mirror of
https://github.com/mpv-player/mpv
synced 2024-12-16 11:55:42 +00:00
66abcc42af
And if set first parameter of this filter to 1, it will do ac3 passthrough like hwac3 did. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@25385 b3059339-0415-0410-9bf9-f77b7e298cf2
615 lines
17 KiB
C
615 lines
17 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#ifdef HAVE_MALLOC_H
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#include <malloc.h>
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#endif
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#include "af.h"
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// Static list of filters
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extern af_info_t af_info_dummy;
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extern af_info_t af_info_delay;
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extern af_info_t af_info_channels;
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extern af_info_t af_info_format;
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extern af_info_t af_info_resample;
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extern af_info_t af_info_volume;
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extern af_info_t af_info_equalizer;
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extern af_info_t af_info_gate;
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extern af_info_t af_info_comp;
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extern af_info_t af_info_pan;
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extern af_info_t af_info_surround;
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extern af_info_t af_info_sub;
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extern af_info_t af_info_export;
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extern af_info_t af_info_volnorm;
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extern af_info_t af_info_extrastereo;
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extern af_info_t af_info_lavcac3enc;
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extern af_info_t af_info_lavcresample;
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extern af_info_t af_info_sweep;
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extern af_info_t af_info_hrtf;
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extern af_info_t af_info_ladspa;
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extern af_info_t af_info_center;
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extern af_info_t af_info_sinesuppress;
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extern af_info_t af_info_karaoke;
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extern af_info_t af_info_scaletempo;
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static af_info_t* filter_list[]={
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&af_info_dummy,
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&af_info_delay,
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&af_info_channels,
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&af_info_format,
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&af_info_resample,
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&af_info_volume,
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&af_info_equalizer,
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&af_info_gate,
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&af_info_comp,
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&af_info_pan,
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&af_info_surround,
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&af_info_sub,
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#ifdef HAVE_SYS_MMAN_H
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&af_info_export,
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#endif
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&af_info_volnorm,
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&af_info_extrastereo,
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#ifdef USE_LIBAVCODEC_A
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&af_info_lavcac3enc,
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#endif
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#ifdef USE_LIBAVCODEC
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&af_info_lavcresample,
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#endif
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&af_info_sweep,
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&af_info_hrtf,
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#ifdef HAVE_LADSPA
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&af_info_ladspa,
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#endif
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&af_info_center,
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&af_info_sinesuppress,
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&af_info_karaoke,
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&af_info_scaletempo,
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NULL
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};
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// Message printing
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af_msg_cfg_t af_msg_cfg={0,NULL,NULL};
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// CPU speed
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int* af_cpu_speed = NULL;
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/* Find a filter in the static list of filters using it's name. This
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function is used internally */
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static af_info_t* af_find(char*name)
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{
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int i=0;
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while(filter_list[i]){
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if(!strcmp(filter_list[i]->name,name))
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return filter_list[i];
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i++;
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}
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af_msg(AF_MSG_ERROR,"Couldn't find audio filter '%s'\n",name);
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return NULL;
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}
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/* Find filter in the dynamic filter list using it's name This
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function is used for finding already initialized filters */
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af_instance_t* af_get(af_stream_t* s, char* name)
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{
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af_instance_t* af=s->first;
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// Find the filter
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while(af != NULL){
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if(!strcmp(af->info->name,name))
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return af;
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af=af->next;
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}
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return NULL;
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}
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/*/ Function for creating a new filter of type name. The name may
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contain the commandline parameters for the filter */
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static af_instance_t* af_create(af_stream_t* s, const char* name_with_cmd)
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{
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char* name = strdup(name_with_cmd);
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char* cmdline = name;
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// Allocate space for the new filter and reset all pointers
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af_instance_t* new=malloc(sizeof(af_instance_t));
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if (!name || !new) {
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af_msg(AF_MSG_ERROR,"[libaf] Could not allocate memory\n");
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goto err_out;
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}
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memset(new,0,sizeof(af_instance_t));
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// Check for commandline parameters
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strsep(&cmdline, "=");
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// Find filter from name
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if(NULL == (new->info=af_find(name)))
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goto err_out;
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/* Make sure that the filter is not already in the list if it is
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non-reentrant */
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if(new->info->flags & AF_FLAGS_NOT_REENTRANT){
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if(af_get(s,name)){
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af_msg(AF_MSG_ERROR,"[libaf] There can only be one instance of"
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" the filter '%s' in each stream\n",name);
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goto err_out;
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}
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}
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af_msg(AF_MSG_VERBOSE,"[libaf] Adding filter %s \n",name);
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// Initialize the new filter
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if(AF_OK == new->info->open(new) &&
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AF_ERROR < new->control(new,AF_CONTROL_POST_CREATE,&s->cfg)){
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if(cmdline){
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if(AF_ERROR>=new->control(new,AF_CONTROL_COMMAND_LINE,cmdline))
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goto err_out;
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}
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free(name);
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return new;
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}
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err_out:
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free(new);
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af_msg(AF_MSG_ERROR,"[libaf] Couldn't create or open audio filter '%s'\n",
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name);
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free(name);
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return NULL;
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}
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/* Create and insert a new filter of type name before the filter in the
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argument. This function can be called during runtime, the return
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value is the new filter */
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static af_instance_t* af_prepend(af_stream_t* s, af_instance_t* af, char* name)
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{
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// Create the new filter and make sure it is OK
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af_instance_t* new=af_create(s,name);
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if(!new)
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return NULL;
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// Update pointers
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new->next=af;
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if(af){
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new->prev=af->prev;
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af->prev=new;
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}
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else
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s->last=new;
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if(new->prev)
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new->prev->next=new;
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else
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s->first=new;
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return new;
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}
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/* Create and insert a new filter of type name after the filter in the
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argument. This function can be called during runtime, the return
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value is the new filter */
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static af_instance_t* af_append(af_stream_t* s, af_instance_t* af, char* name)
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{
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// Create the new filter and make sure it is OK
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af_instance_t* new=af_create(s,name);
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if(!new)
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return NULL;
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// Update pointers
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new->prev=af;
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if(af){
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new->next=af->next;
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af->next=new;
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}
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else
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s->first=new;
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if(new->next)
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new->next->prev=new;
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else
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s->last=new;
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return new;
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}
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// Uninit and remove the filter "af"
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void af_remove(af_stream_t* s, af_instance_t* af)
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{
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if(!af) return;
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// Print friendly message
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af_msg(AF_MSG_VERBOSE,"[libaf] Removing filter %s \n",af->info->name);
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// Notify filter before changing anything
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af->control(af,AF_CONTROL_PRE_DESTROY,0);
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// Detach pointers
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if(af->prev)
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af->prev->next=af->next;
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else
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s->first=af->next;
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if(af->next)
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af->next->prev=af->prev;
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else
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s->last=af->prev;
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// Uninitialize af and free memory
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af->uninit(af);
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free(af);
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}
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/* Reinitializes all filters downstream from the filter given in the
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argument the return value is AF_OK if success and AF_ERROR if
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failure */
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static int af_reinit(af_stream_t* s, af_instance_t* af)
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{
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do{
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af_data_t in; // Format of the input to current filter
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int rv=0; // Return value
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// Check if there are any filters left in the list
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if(NULL == af){
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if(!(af=af_append(s,s->first,"dummy")))
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return AF_UNKNOWN;
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else
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return AF_ERROR;
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}
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// Check if this is the first filter
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if(!af->prev)
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memcpy(&in,&(s->input),sizeof(af_data_t));
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else
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memcpy(&in,af->prev->data,sizeof(af_data_t));
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// Reset just in case...
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in.audio=NULL;
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in.len=0;
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rv = af->control(af,AF_CONTROL_REINIT,&in);
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switch(rv){
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case AF_OK:
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af = af->next;
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break;
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case AF_FALSE:{ // Configuration filter is needed
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// Do auto insertion only if force is not specified
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if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
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af_instance_t* new = NULL;
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// Insert channels filter
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if((af->prev?af->prev->data->nch:s->input.nch) != in.nch){
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// Create channels filter
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if(NULL == (new = af_prepend(s,af,"channels")))
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return AF_ERROR;
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// Set number of output channels
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if(AF_OK != (rv = new->control(new,AF_CONTROL_CHANNELS,&in.nch)))
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return rv;
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// Initialize channels filter
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if(!new->prev)
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memcpy(&in,&(s->input),sizeof(af_data_t));
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else
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memcpy(&in,new->prev->data,sizeof(af_data_t));
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if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
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return rv;
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}
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// Insert format filter
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if((af->prev?af->prev->data->format:s->input.format) != in.format){
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// Create format filter
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if(NULL == (new = af_prepend(s,af,"format")))
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return AF_ERROR;
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// Set output bits per sample
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in.format |= af_bits2fmt(in.bps*8);
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if(AF_OK != (rv = new->control(new,AF_CONTROL_FORMAT_FMT,&in.format)))
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return rv;
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// Initialize format filter
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if(!new->prev)
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memcpy(&in,&(s->input),sizeof(af_data_t));
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else
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memcpy(&in,new->prev->data,sizeof(af_data_t));
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if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
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return rv;
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}
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if(!new){ // Should _never_ happen
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af_msg(AF_MSG_ERROR,"[libaf] Unable to correct audio format. "
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"This error should never uccur, please send bugreport.\n");
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return AF_ERROR;
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}
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af=new->next;
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}
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else {
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af_msg(AF_MSG_ERROR, "[libaf] Automatic filter insertion disabled "
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"but formats do not match. Giving up.\n");
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return AF_ERROR;
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}
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break;
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}
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case AF_DETACH:{ // Filter is redundant and wants to be unloaded
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// Do auto remove only if force is not specified
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if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
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af_instance_t* aft=af->prev;
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af_remove(s,af);
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if(aft)
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af=aft->next;
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else
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af=s->first; // Restart configuration
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}
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break;
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}
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default:
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af_msg(AF_MSG_ERROR,"[libaf] Reinitialization did not work, audio"
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" filter '%s' returned error code %i\n",af->info->name,rv);
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return AF_ERROR;
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}
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}while(af);
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return AF_OK;
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}
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// Uninit and remove all filters
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void af_uninit(af_stream_t* s)
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{
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while(s->first)
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af_remove(s,s->first);
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}
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/* Initialize the stream "s". This function creates a new filter list
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if necessary according to the values set in input and output. Input
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and output should contain the format of the current movie and the
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formate of the preferred output respectively. The function is
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reentrant i.e. if called with an already initialized stream the
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stream will be reinitialized.
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If one of the prefered output parameters is 0 the one that needs
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no conversion is used (i.e. the output format in the last filter).
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The return value is 0 if success and -1 if failure */
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int af_init(af_stream_t* s)
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{
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int i=0;
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// Sanity check
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if(!s) return -1;
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// Precaution in case caller is misbehaving
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s->input.audio = s->output.audio = NULL;
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s->input.len = s->output.len = 0;
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// Figure out how fast the machine is
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if(AF_INIT_AUTO == (AF_INIT_TYPE_MASK & s->cfg.force))
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s->cfg.force = (s->cfg.force & ~AF_INIT_TYPE_MASK) | AF_INIT_TYPE;
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// Check if this is the first call
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if(!s->first){
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// Add all filters in the list (if there are any)
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if(!s->cfg.list){ // To make automatic format conversion work
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if(!af_append(s,s->first,"dummy"))
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return -1;
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}
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else{
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while(s->cfg.list[i]){
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if(!af_append(s,s->last,s->cfg.list[i++]))
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return -1;
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}
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}
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}
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// Init filters
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if(AF_OK != af_reinit(s,s->first))
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return -1;
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// make sure the chain is not empty and valid (e.g. because of AF_DETACH)
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if (!s->first)
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if (!af_append(s,s->first,"dummy") || AF_OK != af_reinit(s,s->first))
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return -1;
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// Check output format
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if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
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af_instance_t* af = NULL; // New filter
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// Check output frequency if not OK fix with resample
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if(s->output.rate && s->last->data->rate!=s->output.rate){
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// try to find a filter that can change samplrate
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af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
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&(s->output.rate));
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if (!af) {
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char *resampler = "resample";
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#ifdef USE_LIBAVCODEC
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if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW)
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resampler = "lavcresample";
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#endif
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if((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW){
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if(!strcmp(s->first->info->name,"format"))
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af = af_append(s,s->first,resampler);
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else
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af = af_prepend(s,s->first,resampler);
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}
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else{
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if(!strcmp(s->last->info->name,"format"))
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af = af_prepend(s,s->last,resampler);
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else
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af = af_append(s,s->last,resampler);
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}
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// Init the new filter
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if(!af || (AF_OK != af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
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&(s->output.rate))))
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return -1;
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// Use lin int if the user wants fast
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if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_FAST) {
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char args[32];
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sprintf(args, "%d", s->output.rate);
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#ifdef USE_LIBAVCODEC
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if (strcmp(resampler, "lavcresample") == 0)
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strcat(args, ":1");
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else
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#endif
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strcat(args, ":0:0");
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af->control(af, AF_CONTROL_COMMAND_LINE, args);
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}
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}
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if(AF_OK != af_reinit(s,af))
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return -1;
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}
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// Check number of output channels fix if not OK
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// If needed always inserted last -> easy to screw up other filters
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if(s->output.nch && s->last->data->nch!=s->output.nch){
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if(!strcmp(s->last->info->name,"format"))
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af = af_prepend(s,s->last,"channels");
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else
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af = af_append(s,s->last,"channels");
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// Init the new filter
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if(!af || (AF_OK != af->control(af,AF_CONTROL_CHANNELS,&(s->output.nch))))
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return -1;
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if(AF_OK != af_reinit(s,af))
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return -1;
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}
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// Check output format fix if not OK
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if(s->output.format != AF_FORMAT_UNKNOWN &&
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s->last->data->format != s->output.format){
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if(strcmp(s->last->info->name,"format"))
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af = af_append(s,s->last,"format");
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else
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af = s->last;
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// Init the new filter
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s->output.format |= af_bits2fmt(s->output.bps*8);
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if(!af || (AF_OK != af->control(af,AF_CONTROL_FORMAT_FMT,&(s->output.format))))
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return -1;
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if(AF_OK != af_reinit(s,af))
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return -1;
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}
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// Re init again just in case
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if(AF_OK != af_reinit(s,s->first))
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return -1;
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if (s->output.format == AF_FORMAT_UNKNOWN)
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s->output.format = s->last->data->format;
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if (!s->output.nch) s->output.nch = s->last->data->nch;
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if (!s->output.rate) s->output.rate = s->last->data->rate;
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if((s->last->data->format != s->output.format) ||
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(s->last->data->nch != s->output.nch) ||
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(s->last->data->rate != s->output.rate)) {
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// Something is stuffed audio out will not work
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af_msg(AF_MSG_ERROR,"[libaf] Unable to setup filter system can not"
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" meet sound-card demands, please send bugreport. \n");
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af_uninit(s);
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return -1;
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}
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}
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return 0;
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}
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/* Add filter during execution. This function adds the filter "name"
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to the stream s. The filter will be inserted somewhere nice in the
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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;
|
|
}
|