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@ -51,7 +51,7 @@ void AudioFilterConvolution_F32::impulse(float32_t *FIR_coef) |
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arm_cfft_f32( &arm_cfft_sR_f32_len1024, FIR_filter_mask, 0, 1); |
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// for 1st time thru, zero out the last sample buffer to 0
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arm_fill_f32(0, last_sample_buffer_L, BUFFER_SIZE *4); |
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arm_fill_f32(0, last_sample_buffer_L, 128*4); |
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state = 0; |
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enabled = 1; //enable audio stream again
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@ -79,9 +79,9 @@ void AudioFilterConvolution_F32::update(void) |
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l++; |
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} |
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} |
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arm_copy_f32 (&buffer[0], &tbuffer[0], BUFFER_SIZE*4); |
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arm_copy_f32 (&buffer[0], &tbuffer[0], 128*4); |
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bp = block->data; |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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for (int i = 0; i < 128; i++) { |
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buffer[i] = *bp; |
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*bp++ = tbuffer[i]; |
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} |
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@ -92,7 +92,7 @@ void AudioFilterConvolution_F32::update(void) |
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case 1: |
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bp = block->data; |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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for (int i = 0; i < 128; i++) { |
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buffer[128+i] = *bp; |
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*bp++ = tbuffer[i+128]; |
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} |
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@ -103,7 +103,7 @@ void AudioFilterConvolution_F32::update(void) |
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case 2: |
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bp = block->data; |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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for (int i = 0; i < 128; i++) { |
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buffer[256 + i] = *bp; |
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*bp++ = tbuffer[i+256]; // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
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} |
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@ -117,7 +117,7 @@ void AudioFilterConvolution_F32::update(void) |
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case 3: |
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bp = block->data; |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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for (int i = 0; i < 128; i++) { |
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buffer[384 + i] = *bp; |
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*bp++ = tbuffer[i + 384]; // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
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} |
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@ -165,7 +165,7 @@ float AudioFilterConvolution_F32::m_sinc(int m, float fc) |
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{ // fc is f_cut/(Fsamp/2)
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// m is between -M and M step 2
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//
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float x = m*PIH; |
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float x = m*PIH_F32; |
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if(m == 0) |
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return 1.0f; |
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else |
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@ -229,8 +229,8 @@ void AudioFilterConvolution_F32::calc_FIR_coeffs |
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if(2*ii==nc) continue; |
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float x =(float)(2*ii - nc)/(float)nc; |
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float w = Izero(Beta*sqrtf(1.0f - x*x))/izb; // Kaiser window
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coeffs[ii-1] = 1.0f/(PIH*(float)(ii-nc/2)) * w ; |
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//coeffs[2*ii+1] = 1.0f/(PIH*(float)(ii-nc/2)) * w ;
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coeffs[ii-1] = 1.0f/(PIH_F32*(float)(ii-nc/2)) * w ; |
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//coeffs[2*ii+1] = 1.0f/(PIH_F32*(float)(ii-nc/2)) * w ;
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} |
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return; // From Hilbert design
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} |
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@ -249,11 +249,11 @@ void AudioFilterConvolution_F32::calc_FIR_coeffs |
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} |
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else if (type==BANDPASS) |
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{ |
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for(jj=0; jj< nc+1; jj++) coeffs[jj] *= 2.0f*cosf(PIH*(2*jj-nc)*fc); |
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for(jj=0; jj< nc+1; jj++) coeffs[jj] *= 2.0f*cosf(PIH_F32*(2*jj-nc)*fc); |
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} |
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else if (type==BANDREJECT) |
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{ |
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for(jj=0; jj< nc+1; jj++) coeffs[jj] *= -2.0f*cosf(PIH*(2*jj-nc)*fc); |
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for(jj=0; jj< nc+1; jj++) coeffs[jj] *= -2.0f*cosf(PIH_F32*(2*jj-nc)*fc); |
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coeffs[nc/2] += 1; |
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} |
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} // END calc_FIR_coef
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