annotate libaf/af.c @ 28615:15e7abed4291

Use the same code to convert fps in float to fraction as used in mencoder, it ensures all the common frame rates work right. If this causes issues, it should be changed in the same way in mencoder.c

/*
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

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

#if HAVE_MALLOC_H
#include <malloc.h>
#endif

#include "af.h"

// Static list of filters
extern af_info_t af_info_dummy;
extern af_info_t af_info_delay;
extern af_info_t af_info_channels;
extern af_info_t af_info_format;
extern af_info_t af_info_resample;
extern af_info_t af_info_volume;
extern af_info_t af_info_equalizer;
extern af_info_t af_info_gate;
extern af_info_t af_info_comp;
extern af_info_t af_info_pan;
extern af_info_t af_info_surround;
extern af_info_t af_info_sub;
extern af_info_t af_info_export;
extern af_info_t af_info_volnorm;
extern af_info_t af_info_extrastereo;
extern af_info_t af_info_lavcac3enc;
extern af_info_t af_info_lavcresample;
extern af_info_t af_info_sweep;
extern af_info_t af_info_hrtf;
extern af_info_t af_info_ladspa;
extern af_info_t af_info_center;
extern af_info_t af_info_sinesuppress;
extern af_info_t af_info_karaoke;
extern af_info_t af_info_scaletempo;

static af_info_t* filter_list[]={ 
   &af_info_dummy,
   &af_info_delay,
   &af_info_channels,
   &af_info_format,
   &af_info_resample,
   &af_info_volume,
   &af_info_equalizer,
   &af_info_gate,
   &af_info_comp,
   &af_info_pan,
   &af_info_surround,
   &af_info_sub,
#ifdef HAVE_SYS_MMAN_H
   &af_info_export,
#endif
   &af_info_volnorm,
   &af_info_extrastereo,
#ifdef CONFIG_LIBAVCODEC_A
   &af_info_lavcac3enc,
#endif
#ifdef CONFIG_LIBAVCODEC
   &af_info_lavcresample,
#endif
   &af_info_sweep,
   &af_info_hrtf,
#ifdef CONFIG_LADSPA
   &af_info_ladspa,
#endif
   &af_info_center,
   &af_info_sinesuppress,
   &af_info_karaoke,
   &af_info_scaletempo,
   NULL 
};

// Message printing
af_msg_cfg_t af_msg_cfg={0,NULL,NULL};

// CPU speed
int* af_cpu_speed = NULL;

/* Find a filter in the static list of filters using it's name. This
   function is used internally */
static af_info_t* af_find(char*name)
{
  int i=0;
  while(filter_list[i]){
    if(!strcmp(filter_list[i]->name,name))
      return filter_list[i];
    i++;
  }
  af_msg(AF_MSG_ERROR,"Couldn't find audio filter '%s'\n",name);
  return NULL;
} 

/* Find filter in the dynamic filter list using it's name This
   function is used for finding already initialized filters */
af_instance_t* af_get(af_stream_t* s, char* name)
{
  af_instance_t* af=s->first; 
  // Find the filter
  while(af != NULL){
    if(!strcmp(af->info->name,name))
      return af;
    af=af->next;
  }
  return NULL;
}

/*/ Function for creating a new filter of type name. The name may
  contain the commandline parameters for the filter */
static af_instance_t* af_create(af_stream_t* s, const char* name_with_cmd)
{
  char* name = strdup(name_with_cmd);
  char* cmdline = name;

  // Allocate space for the new filter and reset all pointers
  af_instance_t* new=malloc(sizeof(af_instance_t));
  if (!name || !new) {
    af_msg(AF_MSG_ERROR,"[libaf] Could not allocate memory\n");
    goto err_out;
  }  
  memset(new,0,sizeof(af_instance_t));

  // Check for commandline parameters
  strsep(&cmdline, "=");

  // Find filter from name
  if(NULL == (new->info=af_find(name)))
    goto err_out;

  /* Make sure that the filter is not already in the list if it is
     non-reentrant */
  if(new->info->flags & AF_FLAGS_NOT_REENTRANT){
    if(af_get(s,name)){
      af_msg(AF_MSG_ERROR,"[libaf] There can only be one instance of" 
	     " the filter '%s' in each stream\n",name);  
      goto err_out;
    }
  }
  
  af_msg(AF_MSG_VERBOSE,"[libaf] Adding filter %s \n",name);
  
  // Initialize the new filter
  if(AF_OK == new->info->open(new) && 
     AF_ERROR < new->control(new,AF_CONTROL_POST_CREATE,&s->cfg)){
    if(cmdline){
      if(AF_ERROR>=new->control(new,AF_CONTROL_COMMAND_LINE,cmdline))
        goto err_out;
    }
    free(name);
    return new; 
  }
  
err_out:
  free(new);
  af_msg(AF_MSG_ERROR,"[libaf] Couldn't create or open audio filter '%s'\n",
	 name);  
  free(name);
  return NULL;
}

/* Create and insert a new filter of type name before the filter in the
   argument. This function can be called during runtime, the return
   value is the new filter */
static af_instance_t* af_prepend(af_stream_t* s, af_instance_t* af, char* name)
{
  // Create the new filter and make sure it is OK
  af_instance_t* new=af_create(s,name);
  if(!new)
    return NULL;
  // Update pointers
  new->next=af;
  if(af){
    new->prev=af->prev;
    af->prev=new;
  }
  else
    s->last=new;
  if(new->prev)
    new->prev->next=new;
  else
    s->first=new;
  return new;
}

/* Create and insert a new filter of type name after the filter in the
   argument. This function can be called during runtime, the return
   value is the new filter */
static af_instance_t* af_append(af_stream_t* s, af_instance_t* af, char* name)
{
  // Create the new filter and make sure it is OK
  af_instance_t* new=af_create(s,name);
  if(!new)
    return NULL;
  // Update pointers
  new->prev=af;
  if(af){
    new->next=af->next;
    af->next=new;
  }
  else
    s->first=new;
  if(new->next)
    new->next->prev=new;
  else
    s->last=new;
  return new;
}

// Uninit and remove the filter "af"
void af_remove(af_stream_t* s, af_instance_t* af)
{
  if(!af) return;

  // Print friendly message 
  af_msg(AF_MSG_VERBOSE,"[libaf] Removing filter %s \n",af->info->name); 

  // Notify filter before changing anything
  af->control(af,AF_CONTROL_PRE_DESTROY,0);

  // Detach pointers
  if(af->prev)
    af->prev->next=af->next;
  else
    s->first=af->next;
  if(af->next)
    af->next->prev=af->prev;
  else
    s->last=af->prev;

  // Uninitialize af and free memory   
  af->uninit(af);
  free(af);
}

/* Reinitializes all filters downstream from the filter given in the
   argument the return value is AF_OK if success and AF_ERROR if
   failure */
static int af_reinit(af_stream_t* s, af_instance_t* af)
{
  do{
    af_data_t in; // Format of the input to current filter
    int rv=0; // Return value

    // Check if there are any filters left in the list
    if(NULL == af){
      if(!(af=af_append(s,s->first,"dummy"))) 
	return AF_UNKNOWN; 
      else
	return AF_ERROR;
    }

    // Check if this is the first filter 
    if(!af->prev) 
      memcpy(&in,&(s->input),sizeof(af_data_t));
    else
      memcpy(&in,af->prev->data,sizeof(af_data_t));
    // Reset just in case...
    in.audio=NULL;
    in.len=0;
    
    rv = af->control(af,AF_CONTROL_REINIT,&in);
    switch(rv){
    case AF_OK:
	af = af->next;
      break;
    case AF_FALSE:{ // Configuration filter is needed
      // Do auto insertion only if force is not specified
      if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
	af_instance_t* new = NULL;
	// Insert channels filter
	if((af->prev?af->prev->data->nch:s->input.nch) != in.nch){
	  // Create channels filter
	  if(NULL == (new = af_prepend(s,af,"channels")))
	    return AF_ERROR;
	  // Set number of output channels
	  if(AF_OK != (rv = new->control(new,AF_CONTROL_CHANNELS,&in.nch)))
	    return rv;
	  // Initialize channels filter
	  if(!new->prev) 
	    memcpy(&in,&(s->input),sizeof(af_data_t));
	  else
	    memcpy(&in,new->prev->data,sizeof(af_data_t));
	  if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
	    return rv;
	}
	// Insert format filter
	if((af->prev?af->prev->data->format:s->input.format) != in.format){
	  // Create format filter
	  if(NULL == (new = af_prepend(s,af,"format")))
	    return AF_ERROR;
	  // Set output bits per sample
	  in.format |= af_bits2fmt(in.bps*8);
	  if(AF_OK != (rv = new->control(new,AF_CONTROL_FORMAT_FMT,&in.format)))
	    return rv;
	  // Initialize format filter
	  if(!new->prev) 
	    memcpy(&in,&(s->input),sizeof(af_data_t));
	  else
	    memcpy(&in,new->prev->data,sizeof(af_data_t));
	  if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
	    return rv;
	}
	if(!new){ // Should _never_ happen
	  af_msg(AF_MSG_ERROR,"[libaf] Unable to correct audio format. " 
		 "This error should never uccur, please send bugreport.\n");
	  return AF_ERROR;
	}
	af=new->next;
      }
      else {
        af_msg(AF_MSG_ERROR, "[libaf] Automatic filter insertion disabled "
               "but formats do not match. Giving up.\n");
        return AF_ERROR;
      }
      break;
    }
    case AF_DETACH:{ // Filter is redundant and wants to be unloaded
      // Do auto remove only if force is not specified
      if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
	af_instance_t* aft=af->prev;
	af_remove(s,af);
	if(aft)
	  af=aft->next;
	else
	  af=s->first; // Restart configuration
      }
      break;
    }
    default:
      af_msg(AF_MSG_ERROR,"[libaf] Reinitialization did not work, audio" 
	     " filter '%s' returned error code %i\n",af->info->name,rv);
      return AF_ERROR;
    }
  }while(af);
  return AF_OK;
}

// Uninit and remove all filters
void af_uninit(af_stream_t* s)
{
  while(s->first)
    af_remove(s,s->first);
}

/* Initialize the stream "s". This function creates a new filter list
   if necessary according to the values set in input and output. Input
   and output should contain the format of the current movie and the
   formate of the preferred output respectively. The function is
   reentrant i.e. if called with an already initialized stream the
   stream will be reinitialized.
   If one of the prefered output parameters is 0 the one that needs
   no conversion is used (i.e. the output format in the last filter).
   The return value is 0 if success and -1 if failure */
int af_init(af_stream_t* s)
{
  int i=0;

  // Sanity check
  if(!s) return -1;

  // Precaution in case caller is misbehaving
  s->input.audio  = s->output.audio  = NULL;
  s->input.len    = s->output.len    = 0;

  // Figure out how fast the machine is
  if(AF_INIT_AUTO == (AF_INIT_TYPE_MASK & s->cfg.force))
    s->cfg.force = (s->cfg.force & ~AF_INIT_TYPE_MASK) | AF_INIT_TYPE;

  // Check if this is the first call
  if(!s->first){
    // Add all filters in the list (if there are any)
    if(!s->cfg.list){      // To make automatic format conversion work
      if(!af_append(s,s->first,"dummy")) 
	return -1; 
    }
    else{
      while(s->cfg.list[i]){
	if(!af_append(s,s->last,s->cfg.list[i++]))
	  return -1;
      }
    }
  }

  // Init filters 
  if(AF_OK != af_reinit(s,s->first))
    return -1;

  // make sure the chain is not empty and valid (e.g. because of AF_DETACH)
  if (!s->first)
    if (!af_append(s,s->first,"dummy") || AF_OK != af_reinit(s,s->first))
      return -1;

  // Check output format
  if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
    af_instance_t* af = NULL; // New filter
    // Check output frequency if not OK fix with resample
    if(s->output.rate && s->last->data->rate!=s->output.rate){
      // try to find a filter that can change samplrate
      af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
               &(s->output.rate));
      if (!af) {
        char *resampler = "resample";
#ifdef CONFIG_LIBAVCODEC
        if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW)
          resampler = "lavcresample";
#endif
	if((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW){
	  if(!strcmp(s->first->info->name,"format"))
	    af = af_append(s,s->first,resampler);
	  else
	    af = af_prepend(s,s->first,resampler);
	}		
	else{
	  if(!strcmp(s->last->info->name,"format"))
	    af = af_prepend(s,s->last,resampler);
	  else
	    af = af_append(s,s->last,resampler);
	}
      // Init the new filter
      if(!af || (AF_OK != af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
				      &(s->output.rate))))
	return -1;
      // Use lin int if the user wants fast
      if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_FAST) {
        char args[32];
	sprintf(args, "%d", s->output.rate);
#ifdef CONFIG_LIBAVCODEC
	if (strcmp(resampler, "lavcresample") == 0)
	  strcat(args, ":1");
	else
#endif
	strcat(args, ":0:0");
	af->control(af, AF_CONTROL_COMMAND_LINE, args);
      }
      }
      if(AF_OK != af_reinit(s,af))
      	return -1;
    }	
      
    // Check number of output channels fix if not OK
    // If needed always inserted last -> easy to screw up other filters
    if(s->output.nch && s->last->data->nch!=s->output.nch){
      if(!strcmp(s->last->info->name,"format"))
	af = af_prepend(s,s->last,"channels");
      else
	af = af_append(s,s->last,"channels");
      // Init the new filter
      if(!af || (AF_OK != af->control(af,AF_CONTROL_CHANNELS,&(s->output.nch))))
	return -1;
      if(AF_OK != af_reinit(s,af))
	return -1;
    }
    
    // Check output format fix if not OK
    if(s->output.format != AF_FORMAT_UNKNOWN &&
		s->last->data->format != s->output.format){
      if(strcmp(s->last->info->name,"format"))
	af = af_append(s,s->last,"format");
      else
	af = s->last;
      // Init the new filter
      s->output.format |= af_bits2fmt(s->output.bps*8);
      if(!af || (AF_OK != af->control(af,AF_CONTROL_FORMAT_FMT,&(s->output.format))))
	return -1;
      if(AF_OK != af_reinit(s,af))
	return -1;
    }

    // Re init again just in case
    if(AF_OK != af_reinit(s,s->first))
      return -1;

    if (s->output.format == AF_FORMAT_UNKNOWN)
	s->output.format = s->last->data->format;
    if (!s->output.nch) s->output.nch = s->last->data->nch;
    if (!s->output.rate) s->output.rate = s->last->data->rate;
    if((s->last->data->format != s->output.format) || 
       (s->last->data->nch    != s->output.nch)    || 
       (s->last->data->rate   != s->output.rate))  {
      // Something is stuffed audio out will not work 
      af_msg(AF_MSG_ERROR,"[libaf] Unable to setup filter system can not" 
	     " meet sound-card demands, please send bugreport. \n");
      af_uninit(s);
      return -1;
    }
  }
  return 0;
}

/* Add filter during execution. This function adds the filter "name"
   to the stream s. The filter will be inserted somewhere nice in the
   list of filters. The return value is a pointer to the new filter,
   If the filter couldn't be added the return value is NULL. */
af_instance_t* af_add(af_stream_t* s, char* name){
  af_instance_t* new;
  // Sanity check
  if(!s || !s->first || !name)
    return NULL;
  // Insert the filter somwhere nice
  if(!strcmp(s->first->info->name,"format"))
    new = af_append(s, s->first, name);
  else
    new = af_prepend(s, s->first, name);
  if(!new)
    return NULL;

  // Reinitalize the filter list
  if(AF_OK != af_reinit(s, s->first)){
    free(new);
    return NULL;
  }
  return new;
}

// Filter data chunk through the filters in the list
af_data_t* af_play(af_stream_t* s, af_data_t* data)
{
  af_instance_t* af=s->first; 
  // Iterate through all filters 
  do{
    if (data->len <= 0) break;
    data=af->play(af,data);
    af=af->next;
  }while(af && data);
  return data;
}

/* Calculate the minimum output buffer size for given input data d
 * when using the RESIZE_LOCAL_BUFFER macro. The +t+1 part ensures the
 * value is >= len*mul rounded upwards to whole samples even if the
 * double 'mul' is inexact. */
int af_lencalc(double mul, af_data_t* d)
{
  int t = d->bps * d->nch;
  return d->len * mul + t + 1;
}

// Calculate average ratio of filter output size to input size
double af_calc_filter_multiplier(af_stream_t* s)
{
  af_instance_t* af=s->first; 
  double mul = 1;
  // Iterate through all filters and calculate total multiplication factor
  do{
      mul *= af->mul;
      af=af->next;
  }while(af);

  return mul;
}

/* Calculate the total delay [bytes output] caused by the filters */
double af_calc_delay(af_stream_t* s)
{
  af_instance_t* af=s->first; 
  register double delay = 0.0;
  // Iterate through all filters 
  while(af){
    delay += af->delay;
    delay *= af->mul;
    af=af->next;
  }
  return delay;
}

/* Helper function called by the macro with the same name this
   function should not be called directly */
int af_resize_local_buffer(af_instance_t* af, af_data_t* data)
{
  // Calculate new length
  register int len = af_lencalc(af->mul,data);
  af_msg(AF_MSG_VERBOSE,"[libaf] Reallocating memory in module %s, " 
	 "old len = %i, new len = %i\n",af->info->name,af->data->len,len);
  // If there is a buffer free it
  if(af->data->audio) 
    free(af->data->audio);
  // Create new buffer and check that it is OK
  af->data->audio = malloc(len);
  if(!af->data->audio){
    af_msg(AF_MSG_FATAL,"[libaf] Could not allocate memory \n");
    return AF_ERROR;
  }
  af->data->len=len;
  return AF_OK;
}

// documentation in af.h
af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg) {
  int res = AF_UNKNOWN;
  af_instance_t* filt = s->last;
  while (filt) {
    res = filt->control(filt, cmd, arg);
    if (res == AF_OK)
      return filt;
    filt = filt->prev;
  }
  return NULL;
}

void af_help (void) {
  int i = 0;
  af_msg(AF_MSG_INFO, "Available audio filters:\n");
  while (filter_list[i]) {
    if (filter_list[i]->comment && filter_list[i]->comment[0])
      af_msg(AF_MSG_INFO, "  %-15s: %s (%s)\n", filter_list[i]->name, filter_list[i]->info, filter_list[i]->comment);
    else
      af_msg(AF_MSG_INFO, "  %-15s: %s\n", filter_list[i]->name, filter_list[i]->info);
    i++;
  }
}

void af_fix_parameters(af_data_t *data)
{
    data->bps = af_fmt2bits(data->format)/8;
}