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303 lines
14 KiB
303 lines
14 KiB
/**************************************************************************//**
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* @file
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* @author Steve Lascos
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* @company Blackaddr Audio
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*
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* LibBasicFunctions is a collection of helpful functions and classes that make
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* it easier to perform common tasks in Audio applications.
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*
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* @copyright This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.*
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*****************************************************************************/
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#include <cstddef>
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#include <new>
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#include <arm_math.h>
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#include "Arduino.h"
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#include "Audio.h"
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#include "BATypes.h"
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#include "LibMemoryManagement.h"
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#ifndef __BAGUITAR_LIBBASICFUNCTIONS_H
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#define __BAGUITAR_LIBBASICFUNCTIONS_H
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namespace BAGuitar {
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/**************************************************************************//**
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* QueuePosition is used for storing the index (in an array of queues) and the
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* offset within an audio_block_t data buffer. Useful for dealing with large
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* windows of audio spread across multiple audio data blocks.
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*****************************************************************************/
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struct QueuePosition {
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int offset; ///< offset in samples within an audio_block_t data buffer
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int index; ///< index in an array of audio data blocks
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};
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/// Calculate the exact sample position in an array of audio blocks that corresponds
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/// to a particular offset given as time.
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/// @param milliseconds length of the interval in milliseconds
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/// @returns a struct containing the index and offset
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QueuePosition calcQueuePosition(float milliseconds);
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/// Calculate the exact sample position in an array of audio blocks that corresponds
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/// to a particular offset given as a number of samples
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/// @param milliseconds length of the interval in milliseconds
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/// @returns a struct containing the index and offset
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QueuePosition calcQueuePosition(size_t numSamples);
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/// Calculate the number of audio samples (rounded up) that correspond to a
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/// given length of time.
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/// @param milliseconds length of the interval in milliseconds
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/// @returns the number of corresonding audio samples.
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size_t calcAudioSamples(float milliseconds);
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/// Calculate a length of time in milliseconds from the number of audio samples.
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/// @param numSamples Number of audio samples to convert to time
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/// @return the equivalent time in milliseconds.
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float calcAudioTimeMs(size_t numSamples);
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/// Calculate the number of audio samples (usually an offset) from
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/// a queue position.
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/// @param position specifies the index and offset within a queue
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/// @returns the number of samples from the start of the queue array to the
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/// specified position.
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size_t calcOffset(QueuePosition position);
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/// Clear the contents of an audio block to zero
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/// @param block pointer to the audio block to clear
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void clearAudioBlock(audio_block_t *block);
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/// Perform an alpha blend between to audio blocks. Performs <br>
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/// out = dry*(1-mix) + wet*(mix)
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/// @param out pointer to the destination audio block
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/// @param dry pointer to the dry audio
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/// @param wet pointer to the wet audio
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/// @param mix float between 0.0 and 1.0.
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void alphaBlend(audio_block_t *out, audio_block_t *dry, audio_block_t* wet, float mix);
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/// Applies a gain to the audio via fixed-point scaling accoring to <br>
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/// out = int * (vol * 2^coeffShift)
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/// @param out pointer to output audio block
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/// @param in pointer to input audio block
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/// @param vol volume cofficient between -1.0 and +1.0
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/// @param coeffShift number of bits to shiftt the coefficient
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void gainAdjust(audio_block_t *out, audio_block_t *in, float vol, int coeffShift = 0);
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template <class T>
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class RingBuffer; // forward declare so AudioDelay can use it.
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/**************************************************************************//**
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* Audio delays are a very common function in audio processing. In addition to
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* being used for simply create a delay effect, it can also be used for buffering
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* a sliding window in time of audio samples. This is useful when combining
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* several audio_block_t data buffers together to form one large buffer for
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* FFTs, etc.
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* @details The buffer works like a queue. You add new audio_block_t when available,
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* and the class will return an old buffer when it is to be discarded from the queue.<br>
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* Note that using INTERNAL memory means the class will only store a queue
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* of pointers to audio_block_t buffers, since the Teensy Audio uses a shared memory
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* approach. When using EXTERNAL memory, data is actually copyied to/from an external
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* SRAM device.
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*****************************************************************************/
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constexpr size_t AUDIO_BLOCK_SIZE = sizeof(int16_t)*AUDIO_BLOCK_SAMPLES;
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class AudioDelay {
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public:
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AudioDelay() = delete;
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/// Construct an audio buffer using INTERNAL memory by specifying the max number
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/// of audio samples you will want.
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/// @param maxSamples equal or greater than your longest delay requirement
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AudioDelay(size_t maxSamples);
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/// Construct an audio buffer using INTERNAL memory by specifying the max amount of
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/// time you will want available in the buffer.
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/// @param maxDelayTimeMs max length of time you want in the buffer specified in milliseconds
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AudioDelay(float maxDelayTimeMs);
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/// Construct an audio buffer using a slot configured with the BAGuitar::ExternalSramManager
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/// @param slot a pointer to the slot representing the memory you wish to use for the buffer.
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AudioDelay(ExtMemSlot *slot);
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~AudioDelay();
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/// Add a new audio block into the buffer. When the buffer is filled,
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/// adding a new block will push out the oldest once which is returned.
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/// @param blockIn pointer to the most recent block of audio
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/// @returns the buffer to be discarded, or nullptr if not filled (INTERNAL), or
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/// not applicable (EXTERNAL).
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audio_block_t *addBlock(audio_block_t *blockIn);
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/// When using INTERNAL memory, returns the pointer for the specified index into buffer.
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/// @details, the most recent block is 0, 2nd most recent is 1, ..., etc.
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/// @param index the specifies how many buffers older than the current to retrieve
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/// @returns a pointer to the requested audio_block_t
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audio_block_t *getBlock(size_t index);
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/// Retrieve an audio block (or samples) from the buffer.
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/// @details when using INTERNAL memory, only supported size is AUDIO_BLOCK_SAMPLES. When using
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/// EXTERNAL, a size smaller than AUDIO_BLOCK_SAMPLES can be requested.
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/// @param dest pointer to the target audio block to write the samples to.
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/// @param offsetSamples data will start being transferred offset samples from the start of the audio buffer
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/// @param numSamples default value is AUDIO_BLOCK_SAMPLES, so typically you don't have to specify this parameter.
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/// @returns true on success, false on error.
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bool getSamples(audio_block_t *dest, size_t offsetSamples, size_t numSamples = AUDIO_BLOCK_SAMPLES);
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/// When using EXTERNAL memory, this function can return a pointer to the underlying ExtMemSlot object associated
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/// with the buffer.
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/// @returns pointer to the underlying ExtMemSlot.
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ExtMemSlot *getSlot() const { return m_slot; }
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/// Ween using INTERNAL memory, thsi function can return a pointer to the underlying RingBuffer that contains
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/// audio_block_t * pointers.
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/// @returns pointer to the underlying RingBuffer
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RingBuffer<audio_block_t*> *getRingBuffer() const { return m_ringBuffer; }
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private:
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/// enumerates whether the underlying memory buffer uses INTERNAL or EXTERNAL memory
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enum class MemType : unsigned {
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MEM_INTERNAL = 0, ///< internal audio_block_t from the Teensy Audio Library is used
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MEM_EXTERNAL ///< external SPI based ram is used
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};
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MemType m_type; ///< when 0, INTERNAL memory, when 1, external MEMORY.
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RingBuffer<audio_block_t *> *m_ringBuffer = nullptr; ///< When using INTERNAL memory, a RingBuffer will be created.
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ExtMemSlot *m_slot = nullptr; ///< When using EXTERNAL memory, an ExtMemSlot must be provided.
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};
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/**************************************************************************//**
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* IIR BiQuad Filter - Direct Form I <br>
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* y[n] = b0 * x[n] + b1 * x[n-1] + b2 * x[n-2] + a1 * y[n-1] + a2 * y[n-2]<br>
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* Some design tools (like Matlab assume the feedback coefficients 'a' are negated. You
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* may have to negate your 'a' coefficients.
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* @details Note that the ARM CMSIS-DSP library requires an extra zero between first
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* and second 'b' coefficients. E.g. <br>
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* {b10, 0, b11, b12, a11, a12, b20, 0, b21, b22, a21, a22, ...}
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*****************************************************************************/
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class IirBiQuadFilter {
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public:
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IirBiQuadFilter() = delete;
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/// Construct a Biquad filter with specified number of stages, coefficients and scaling.
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/// @details See CMSIS-DSP documentation for more details
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/// @param numStages number of biquad stages. Each stage has 5 coefficients.
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/// @param coeffs pointer to an array of Q31 fixed-point coefficients (range -1 to +0.999...)
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/// @param coeffShift coeffs are multiplied by 2^coeffShift to support coefficient range scaling
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IirBiQuadFilter(unsigned numStages, const int32_t *coeffs, int coeffShift = 0);
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virtual ~IirBiQuadFilter();
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/// Reconfigure the filter coefficients.
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/// @details See CMSIS-DSP documentation for more details
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/// @param numStages number of biquad stages. Each stage has 5 coefficients.
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/// @param coeffs pointer to an array of Q31 fixed-point coefficients (range -1 to +0.999...)
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/// @param coeffShift coeffs are multiplied by 2^coeffShift to support coefficient range scaling
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void changeFilterCoeffs(unsigned numStages, const int32_t *coeffs, int coeffShift = 0);
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/// Process the data using the configured IIR filter
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/// @details output and input can be the same pointer if in-place modification is desired
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/// @param output pointer to where the output results will be written
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/// @param input pointer to where the input data will be read from
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/// @param numSampmles number of samples to process
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bool process(int16_t *output, int16_t *input, size_t numSamples);
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private:
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const unsigned NUM_STAGES;
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int32_t *m_coeffs = nullptr;
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// ARM DSP Math library filter instance
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arm_biquad_casd_df1_inst_q31 m_iirCfg;
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int32_t *m_state = nullptr;
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};
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/**************************************************************************//**
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* A High-precision version of IirBiQuadFilter often necessary for complex, multistage
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* filters. This class uses CMSIS-DSP biquads with 64-bit internal precision instead
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* of 32-bit.
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*****************************************************************************/
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class IirBiQuadFilterHQ {
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public:
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IirBiQuadFilterHQ() = delete;
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/// Construct a Biquad filter with specified number of stages, coefficients and scaling.
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/// @details See CMSIS-DSP documentation for more details
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/// @param numStages number of biquad stages. Each stage has 5 coefficients.
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/// @param coeffs pointer to an array of Q31 fixed-point coefficients (range -1 to +0.999...)
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/// @param coeffShift coeffs are multiplied by 2^coeffShift to support coefficient range scaling
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IirBiQuadFilterHQ(unsigned maxNumStages, const int32_t *coeffs, int coeffShift = 0);
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virtual ~IirBiQuadFilterHQ();
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/// Reconfigure the filter coefficients.
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/// @details See CMSIS-DSP documentation for more details
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/// @param numStages number of biquad stages. Each stage has 5 coefficients.
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/// @param coeffs pointer to an array of Q31 fixed-point coefficients (range -1 to +0.999...)
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/// @param coeffShift coeffs are multiplied by 2^coeffShift to support coefficient range scaling
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void changeFilterCoeffs(unsigned numStages, const int32_t *coeffs, int coeffShift = 0);
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/// Process the data using the configured IIR filter
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/// @details output and input can be the same pointer if in-place modification is desired
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/// @param output pointer to where the output results will be written
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/// @param input pointer to where the input data will be read from
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/// @param numSampmles number of samples to process
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bool process(int16_t *output, int16_t *input, size_t numSamples);
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private:
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const unsigned NUM_STAGES;
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int32_t *m_coeffs = nullptr;
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// ARM DSP Math library filter instance
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arm_biquad_cas_df1_32x64_ins_q31 m_iirCfg;
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int64_t *m_state = nullptr;
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};
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/**************************************************************************//**
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* A single-precision floating-point biquad using CMSIS-DSP hardware instructions.
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* @details Use this when IirBiQuadFilterHQ is insufficient, since that version
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* is still faster with 64-bit fixed-point arithmetic.
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*****************************************************************************/
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class IirBiQuadFilterFloat {
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public:
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IirBiQuadFilterFloat() = delete;
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/// Construct a Biquad filter with specified number of stages and coefficients
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/// @details See CMSIS-DSP documentation for more details
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/// @param numStages number of biquad stages. Each stage has 5 coefficients.
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/// @param coeffs pointer to an array of single-precision floating-point coefficients
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IirBiQuadFilterFloat(unsigned maxNumStages, const float *coeffs);
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virtual ~IirBiQuadFilterFloat();
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/// Reconfigure the filter coefficients.
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/// @details See CMSIS-DSP documentation for more details
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/// @param numStages number of biquad stages. Each stage has 5 coefficients.
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/// @param coeffs pointer to an array of single-precision floating-point coefficients
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void changeFilterCoeffs(unsigned numStages, const float *coeffs);
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/// Process the data using the configured IIR filter
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/// @details output and input can be the same pointer if in-place modification is desired
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/// @param output pointer to where the output results will be written
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/// @param input pointer to where the input data will be read from
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/// @param numberSampmles number of samples to process
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bool process(float *output, float *input, size_t numSamples);
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private:
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const unsigned NUM_STAGES;
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float *m_coeffs = nullptr;
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// ARM DSP Math library filter instance
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arm_biquad_cascade_df2T_instance_f32 m_iirCfg;
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float *m_state = nullptr;
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};
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}
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#endif /* __BAGUITAR_LIBBASICFUNCTIONS_H */
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