view libao2/pl_delay.c @ 8576:ed132c268686

- Fix 8-bit sound in arts audio driver (logic in OBTAIN_BITRATE macro was broken) - arts server always expects 16-bit sound in little endian byte order, even in the case the artsd server runs on a big endian machine. Make sure that mplayer's audio filters convert the samples into one of the arts supported sound formats.

/* Audio out plugin it doesnt't really do anything useful but serves
   an example of how audio plugins work. It delays the output signal
   by the nuber of samples set by delay=n where n is the number of
   bytes.  
*/
#define PLUGIN

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

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

static ao_info_t info =
{
        "Delay audio plugin",
        "delay",
        "Anders",
        ""
};

LIBAO_PLUGIN_EXTERN(delay)

// local data
typedef struct pl_delay_s
{
  void* data;       // local audio data block
  void* delay; 	    // data block used for delaying audio signal
  int len;          // local buffer length
  int rate;         // local data rate
  int channels;     // local number of channels
  int format;       // local format
} pl_delay_t;

static pl_delay_t pl_delay={NULL,NULL,0,0,0,0};

// to set/get/query special features/parameters
static int control(int cmd,int arg){
  switch(cmd){
  case AOCONTROL_PLUGIN_SET_LEN:
    if(pl_delay.data) 
      free(pl_delay.data);
    pl_delay.len = ao_plugin_data.len;
    pl_delay.data=(void*)malloc(ao_plugin_data.len);
    if(!pl_delay.data)
      return CONTROL_ERROR;
    return CONTROL_OK;
  }
  return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(){
  float time_delay; // The time in [s] this plugin delays the output data
  
  /* If the output format of any of the below parameters differs from
     what is give it should be changed. See ao_plugin init() */
  pl_delay.rate=ao_plugin_data.rate;
  pl_delay.channels=ao_plugin_data.channels; //1=mono 2=stereo
  pl_delay.format=ao_plugin_data.format;

  // Tell ao_plugin how much this plugin adds to the overall time delay
  time_delay=-1*(float)ao_plugin_cfg.pl_delay_len/((float)pl_delay.channels*(float)pl_delay.rate);
  if(pl_delay.format != AFMT_U8 && pl_delay.format != AFMT_S8)
    time_delay/=2;
  ao_plugin_data.delay_fix+=time_delay;

  // Create buffer for the delayed data
  pl_delay.delay=(void*)malloc(ao_plugin_cfg.pl_delay_len);
  if(!pl_delay.delay)
    return 0;
  memset(pl_delay.delay, 0, ao_plugin_cfg.pl_delay_len);

  // Print some cool remark of what the plugin does
  printf("[pl_delay] Output sound delayed by %i bytes\n",ao_plugin_cfg.pl_delay_len);
  return 1;
}

// close plugin
static void uninit(){
  if(pl_delay.delay) 
    free(pl_delay.delay);
  if(pl_delay.data) 
    free(pl_delay.data);
  ao_plugin_cfg.pl_delay_len=0;
}

// empty buffers
static void reset(){
  int i = 0;
  for(i=0;i<pl_delay.len;i++)
    ((char*)pl_delay.data)[i]=0;
  for(i=0;i<ao_plugin_cfg.pl_delay_len;i++)
    ((char*)pl_delay.delay)[i]=0;
}

// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
  // Copy end of prev block to begining of buffer
  memcpy(pl_delay.data,pl_delay.delay,ao_plugin_cfg.pl_delay_len);
  // Copy current block except end
  memcpy(pl_delay.data+ao_plugin_cfg.pl_delay_len,
  	 ao_plugin_data.data,
  	 ao_plugin_data.len-ao_plugin_cfg.pl_delay_len);
  // Save away end of current block for next call
  memcpy(pl_delay.delay,
	 ao_plugin_data.data+ao_plugin_data.len-ao_plugin_cfg.pl_delay_len,
	 ao_plugin_cfg.pl_delay_len);
  // Set output data block
  ao_plugin_data.data=pl_delay.data;
  return 1;
}