annotate libao2/pl_eq.c @ 12022:293141b57c01

Use MultiplyElement to control volume. Works with multiple videos at the same time and even when NAS does not control the mixer or it is unavailable. Show buffer underrun hint only once and add missing linebreaks.

/*=============================================================================
//	
//  This software has been released under the terms of the GNU Public
//  license. See http://www.gnu.org/copyleft/gpl.html for details.
//
//  Copyright 2001 Anders Johansson [e-mail]
//
//=============================================================================
*/

/* Equalizer plugin, implementation of a 10 band time domain graphic
   equalizer using IIR filters. The IIR filters are implemented using a
   Direct Form II approach. But has been modified (b1 == 0 always) to
   save computation.
*/
#define PLUGIN

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>
#include <math.h>

#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
#include "eq.h"

static ao_info_t info =
{
        "Equalizer audio plugin",
        "eq",
        "Anders",
        ""
};

LIBAO_PLUGIN_EXTERN(eq)


#define CH  6     // Max number of channels
#define L   2     // Storage for filter taps
#define KM  10    // Max number of octaves

#define Q   1.2247 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
		      gives 4dB suppression @ Fc*2 and Fc/2 */

// Center frequencies for band-pass filters
#define CF  {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}

// local data
typedef struct pl_eq_s
{
  int16_t   a[KM][L];        // A weights
  int16_t   b[KM][L];        // B weights
  int16_t   wq[CH][KM][L];   // Circular buffer for W data
  int16_t   g[CH][KM];       // Gain factor for each channel and band
  int16_t   K; 		     // Number of used eq bands
  int       channels;        // Number of channels
} pl_eq_t;

static pl_eq_t pl_eq;

// to set/get/query special features/parameters
static int control(int cmd,void *arg){
  switch(cmd){
  case AOCONTROL_PLUGIN_SET_LEN:
    return CONTROL_OK;
  case AOCONTROL_PLUGIN_EQ_SET_GAIN:{
    float gain = ((equalizer_t*)arg)->gain;
    int ch     =((equalizer_t*)arg)->channel;
    int band   =((equalizer_t*)arg)->band;
    if(ch > CH || ch < 0 || band > KM || band < 0)
      return CONTROL_ERROR;
    
    pl_eq.g[ch][band]=(int16_t) 4096 * (pow(10.0,gain/20.0)-1.0);
    return CONTROL_OK;
  }
  case AOCONTROL_PLUGIN_EQ_GET_GAIN:{
    int ch     =((equalizer_t*)arg)->channel;
    int band   =((equalizer_t*)arg)->band;
    if(ch > CH || ch < 0 || band > KM || band < 0)
      return CONTROL_ERROR;
    
    ((equalizer_t*)arg)->gain = log10((float)pl_eq.g[ch][band]/4096.0+1) * 20.0;
    return CONTROL_OK;
  }
  }
  return CONTROL_UNKNOWN;
}

// return rounded 16bit int
static inline int16_t lround16(double n){
  return (int16_t)((n)>=0.0?(n)+0.5:(n)-0.5);
}

// 2nd order Band-pass Filter design
void bp2(int16_t* a, int16_t* b, float fc, float q){
  double th=2*3.141592654*fc;
  double C=(1 - tan(th*q/2))/(1 + tan(th*q/2));
  
  a[0] = lround16( 16383.0 * (1 + C) * cos(th));
  a[1] = lround16(-16383.0 * C);
  
  b[0] = lround16(-16383.0 * (C - 1)/2);
  b[1] = lround16(-16383.0 * 1.0050);
}

// empty buffers
static void reset(){
  int k,l,c;
  for(c=0;c<pl_eq.channels;c++)
    for(k=0;k<pl_eq.K;k++)
      for(l=0;l<L*2;l++)
	pl_eq.wq[c][k][l]=0;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(){
  int   k   = 0;
  float F[KM] = CF;
  
  // Check input format
  if(ao_plugin_data.format != AFMT_S16_NE){
    fprintf(stderr,"[pl_eq] Input audio format not yet supported. \n");
    return 0;
  }

  // Check number of channels
  if(ao_plugin_data.channels>CH){
     fprintf(stderr,"[pl_eq] Too many channels, max is 6.\n");
    return 0;
  }
  pl_eq.channels=ao_plugin_data.channels;

  // Calculate number of active filters
  pl_eq.K=KM;
  while(F[pl_eq.K-1] > (float)ao_plugin_data.rate/2)
    pl_eq.K--;

  // Generate filter taps
  for(k=0;k<pl_eq.K;k++)
    bp2(pl_eq.a[k],pl_eq.b[k],F[k]/((float)ao_plugin_data.rate),Q);

  // Reset buffers
  reset();

  // Tell ao_plugin how much this plugin adds to the overall time delay
  ao_plugin_data.delay_fix-=2/((float)pl_eq.channels*(float)ao_plugin_data.rate);
  // Print some cool remark of what the plugin does
  printf("[pl_eq] Equalizer in use.\n");
  return 1;
}

// close plugin
static void uninit(){
}

// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
  uint16_t  	ci    	= pl_eq.channels; 	// Index for channels
  uint16_t	nch   	= pl_eq.channels;   	// Number of channels

  while(ci--){
    int16_t*	g	= pl_eq.g[ci]; 	// Gain factor 
    int16_t*	in    	= ((int16_t*)ao_plugin_data.data)+ci;
    int16_t*	out   	= ((int16_t*)ao_plugin_data.data)+ci;
    int16_t* 	end   	= in+ao_plugin_data.len/2; // Block loop end

    while(in < end){
      register int16_t k   = 0;	   	// Frequency band index
      register int32_t yt  = 0;    	// Total output from filters
      register int16_t x   = *in; 	// Current input sample
      in+=nch;

      // Run the filters 
      for(;k<pl_eq.K;k++){
	// Pointer to circular buffer wq
	register int16_t* wq = pl_eq.wq[ci][k];
#if 0
	// Calculate output from AR part of current filter
	register int32_t xt = (x*pl_eq.b[k][0]) >> 4;
	register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1];
	// Calculate output form MA part of current filter
	yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12;
	// Update circular buffer
	wq[1] = wq[0]; wq[0] = w >> 14;
      }	

      // Calculate output 
      *out=(int16_t)(yt+x);
#else
	// Calculate output from AR part of current filter
	register int32_t xt = (x*pl_eq.b[k][0]) / 48;
	register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1];
	// Calculate output form MA part of current filter
	yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12;
	// Update circular buffer
	wq[1] = wq[0]; wq[0] = w / 24576;
      }	

      // Calculate output 
      *out=(int16_t)(yt * 0.25 + x * 0.5);
#endif
      out+=nch;
    }
  }
  return 1;
}