annotate libaf/af.c @ 36959:97a4746e7888

Utilize item defaults given in the skin configuration as start values. Do so for movie position, volume and balance (and hence show these values before playback). Add new btnValue() to retrieve the value and replace the btnModify() calls with btnValue() calls. (The btnModify() calls will be performed in the windows' draw handler prior to rendering anyway.) Initialize last_balance with -1 in order to be able to initialize balance (and volume) with the start values prior to playback (in GUI_SET_AUDIO).

/*
 * 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.
 */

#define _BSD_SOURCE

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

#include "osdep/strsep.h"
#include "libmpcodecs/dec_audio.h"
#include "mp_msg.h"
#include "af.h"

// Static list of filters
extern const af_info_t af_info_dummy;
extern const af_info_t af_info_delay;
extern const af_info_t af_info_channels;
extern const af_info_t af_info_format;
extern const af_info_t af_info_resample;
extern const af_info_t af_info_volume;
extern const af_info_t af_info_equalizer;
extern const af_info_t af_info_gate;
extern const af_info_t af_info_comp;
extern const af_info_t af_info_pan;
extern const af_info_t af_info_surround;
extern const af_info_t af_info_sub;
extern const af_info_t af_info_export;
extern const af_info_t af_info_volnorm;
extern const af_info_t af_info_extrastereo;
extern const af_info_t af_info_lavcac3enc;
extern const af_info_t af_info_lavcresample;
extern const af_info_t af_info_sweep;
extern const af_info_t af_info_hrtf;
extern const af_info_t af_info_ladspa;
extern const af_info_t af_info_center;
extern const af_info_t af_info_sinesuppress;
extern const af_info_t af_info_karaoke;
extern const af_info_t af_info_scaletempo;
extern const af_info_t af_info_stats;
extern const af_info_t af_info_bs2b;

static const af_info_t * const 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,
#if HAVE_SYS_MMAN_H
   &af_info_export,
#endif
   &af_info_volnorm,
   &af_info_extrastereo,
#ifdef CONFIG_FFMPEG
   &af_info_lavcac3enc,
   &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,
   &af_info_stats,
#ifdef CONFIG_LIBBS2B
   &af_info_bs2b,
#endif
   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 const 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++;
  }
  mp_msg(MSGT_AFILTER, MSGL_ERR, "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) {
    mp_msg(MSGT_AFILTER, MSGL_ERR, "[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)){
      mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] There can only be one instance of"
	     " the filter '%s' in each stream\n",name);
      goto err_out;
    }
  }

  mp_msg(MSGT_AFILTER, MSGL_V, "[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);
  mp_msg(MSGT_AFILTER, MSGL_ERR, "[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, const 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, const 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
  mp_msg(MSGT_AFILTER, MSGL_V, "[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);
}

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(!af)
      return af_append(s,s->first,"dummy") ? AF_ERROR : AF_UNKNOWN;

    // 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
	  mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. "
		 "This error should never occur, please send a bug report.\n");
	  return AF_ERROR;
	}
	af=new->next;
      }
      else {
        mp_msg(MSGT_AFILTER, MSGL_ERR, "[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:
      mp_msg(MSGT_AFILTER, MSGL_ERR, "[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);
}

/**
 * Extend the filter chain so we get the required output format at the end.
 * \return AF_ERROR on error, AF_OK if successful.
 */
static int fixup_output_format(af_stream_t* s)
{
    af_instance_t* af = NULL;
    // 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 AF_ERROR;
      if(AF_OK != af_reinit(s,af))
	return AF_ERROR;
    }

    // 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 AF_ERROR;
      if(AF_OK != af_reinit(s,af))
	return AF_ERROR;
    }

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

    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))  {
      return AF_ERROR;
    }
    return AF_OK;
}

/**
 * Automatic downmix to stereo in case the codec does not implement it.
 */
static void af_downmix(af_stream_t* s)
{
    static const char * const downmix_strs[AF_NCH + 1] = {
        /*                FL       FR       RL       RR          FC          LF         AL      AR */
        [3] = "pan=2:" "0.6:0:" "0:0.6:"                     "0.4:0.4",
        [4] = "pan=2:" "0.6:0:" "0:0.6:" "0.4:0:"  "0:0.4",
        [5] = "pan=2:" "0.5:0:" "0:0.5:" "0.2:0:"  "0:0.2:"  "0.3:0.3",
        [6] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:"  "0:0.2:"  "0.3:0.3:"   "0.1:0.1",
        [7] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:"  "0:0.2:"  "0.3:0.3:"              "0.1:0:" "0:0.1",
        [8] = "pan=2:" "0.4:0:" "0:0.4:" "0.15:0:" "0:0.15:" "0.25:0.25:" "0.1:0.1:" "0.1:0:" "0:0.1",
    };
    const char *af_pan_str = downmix_strs[s->input.nch];

    if (af_pan_str)
        af_append(s, s->first, af_pan_str);
}

/* 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){
    // Append a downmix pan filter at the beginning of the chain if needed
    if (s->input.nch != audio_output_channels && audio_output_channels == 2)
      af_downmix(s);
    // Add all filters in the list (if there are any)
    if (s->cfg.list) {
      while(s->cfg.list[i]){
	if(!af_append(s,s->last,s->cfg.list[i++]))
	  return -1;
      }
    }
  }

  // If we do not have any filters otherwise
  // add dummy to make automatic format conversion work
  if (!s->first && !af_append(s, s->first, "dummy"))
    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) {
        const char *resampler = "resample";
#ifdef CONFIG_FFMPEG
        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_FFMPEG
	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;
    }
    if (AF_OK != fixup_output_format(s)) {
      // Something is stuffed audio out will not work
      mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to setup filter system can not"
	     " meet sound-card demands, please send a bug report. \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 somewhere 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) ||
     AF_OK != fixup_output_format(s)){
    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);
  mp_msg(MSGT_AFILTER, MSGL_V, "[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
  free(af->data->audio);
  // Create new buffer and check that it is OK
  af->data->audio = malloc(len);
  if(!af->data->audio){
    mp_msg(MSGT_AFILTER, MSGL_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;
  mp_msg(MSGT_AFILTER, MSGL_INFO, "Available audio filters:\n");
  while (filter_list[i]) {
    if (filter_list[i]->comment && filter_list[i]->comment[0])
      mp_msg(MSGT_AFILTER, MSGL_INFO, "  %-15s: %s (%s)\n", filter_list[i]->name, filter_list[i]->info, filter_list[i]->comment);
    else
      mp_msg(MSGT_AFILTER, MSGL_INFO, "  %-15s: %s\n", filter_list[i]->name, filter_list[i]->info);
    i++;
  }
}

void af_fix_parameters(af_data_t *data)
{
    if (data->nch < 0 || data->nch > AF_NCH) {
      mp_msg(MSGT_AFILTER, MSGL_ERR, "Invalid number of channels %i, assuming 2.\n", data->nch);
      data->nch = 2;
    }
    data->bps = af_fmt2bits(data->format)/8;
}