Tool/software: Code Composer Studio
Hello.
I'm studying ANC algorithm from LG Electronics, and I'm using TMS320C6747/OMAP-L137 DSP STARTER KIT.
First, I downloaded OMAPL137 RTOS SDK from http://software-dl.ti.com/processor-sdk-rtos/esd/OMAPL137/latest/index_FDS.html.
After that I used mcasp_test.c example code from C:\ti\pdk_omapl137_1_0_10\packages\ti\board\diag\mcasp\src\mcasp_test.c
I am trying to implement a FIR filter(low pass) on a TMS320C6747
Please see my code below.
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short SampleCh0;
short SampleCh1;
float RegX1 = 0.0;
float RegX2 = 0.0;
float RegX3 = 0.0;
float RegX4 = 0.0;
float RegX5 = 0.0;
float RegX6 = 0.0;
float RegX7 = 0.0;
float RegX8 = 0.0;
float RegX9 = 0.0;
float RegX1_1 = 0.0;
float RegX2_1 = 0.0;
float RegX3_1 = 0.0;
float RegX4_1 = 0.0;
float RegX5_1 = 0.0;
float RegX6_1 = 0.0;
float RegX7_1 = 0.0;
float RegX8_1 = 0.0;
float RegX9_1 = 0.0;
float b[10] = {FIR filter coefficients};
......
/*
** Loop forever. if a new buffer is received, the lastFullRxBuf will be
** updated in the rx completion ISR. if it is not the lastSentTxBuf,
** buffer is to be sent. This has to be mapped to proper paRAM set.
*/
while(1)
{
if(lastFullRxBuf != lastSentTxBuf)
{
// if(sampleCnt > TEST_SAMPLE_COUNT)
// {
// break;
// }
/*
** Start the transmission from the link paramset. The param set
** 1 will be linked to param set at PAR_TX_START. So do not
** update paRAM set1.
*/
parToSend = PAR_TX_START + (parOffTxToSend % NUM_PAR);
parOffTxToSend = (parOffTxToSend + 1) % NUM_PAR;
parToLink = PAR_TX_START + parOffTxToSend;
lastSentTxBuf = (lastSentTxBuf + 1) % NUM_BUF;
secondLastSentTxBuf = (secondLastSentTxBuf + 1) % NUM_BUF;
SampleCh0=*((void *)rxBufPtr[lastFullRxBuf] + 2 * i );
SampleCh1=*((void *)rxBufPtr[lastFullRxBuf] + 2 * i + 1 );
/* FIR filter processing */
for(i = 0; i < NUM_SAMPLES_PER_AUDIO_BUF; i++)
{
SampleCh0_1 = b[0]*SampleCh0 + b[1]*RegX1 + b[2]*RegX2 + b[3]*RegX3 + b[4]*RegX4 + b[5]*RegX5 + b[6]*RegX6 + b[7]*RegX7 + b[8]*RegX8 + b[9]*RegX9;
SampleCh1_1 = b[0]*SampleCh1 + b[1]*RegX1_1 + b[2]*RegX2_1 + b[3]*RegX3_1 + b[4]*RegX4_1 + b[5]*RegX5_1 + b[6]*RegX6_1 + b[7]*RegX7_1 + b[8]*RegX8_1 + b[9]*RegX9_1;
RegX9=RegX8;
RegX8=RegX7;
RegX7=RegX6;
RegX6=RegX5;
RegX5=RegX4;
RegX4=RegX3;
RegX3=RegX2;
RegX2=RegX1;
RegX1=SampleCh0;
RegX9_1=RegX8_1;
RegX8_1=RegX7_1;
RegX7_1=RegX6_1;
RegX6_1=RegX5_1;
RegX5_1=RegX4_1;
RegX4_1=RegX3_1;
RegX3_1=RegX2_1;
RegX2_1=RegX1_1;
RegX1_1=SampleCh1;
}
*((void*)txBufPtr[lastSentTxBuf] + 2 * i) = SampleCh0_1;
*((void*)txBufPtr[lastSentTxBuf] + 2 * i + 1) = SampleCh1_1;
BoardDiag_BufferTxDMAActivate(lastSentTxBuf, NUM_SAMPLES_PER_AUDIO_BUF,
(unsigned short)parToSend,
(unsigned short)parToLink);
sampleCnt += NUM_SAMPLES_PER_AUDIO_BUF;
}
}
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I separated audio input two channels (SampleCh0, SampleCh1), and applied the FIR filter algorithm.
Please let me know my code is right or not.
PS. I also tried another FIR filtering code example from https://e2e.ti.com/support/processors/f/791/t/409101, but I just heard noise with original input signal from output.
If you have any FIR filter, LMS, filtered-x LMS alogrithm code example, please let me know.
Any help is much appreciated. Thanks.
