(). + // nChannels Number of input/output channels. + // sampleRate Sample rate, in Hz. + void configure(int nChannels, int sampleRate); + + // Resets the internal state of the filter. + // Coefficients are reset to identity, state becomes disabled. This change + // happens immediately and might cause discontinuities in the output. + // Delay lines are not cleared. + void reset(); + + // Clears the delay lines. + // This change happens immediately and might cause discontinuities in the + // output. + void clear(); + + // Sets the coefficients. + // If called when filter is disabled, will have no immediate effect, but the + // new coefficients will be set and used next time the filter is enabled. + // coefs The new coefficients. + // immediate If true, transitions to new coefficients smoothly, without + // introducing discontinuities in the output. Otherwise, + // transitions immediately. + void setCoefs(const audio_coef_t coefs[NUM_COEFS], bool immediate = false); + + // Process a buffer of data. Always processes frameCount multi-channel + // samples. Processing can be done in-place, by passing the same buffer as + // both arguments. + // in The input buffer. Should be of size frameCount * nChannels. + // out The output buffer. Should be of size frameCount * nChannels. + // frameCount Number of multi-channel samples to process. + void process(const audio_sample_t in[], audio_sample_t out[], + int frameCount); + + // Enables (activates) the filter. + // immediate If true, transitions to new state smoothly, without + // introducing discontinuities in the output. Otherwise, + // transitions immediately. + void enable(bool immediate = false); + + // Disables (bypasses) the filter. + // immediate If true, transitions to new state smoothly, without + // introducing discontinuities in the output. Otherwise, + // transitions immediately. + void disable(bool immediate = false); + +private: + // A prototype of the actual processing function. Has the same semantics as + // the process() method. + typedef void (AudioBiquadFilter::*process_func)(const audio_sample_t[], + audio_sample_t[], + int frameCount); + + // The maximum rate of coefficient change, measured in coefficient units per + // second. + static const audio_coef_t MAX_DELTA_PER_SEC = 2000; + + // Coefficients of identity transformation. + static const audio_coef_t IDENTITY_COEFS[NUM_COEFS]; + + // Filter state. + enum state_t { + // Bypass. + STATE_BYPASS = 0x01, + // In the process of smooth transition to bypass state. + STATE_TRANSITION_TO_BYPASS = 0x02, + // In the process of smooth transition to normal (enabled) state. + STATE_TRANSITION_TO_NORMAL = 0x04, + // In normal (enabled) state. + STATE_NORMAL = 0x05, + // A bit-mask for determining whether the filter is enabled or disabled + // in the eyes of the client. + STATE_ENABLED_MASK = 0x04 + }; + + // Number of channels. + int mNumChannels; + // Current state. + state_t mState; + // Maximum coefficient delta per sample. + audio_coef_t mMaxDelta; + + // A bit-mask designating for which coefficients the current value is not + // necessarily identical to the target value (since we're in transition + // state). + uint32_t mCoefDirtyBits; + // The current coefficients. + audio_coef_t mCoefs[NUM_COEFS]; + // The target coefficients. Will not be identical to mCoefs if we are in a + // transition state. + audio_coef_t mTargetCoefs[NUM_COEFS]; + + // The delay lines. + audio_sample_t mDelays[MAX_CHANNELS][4]; + + // Current processing function (determines according to current state and + // number of channels). + process_func mCurProcessFunc; + + // Sets a new state. Updates the processing function accordingly, and sets + // the dirty bits if changing to a transition state. + void setState(state_t state); + + // In a transition state, modifies the current coefs towards the passed + // coefs, while keeping a smooth change rate. Whenever a coef reaches its + // target value, the dirty bit is cleared. If all are clear, the function + // returns true, and we can then change to our target state. + bool updateCoefs(const audio_coef_t coefs[NUM_COEFS], int frameCount); + + // Processing function when in disabled state. + void process_bypass(const audio_sample_t * in, audio_sample_t * out, + int frameCount); + // Processing function when in normal state, mono. + void process_normal_mono(const audio_sample_t * in, audio_sample_t * out, + int frameCount); + // Processing function when transitioning to normal state, mono. + void process_transition_normal_mono(const audio_sample_t * in, + audio_sample_t * out, int frameCount); + // Processing function when transitioning to bypass state, mono. + void process_transition_bypass_mono(const audio_sample_t * in, + audio_sample_t * out, int frameCount); + // Processing function when in normal state, multi-channel. + void process_normal_multi(const audio_sample_t * in, audio_sample_t * out, + int frameCount); + // Processing function when transitioning to normal state, multi-channel. + void process_transition_normal_multi(const audio_sample_t * in, + audio_sample_t * out, int frameCount); + // Processing function when transitioning to bypass state, multi-channel. + void process_transition_bypass_multi(const audio_sample_t * in, + audio_sample_t * out, int frameCount); +}; +} + +#endif // ANDROID_AUDIO_BIQUAD_FILTER_H |