Acoustic Echo cancelllation for DM365

I was able to make the AEC working.

1) add this to the cfg file

var AEC = xdc.useModule('ittiam.codecs.aec.ce.AEC');

var Engine = xdc.useModule('ti.sdo.ce.Engine');
var myEngine = Engine.create("enginename", [
    {name: "jpegenc", mod: JPEGENC, local: true, groupId: 1},  
    {name: "jpegdec", mod: JPEGDEC, local: true, groupId: 1}, 

....
    {name: "aec",     mod: AEC, local: true, groupId: 1}, 
]);
algSettings = xdc.useModule('ti.sdo.ce.alg.Settings');
algSettings.useCache = true;

2) Follow the aec_ce_sapp.c in the aec_app directory.

I splitted the processing function into 3 subfunctions:

- aecInit()

- aecProcess()   which takes far end and near end signals and gives an updated far end and the residual.

- aecClose()

The AEC is working for sure (mostly in half duplex, maybe because I have the ERL too low). I have to perform further tests to evaluate the performances of the algorithms.

8360.aaec.rar

hi,

I'm trying to add AEC to dmai, see attached files.

and I use it like below:

##########################################################################################

Creation:

 AAec_Handle hAec    = NULL;
 AEC_ITTIAM_Params    AecParams;
 AEC_ITTIAM_DynamicParams AecDynParams;
 
 Buffer_Handle hFarEndBuf   = NULL;
 Buffer_Handle hNearEndBuf  = NULL;
 Buffer_Handle hResidueBuf  = NULL;
 Buffer_Handle hUpdatedFarEndBuf = NULL;

 if(envp->sampleRate > 16000)
 {
  __ERR("Aec only support samplerate less than 16K!!!\n");   
 }
 else
 {  
  AecParams.base.size = sizeof(AEC_ITTIAM_Params);
  AecDynParams.base.size = sizeof(AEC_ITTIAM_DynamicParams);
  
  AecParams.samp_freq    = envp->sampleRate;  /* sampling frequency for AEC in Hz  */
  AecParams.tail_length   = 2048*2;  /* X32 */
  AecParams.frame_length   = 256;//160*2;   /* X8, Number of samples to be processed  in this process call */
  AecParams.erle_thresh_level  = 36;      /* ERLE threshold for determining    */ 
  
  AecDynParams.inhibit_conv   = 0;
  AecDynParams.enable_nlp   = 1;
  AecDynParams.enable_cng   = 1;
  AecDynParams.enable_aec   = 1;  /* Enable/Disable AEC echo cancel    */
  AecDynParams.enable_tx_sat_det = 0;  /* 1:enable, 0:disable Tx saturation detector                          */
  AecDynParams.enable_rx_sat_det = 0;  /* 1:enable, 0:disable Rx saturation detector                          */
  AecDynParams.tx_sat_thresh  = 0;  /* Tx saturation detector thresh: Q0 */
  AecDynParams.rx_sat_thresh  = 0;  /* Rx saturation detector thresh: Q0 */
  AecDynParams.tx_digital_gain = 0;  /* Tx path digital gain applied within AEC, in 0.5dB steps, Q1 (dB)      */
  AecDynParams.rx_digital_gain = 0;  /* Rx path digital gain applied within AEC, in 0.5dB steps, Q1 (dB)      */
  AecDynParams.operating_mode  = 1;
   
  hAec = AAec_create(hEngine, &AecParams, &AecDynParams);
  if(hAec == NULL)
  {
   __ERR("Failed to create AEC!!!\n");   
   //cleanup(THREAD_FAILURE);
  }
  
  __ERR("create AEC OK!!!\n");
  
    hFarEndBuf = Buffer_create(AAec_getInBufSize(hAec,0), &bAttrs);
     if (hFarEndBuf == NULL)
  {
         __ERR("Failed to allocate hFarEndBuf.\n");
         cleanup(THREAD_FAILURE);
     }  
    
    hNearEndBuf = Buffer_create(AAec_getInBufSize(hAec,1), &bAttrs);
     if (hNearEndBuf == NULL)
  {
         __ERR("Failed to allocate hNearEndBuf.\n");
         cleanup(THREAD_FAILURE);
     }  
    
    hResidueBuf = Buffer_create(AAec_getOutBufSize(hAec,0), &bAttrs);
     if (hResidueBuf == NULL)
  {
         __ERR("Failed to allocate hResidueBuf.\n");
         cleanup(THREAD_FAILURE);
     }  
    
    hUpdatedFarEndBuf = Buffer_create(AAec_getOutBufSize(hAec,1), &bAttrs);
     if (hUpdatedFarEndBuf == NULL)
  {
         __ERR("Failed to allocate hUpdatedFarEndBuf.\n");
         cleanup(THREAD_FAILURE);
     }
    
     __ERR("create AEC 4 buffers OK!!!\n");             
 }

Process:

...

       /* Read samples from the Sound device */
        if (Sound_read(hSound, hInBuf) < 0)
  {
            __ERR("Failed to read speech buffer\n");
            cleanup(THREAD_FAILURE);
  }

  if(hAec)
  {
   int blksize = 256*2;
   int count = 4096/blksize;
   int i;
   struct timeval t1,t2;
   int usetime;
   
   gettimeofday(&t1,NULL);
   
   for(i=0;i<count;i++)
   {
    memcpy(Buffer_getUserPtr(hFarEndBuf), Buffer_getUserPtr(hInBuf) + i*blksize, blksize);
    memset(Buffer_getUserPtr(hNearEndBuf), 0, blksize);     //just for test, should use the playing sound buffer
    Buffer_setNumBytesUsed(hFarEndBuf, blksize);
    Buffer_setNumBytesUsed(hNearEndBuf, blksize);
        
    if(Dmai_EOK != AAec_process(hAec, hFarEndBuf, hNearEndBuf, hResidueBuf, hUpdatedFarEndBuf))
    {     
     __ERR("Failed to do AEC process!!!\n"); 
     break;
    }
    
    memcpy(Buffer_getUserPtr(hInBuf) + i*blksize, Buffer_getUserPtr(hResidueBuf), blksize);    
   }
      
   gettimeofday(&t2, NULL);   
   usetime = (t2.tv_sec - t1.tv_sec)*1000 + (t2.tv_usec - t1.tv_usec)/1000;
   //__ERR("AEC use time: %d\n", usetime);
  }

Problem:

1. can we set  AecParams.frame_length   to a bigger one such as 4096, the size of our read buffer? I tried but this will cause AAec_create() fail.

2. AEC will use a long time(more than 700ms) to handle one frame, it's unaccepable.

3. I just want to adjust the readed sound buffer and then encode it, do I need to adjust the buffer to be played?

Please give me some advice.

Thank you.

Regards,

Johnny Ling