update()

void erisAudioAnalyzeFFT1024::update ( void  )

virtual

Implements AudioStream.

Definition at line 178 of file eris_analyze_fft1024.cpp.

{
    audio_block_t *block;
 
    block = receiveReadOnly();
    if (!block) return;
 
 
    //FAT Audio - add the ability to turn FFT on or OFF to save CPU
    if (!enabled){
        release(block);
        return;
    };
 
    uint16_t ofs;
    //calcuate active subsampling step and offset
    if (ssr == SS_LOWFREQ){
        MEM_STEP = subsample_lowfreqrange;
        subsample_by = (int)subsample_lowfreqrange;
    }
    else if (ssr == SS_HIGHFREQ){
        MEM_STEP = subsample_highfreqrange;
        subsample_by = (int)subsample_highfreqrange;
    }
    BLOCKS_PER_FFT = ((1024 / AUDIO_BLOCK_SAMPLES) * subsample_by);
    BLOCK_REFRESH_SIZE = BLOCKS_PER_FFT;
    if (ssr == SS_LOWFREQ) BLOCK_REFRESH_SIZE = BLOCKS_PER_FFT/2;//((subsample_lowfreqrange/subsample_highfreqrange) * 2);//BLOCK_REFRESH_SIZE/(subsample_lowfreqrange/2);//BLOCKS_PER_FFT/4;
 
    ofs = (AUDIO_BLOCK_SAMPLES/subsample_by) * (sample_block);
 
    if (sample_block < BLOCKS_PER_FFT -1){
        copy_to_fft_buffer(buffer+ofs, block->data,subsample_by);
        release(block);
        sample_block++;
    } else{
        copy_to_fft_buffer(buffer+ofs, block->data,subsample_by);
        release(block);
        #ifdef ENABLE_F32_FFT
 
        //copy buffer while casting to float and scale to the range -1 to 1
        //(NVIC_DISABLE_IRQ(IRQ_SOFTWARE));
        if(outputflag == false && is_analyzed){
            for (int16_t i=0; i < 1024; i++){
                tmp_buffer[i] = ((float32_t)buffer[i] / (float32_t)32768.0);
                //if (SS_LOWFREQ) tmp_buffer[i] *= subsample_lowfreqrange/(float32_t)subsample_highfreqrange; //scale match the low range
                if (!std::isfinite(tmp_buffer[i])) tmp_buffer[i] = 0.0;
            }
            outputflag = true;
        }
        //(NVIC_ENABLE_IRQ(IRQ_SOFTWARE));
 
        #else
        memcpy(tmp_buffer,buffer,2048 * sizeof(int16_t));
        apply_window_to_fft_buffer(tmp_buffer, window);
        arm_cfft_radix4_q15(&fft_inst, tmp_buffer);
        outputflag = false; //output flag is false while updating the fft results
        
        for (int i=0; i < 512; i++) {
            uint32_t tmp = *((uint32_t *)tmp_buffer + i); // real & imag
            uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
            output[i] = sqrt_uint32_approx(magsq);
            output_packed[i] = tmp;//normalized_atan2((float)real,(float)imag)*180/PI;//atan2(imag,real)*180/PI;
        }
        #endif
 
        if (sample_block!= 0){
            sample_block = BLOCKS_PER_FFT - BLOCK_REFRESH_SIZE;
            ofs = (AUDIO_BLOCK_SAMPLES/subsample_by) * sample_block;
            //fft overlap - restore tmp cpy of last half to first half
            memmove(buffer,&buffer[(AUDIO_BLOCK_SAMPLES/subsample_by)*BLOCK_REFRESH_SIZE], (AUDIO_BLOCK_SAMPLES/subsample_by) * (BLOCKS_PER_FFT - BLOCK_REFRESH_SIZE) * sizeof(int16_t));
        }
 
    }
}

References apply_window_to_fft_buffer(), BLOCK_REFRESH_SIZE, BLOCKS_PER_FFT, copy_to_fft_buffer(), audio_block_struct::data, enabled, fft_inst, is_analyzed, MEM_STEP, outputflag, AudioStream::receiveReadOnly(), AudioStream::release(), sample_block, SS_HIGHFREQ, SS_LOWFREQ, ssr, subsample_by, subsample_highfreqrange, subsample_lowfreqrange, and window.

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