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448 lines
18 KiB
448 lines
18 KiB
8 months ago
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/*
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AudioEffectCompressor
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Created: Chip Audette, Dec 2016 - Jan 2017
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Purpose; Apply dynamic range compression to the audio stream.
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Assumes floating-point data.
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This processes a single stream fo audio data (ie, it is mono)
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MIT License. use at your own risk.
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Stereo version - Piotr Zapart www.hexefx.com 03.2024
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*/
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#ifndef _EFFECT_COMPRESSORSTEREO_F32
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#define _EFFECT_COMPRESSORSTEREO_F32
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#include <arm_math.h> //ARM DSP extensions. https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html
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#include <AudioStream_F32.h>
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class AudioEffectCompressorStereo_F32 : public AudioStream_F32
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{
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// GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
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public:
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// constructor
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AudioEffectCompressorStereo_F32(void) : AudioStream_F32(2, inputQueueArray_f32)
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{
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setDefaultValues(AUDIO_SAMPLE_RATE);
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resetStates();
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};
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AudioEffectCompressorStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32)
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{
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setDefaultValues(settings.sample_rate_Hz);
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resetStates();
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};
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typedef enum
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{
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COMP_SIDECHAIN_SRC_LR, // l + r separate
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COMP_SIDECHAIN_SRC_LRSUM // l + r sum / 2
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}sideChainMode_t;
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void setDefaultValues(const float sample_rate_Hz)
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{
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fs_Hz = sample_rate_Hz;
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setThresh_dBFS(-20.0f); // set the default value for the threshold for compression
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setCompressionRatio(5.0f); // set the default copression ratio
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setAttack_sec(0.005f); // default to this value
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setRelease_sec(0.200f); // default to this value
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setHPFilterCoeff();
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enableHPFilter(true); // enable the HP filter to remove any DC offset from the audio
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sidechainMode = COMP_SIDECHAIN_SRC_LRSUM;
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}
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// here's the method that does all the work
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void update(void)
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{
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audio_block_f32_t *blockL, *blockR;
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if (bp) // handle bypass
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{
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blockL = AudioStream_F32::receiveReadOnly_f32(0);
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blockR = AudioStream_F32::receiveReadOnly_f32(1);
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if (!blockL || !blockR)
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{
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if (blockL) AudioStream_F32::release(blockL);
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if (blockR) AudioStream_F32::release(blockR);
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return;
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}
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AudioStream_F32::transmit(blockL, 0);
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AudioStream_F32::transmit(blockR, 1);
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AudioStream_F32::release(blockL);
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AudioStream_F32::release(blockR);
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return;
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}
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blockL = AudioStream_F32::receiveWritable_f32(0);
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blockR = AudioStream_F32::receiveWritable_f32(1);
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if (!blockL || !blockR)
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{
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if (blockL) AudioStream_F32::release(blockL);
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if (blockR) AudioStream_F32::release(blockR);
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return;
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}
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// allocate blocks required for gain calculations
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audio_block_f32_t* audio_level_dB_blockL = AudioStream_F32::allocate_f32();
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audio_block_f32_t* audio_level_dB_blockR = AudioStream_F32::allocate_f32();
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audio_block_f32_t *gain_blockL = AudioStream_F32::allocate_f32();
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audio_block_f32_t *gain_blockR = AudioStream_F32::allocate_f32();
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// no memory for the audio gain blocks
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if ( !audio_level_dB_blockL || !audio_level_dB_blockR || !gain_blockL || !gain_blockL)
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{
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if (audio_level_dB_blockL) AudioStream_F32::release(audio_level_dB_blockL);
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if (audio_level_dB_blockR) AudioStream_F32::release(audio_level_dB_blockR);
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if (gain_blockL) AudioStream_F32::release(gain_blockL);
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if (gain_blockR) AudioStream_F32::release(gain_blockR);
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AudioStream_F32::release(blockL);
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AudioStream_F32::release(blockR);
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return;
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}
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// apply a high-pass filter to get rid of the DC offset
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if (use_HP_prefilter)
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{
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arm_biquad_cascade_df1_f32(&hp_filt_structL, blockL->data, blockL->data, blockL->length);
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arm_biquad_cascade_df1_f32(&hp_filt_structR, blockR->data, blockR->data, blockR->length);
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}
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// apply the pre-gain...a negative gain value will disable
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if (pre_gain > 0.0f)
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{
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arm_scale_f32(blockL->data, pre_gain, blockL->data, blockL->length); // use ARM DSP for speed!
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arm_scale_f32(blockR->data, pre_gain, blockR->data, blockR->length);
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}
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// Side chain processing
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switch (sidechainMode)
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{
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case COMP_SIDECHAIN_SRC_LR: // l + r separate
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calcAudioLevel_dB(blockL, audio_level_dB_blockL);
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calcAudioLevel_dB(blockR, audio_level_dB_blockR);
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calcGain(audio_level_dB_blockL, gain_blockL);
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calcGain(audio_level_dB_blockR, gain_blockR);
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arm_mult_f32(blockL->data, gain_blockL->data, blockL->data, blockL->length);
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arm_mult_f32(blockR->data, gain_blockR->data, blockR->data, blockR->length);
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break;
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case COMP_SIDECHAIN_SRC_LRSUM: // l + r sum / 2
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arm_add_f32(blockL->data, blockR->data, audio_level_dB_blockL->data, audio_level_dB_blockL->length); // L+R -> db_L
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arm_scale_f32(audio_level_dB_blockL->data, 0.5f, audio_level_dB_blockL->data, audio_level_dB_blockL->length); // L+R / 2
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calcAudioLevel_dB(audio_level_dB_blockL, audio_level_dB_blockL); // chn L used for L&R
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calcGain(audio_level_dB_blockL, gain_blockL);
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arm_mult_f32(blockL->data, gain_blockL->data, blockL->data, blockL->length);
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arm_mult_f32(blockR->data, gain_blockL->data, blockR->data, blockR->length);
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break;
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default:
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break;
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}
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if (post_gain > 0.0f)
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{
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arm_scale_f32(blockL->data, post_gain, blockL->data, blockL->length); // use ARM DSP for speed!
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arm_scale_f32(blockR->data, post_gain, blockR->data, blockR->length);
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}
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// transmit the block and release memory
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AudioStream_F32::transmit(blockL, 0);
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AudioStream_F32::transmit(blockR, 1);
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AudioStream_F32::release(blockL);
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AudioStream_F32::release(blockR);
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AudioStream_F32::release(gain_blockL);
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AudioStream_F32::release(gain_blockR);
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AudioStream_F32::release(audio_level_dB_blockL);
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AudioStream_F32::release(audio_level_dB_blockR);
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}
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// Here's the method that estimates the level of the audio (in dB)
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// It squares the signal and low-pass filters to get a time-averaged
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// signal power. It then
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void calcAudioLevel_dB(audio_block_f32_t *wav_block, audio_block_f32_t *level_dB_block)
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{
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// calculate the instantaneous signal power (square the signal)
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audio_block_f32_t *wav_pow_block = AudioStream_F32::allocate_f32();
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arm_mult_f32(wav_block->data, wav_block->data, wav_pow_block->data, wav_block->length);
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// low-pass filter and convert to dB
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float c1 = level_lp_const, c2 = 1.0f - c1; // prepare constants
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for (int i = 0; i < wav_pow_block->length; i++)
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{
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// first-order low-pass filter to get a running estimate of the average power
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wav_pow_block->data[i] = c1 * prev_level_lp_pow + c2 * wav_pow_block->data[i];
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// save the state of the first-order low-pass filter
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prev_level_lp_pow = wav_pow_block->data[i];
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// now convert the signal power to dB (but not yet multiplied by 10.0)
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level_dB_block->data[i] = log10f_approx(wav_pow_block->data[i]);
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}
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// limit the amount that the state of the smoothing filter can go toward negative infinity
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if (prev_level_lp_pow < (1.0E-13))
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prev_level_lp_pow = 1.0E-13; // never go less than -130 dBFS
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// scale the wav_pow_block by 10.0 to complete the conversion to dB
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arm_scale_f32(level_dB_block->data, 10.0f, level_dB_block->data, level_dB_block->length); // use ARM DSP for speed!
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// release memory and return
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AudioStream_F32::release(wav_pow_block);
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return; // output is passed through level_dB_block
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}
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// This method computes the desired gain from the compressor, given an estimate
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// of the signal level (in dB)
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void calcGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *gain_block)
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{
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// first, calculate the instantaneous target gain based on the compression ratio
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audio_block_f32_t *inst_targ_gain_dB_block = AudioStream_F32::allocate_f32();
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calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block);
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// second, smooth in time (attack and release) by stepping through each sample
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audio_block_f32_t *gain_dB_block = AudioStream_F32::allocate_f32();
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calcSmoothedGain_dB(inst_targ_gain_dB_block, gain_dB_block);
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// finally, convert from dB to linear gain: gain = 10^(gain_dB/20); (ie this takes care of the sqrt, too!)
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arm_scale_f32(gain_dB_block->data, 1.0f / 20.0f, gain_dB_block->data, gain_dB_block->length); // divide by 20
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for (int i = 0; i < gain_dB_block->length; i++)
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gain_block->data[i] = pow10f(gain_dB_block->data[i]); // do the 10^(x)
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// release memory and return
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AudioStream_F32::release(gain_dB_block);
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AudioStream_F32::release(inst_targ_gain_dB_block);
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return; // output is passed through gain_block
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}
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// Compute the instantaneous desired gain, including the compression ratio and
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// threshold for where the comrpession kicks in
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void calcInstantaneousTargetGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *inst_targ_gain_dB_block)
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{
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// how much are we above the compression threshold?
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audio_block_f32_t *above_thresh_dB_block = AudioStream_F32::allocate_f32();
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arm_offset_f32(audio_level_dB_block->data, // CMSIS DSP for "add a constant value to all elements"
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-thresh_dBFS, // this is the value to be added
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above_thresh_dB_block->data, // this is the output
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audio_level_dB_block->length);
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// scale by the compression ratio...this is what the output level should be (this is our target level)
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arm_scale_f32(above_thresh_dB_block->data, // CMSIS DSP for "multiply all elements by a constant value"
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1.0f / comp_ratio, // this is the value to be multiplied
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inst_targ_gain_dB_block->data, // this is the output
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above_thresh_dB_block->length);
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// compute the instantaneous gain...which is the difference between the target level and the original level
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arm_sub_f32(inst_targ_gain_dB_block->data, // CMSIS DSP for "subtract two vectors element-by-element"
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above_thresh_dB_block->data, // this is the vector to be subtracted
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inst_targ_gain_dB_block->data, // this is the output
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inst_targ_gain_dB_block->length);
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// limit the target gain to attenuation only (this part of the compressor should not make things louder!)
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for (int i = 0; i < inst_targ_gain_dB_block->length; i++)
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{
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if (inst_targ_gain_dB_block->data[i] > 0.0f)
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inst_targ_gain_dB_block->data[i] = 0.0f;
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}
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// release memory before returning
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AudioStream_F32::release(above_thresh_dB_block);
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return; // output is passed through inst_targ_gain_dB_block
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}
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// this method applies the "attack" and "release" constants to smooth the
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// target gain level through time.
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void calcSmoothedGain_dB(audio_block_f32_t *inst_targ_gain_dB_block, audio_block_f32_t *gain_dB_block)
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{
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float32_t gain_dB;
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float32_t one_minus_attack_const = 1.0f - attack_const;
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float32_t one_minus_release_const = 1.0f - release_const;
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for (int i = 0; i < inst_targ_gain_dB_block->length; i++)
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{
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gain_dB = inst_targ_gain_dB_block->data[i];
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// smooth the gain using the attack or release constants
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if (gain_dB < prev_gain_dB)
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{ // are we in the attack phase?
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gain_dB_block->data[i] = attack_const * prev_gain_dB + one_minus_attack_const * gain_dB;
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}
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else
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{ // or, we're in the release phase
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gain_dB_block->data[i] = release_const * prev_gain_dB + one_minus_release_const * gain_dB;
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}
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// save value for the next time through this loop
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prev_gain_dB = gain_dB_block->data[i];
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}
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// return
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return; // the output here is gain_block
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}
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// methods to set parameters of this module
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void resetStates(void)
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{
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prev_level_lp_pow = 1.0f;
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prev_gain_dB = 0.0f;
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// initialize the HP filter. (This also resets the filter states,)
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arm_biquad_cascade_df1_init_f32(&hp_filt_structL, hp_nstages, hp_coeff, hp_stateL);
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arm_biquad_cascade_df1_init_f32(&hp_filt_structR, hp_nstages, hp_coeff, hp_stateR);
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}
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void setPreGain(float g) { pre_gain = g; }
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void setPreGain_dB(float gain_dB) { setPreGain(pow(10.0f, gain_dB / 20.0f)); }
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void setPostGain(float g) { post_gain = g; }
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void setPostGain_dB(float gain_dB) { setPostGain(pow(10.0f, gain_dB / 20.0f)); }
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void setCompressionRatio(float cr)
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{
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comp_ratio = max(0.001f, cr); // limit to positive values
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updateThresholdAndCompRatioConstants();
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}
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void setAttack_sec(float a)
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{
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attack_sec = a;
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attack_const = expf(-1.0f / (attack_sec * fs_Hz)); // expf() is much faster than exp()
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// also update the time constant for the envelope extraction
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setLevelTimeConst_sec(min(attack_sec, release_sec) / 5.0f); // make the level time-constant one-fifth the gain time constants
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}
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void setRelease_sec(float r)
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{
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release_sec = r;
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release_const = expf(-1.0f / (release_sec * fs_Hz)); // expf() is much faster than exp()
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// also update the time constant for the envelope extraction
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setLevelTimeConst_sec(min(attack_sec, release_sec) / 5.0f); // make the level time-constant one-fifth the gain time constants
|
||
|
|
}
|
||
|
|
void setLevelTimeConst_sec(float t_sec)
|
||
|
|
{
|
||
|
|
const float min_t_sec = 0.002f; // this is the minimum allowed value
|
||
|
|
level_lp_sec = max(min_t_sec, t_sec);
|
||
|
|
level_lp_const = expf(-1.0f / (level_lp_sec * fs_Hz)); // expf() is much faster than exp()
|
||
|
|
}
|
||
|
|
void setThresh_dBFS(float val)
|
||
|
|
{
|
||
|
|
thresh_dBFS = val;
|
||
|
|
setThreshPow(pow(10.0f, thresh_dBFS / 10.0f));
|
||
|
|
}
|
||
|
|
void enableHPFilter(boolean flag) { use_HP_prefilter = flag; };
|
||
|
|
|
||
|
|
// methods to return information about this module
|
||
|
|
float32_t getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain); }
|
||
|
|
float32_t getAttack_sec(void) { return attack_sec; }
|
||
|
|
float32_t getRelease_sec(void) { return release_sec; }
|
||
|
|
float32_t getLevelTimeConst_sec(void) { return level_lp_sec; }
|
||
|
|
float32_t getThresh_dBFS(void) { return thresh_dBFS; }
|
||
|
|
float32_t getCompressionRatio(void) { return comp_ratio; }
|
||
|
|
float32_t getCurrentLevel_dBFS(void) { return 10.0 * log10f_approx(prev_level_lp_pow); }
|
||
|
|
float32_t getCurrentGain_dB(void) { return prev_gain_dB; }
|
||
|
|
|
||
|
|
void setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[])
|
||
|
|
{
|
||
|
|
// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
|
||
|
|
// Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
|
||
|
|
hp_coeff[0] = b[0];
|
||
|
|
hp_coeff[1] = b[1];
|
||
|
|
hp_coeff[2] = b[2]; // here are the matlab "b" coefficients
|
||
|
|
hp_coeff[3] = -a[1];
|
||
|
|
hp_coeff[4] = -a[2]; // the DSP needs the "a" terms to have opposite sign vs Matlab
|
||
|
|
}
|
||
|
|
bool bypass_get(void) {return bp;}
|
||
|
|
void bypass_set(bool state) {bp = state;}
|
||
|
|
bool bypass_tgl(void)
|
||
|
|
{
|
||
|
|
bp ^= 1;
|
||
|
|
return bp;
|
||
|
|
}
|
||
|
|
void setSideChainMode(sideChainMode_t newMode) {sidechainMode = newMode;}
|
||
|
|
private:
|
||
|
|
// state-related variables
|
||
|
|
audio_block_f32_t *inputQueueArray_f32[2]; // memory pointer for the input to this module
|
||
|
|
float32_t prev_level_lp_pow = 1.0f;
|
||
|
|
float32_t prev_gain_dB = 0.0f; // last gain^2 used
|
||
|
|
float32_t fs_Hz = AUDIO_SAMPLE_RATE_EXACT;
|
||
|
|
bool bp = true; // bypass flag
|
||
|
|
sideChainMode_t sidechainMode = COMP_SIDECHAIN_SRC_LRSUM;
|
||
|
|
// HP filter state-related variables
|
||
|
|
arm_biquad_casd_df1_inst_f32 hp_filt_structL;
|
||
|
|
arm_biquad_casd_df1_inst_f32 hp_filt_structR;
|
||
|
|
static const uint8_t hp_nstages = 1;
|
||
|
|
float32_t hp_coeff[5 * hp_nstages] = {1.0f, 0.0f, 0.0f, 0.0f, 0.0f}; // no filtering. actual filter coeff set later
|
||
|
|
float32_t hp_stateL[4 * hp_nstages];
|
||
|
|
float32_t hp_stateR[4 * hp_nstages];
|
||
|
|
void setHPFilterCoeff(void)
|
||
|
|
{
|
||
|
|
// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
|
||
|
|
// Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
|
||
|
|
const float32_t b[] = {9.979871156751189e-01, -1.995974231350238e+00, 9.979871156751189e-01}; // from Matlab
|
||
|
|
const float32_t a[] = {1.000000000000000e+00, -1.995970179642828e+00, 9.959782830576472e-01}; // from Matlab
|
||
|
|
setHPFilterCoeff_N2IIR_Matlab((float32_t *)b, (float32_t *)a);
|
||
|
|
}
|
||
|
|
|
||
|
|
// private parameters related to gain calculation
|
||
|
|
float32_t attack_const, release_const, level_lp_const; // used in calcGain(). set by setAttack_sec() and setRelease_sec();
|
||
|
|
float32_t comp_ratio_const, thresh_pow_FS_wCR; // used in calcGain(); set in updateThresholdAndCompRatioConstants()
|
||
|
|
void updateThresholdAndCompRatioConstants(void)
|
||
|
|
{
|
||
|
|
comp_ratio_const = 1.0f - (1.0f / comp_ratio);
|
||
|
|
thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);
|
||
|
|
}
|
||
|
|
|
||
|
|
// settings
|
||
|
|
float32_t attack_sec = 0.002f, release_sec = 0.2f, level_lp_sec;
|
||
|
|
float32_t thresh_dBFS = 0.0f; // threshold for compression, relative to digital full scale
|
||
|
|
float32_t thresh_pow_FS = 1.0f; // same as above, but not in dB
|
||
|
|
void setThreshPow(float t_pow)
|
||
|
|
{
|
||
|
|
thresh_pow_FS = t_pow;
|
||
|
|
updateThresholdAndCompRatioConstants();
|
||
|
|
}
|
||
|
|
float32_t comp_ratio = 1.0f; // compression ratio
|
||
|
|
float32_t pre_gain = -1.0f; // gain to apply before the compression. negative value disables
|
||
|
|
float32_t post_gain = -1.0f;
|
||
|
|
boolean use_HP_prefilter;
|
||
|
|
|
||
|
|
// Accelerate the powf(10.0,x) function
|
||
|
|
static float32_t pow10f(float x)
|
||
|
|
{
|
||
|
|
// return powf(10.0f,x) //standard, but slower
|
||
|
|
return expf(2.302585092994f * x); // faster: exp(log(10.0f)*x)
|
||
|
|
}
|
||
|
|
|
||
|
|
// Accelerate the log10f(x) function?
|
||
|
|
static float32_t log10f_approx(float x)
|
||
|
|
{
|
||
|
|
// return log10f(x); //standard, but slower
|
||
|
|
return log2f_approx(x) * 0.3010299956639812f; // faster: log2(x)/log2(10)
|
||
|
|
}
|
||
|
|
|
||
|
|
/* ----------------------------------------------------------------------
|
||
|
|
** Fast approximation to the log2() function. It uses a two step
|
||
|
|
** process. First, it decomposes the floating-point number into
|
||
|
|
** a fractional component F and an exponent E. The fraction component
|
||
|
|
** is used in a polynomial approximation and then the exponent added
|
||
|
|
** to the result. A 3rd order polynomial is used and the result
|
||
|
|
** when computing db20() is accurate to 7.984884e-003 dB.
|
||
|
|
** ------------------------------------------------------------------- */
|
||
|
|
// https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
|
||
|
|
// float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
|
||
|
|
static float log2f_approx(float X)
|
||
|
|
{
|
||
|
|
// float *C = &log2f_approx_coeff[0];
|
||
|
|
float Y;
|
||
|
|
float F;
|
||
|
|
int E;
|
||
|
|
|
||
|
|
// This is the approximation to log2()
|
||
|
|
F = frexpf(fabsf(X), &E);
|
||
|
|
// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
|
||
|
|
// Y = *C++;
|
||
|
|
Y = 1.23149591368684f;
|
||
|
|
Y *= F;
|
||
|
|
// Y += (*C++);
|
||
|
|
Y += -4.11852516267426f;
|
||
|
|
Y *= F;
|
||
|
|
// Y += (*C++);
|
||
|
|
Y += 6.02197014179219f;
|
||
|
|
Y *= F;
|
||
|
|
// Y += (*C++);
|
||
|
|
Y += -3.13396450166353f;
|
||
|
|
Y += E;
|
||
|
|
|
||
|
|
return (Y);
|
||
|
|
}
|
||
|
|
};
|
||
|
|
|
||
|
|
#endif
|