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@ -16,6 +16,47 @@ |
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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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// //// Accelerate the powf(10.0,x) function???
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//#define POW10_FUNC(x) powf(10.0f,x) //standard, but slower
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#define POW10_FUNC(x) expf(2.302585092994f*x) //faster: exp(log(10.0f)*x)
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// //// Accelerate the log10f(x) function?
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//#define LOG10_FUNC(x) log10f(x) //standard, but slower
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#define LOG10_FUNC(x) log2f_approx(x)*0.3010299956639812f; //faster: log2(x)/log2(10)
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//https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
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/* ----------------------------------------------------------------------
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** Fast approximation to the log2() function. It uses a two step |
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** process. First, it decomposes the floating-point number into |
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** a fractional component F and an exponent E. The fraction component |
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** is used in a polynomial approximation and then the exponent added |
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** to the result. A 3rd order polynomial is used and the result |
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** when computing db20() is accurate to 7.984884e-003 dB. |
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** ------------------------------------------------------------------- */ |
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float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f}; |
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float log2f_approx(float X) { |
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float *C = &log2f_approx_coeff[0]; |
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float Y; |
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float F; |
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int E; |
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// This is the approximation to log2()
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F = frexpf(fabsf(X), &E); |
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// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
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Y = *C++; |
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Y *= F; |
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Y += (*C++); |
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Y *= F; |
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Y += (*C++); |
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Y *= F; |
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Y += (*C++); |
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Y += E; |
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return(Y); |
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} |
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class AudioEffectCompressor_F32 : public AudioStream_F32 |
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{ |
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public: |
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@ -78,7 +119,7 @@ class AudioEffectCompressor_F32 : public AudioStream_F32 |
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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(wav_pow_block->data[i]); |
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level_dB_block->data[i] = LOG10_FUNC(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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@ -105,8 +146,9 @@ class AudioEffectCompressor_F32 : public AudioStream_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++) gain_block->data[i] = powf(10.0f,gain_dB_block->data[i]); //do the 10^(x)
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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++) gain_block->data[i] = POW10_FUNC(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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