Mercurial > mplayer.hg
view libaf/af.c @ 21074:738425edf828
100l typos, noticed by Carl Eugen Hoyos, cehoyos (at) ag or at
author | diego |
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date | Mon, 20 Nov 2006 00:55:57 +0000 |
parents | 1e78e35b6c7b |
children | 484b8eaaf28f |
line wrap: on
line source
#include <stdio.h> #include <stdlib.h> #include <string.h> #ifdef 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_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; 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 USE_LIBAVCODEC &af_info_lavcresample, #endif &af_info_sweep, &af_info_hrtf, #ifdef HAVE_LADSPA &af_info_ladspa, #endif &af_info_center, &af_info_sinesuppress, &af_info_karaoke, 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, char* name) { char* cmdline = name; // Allocate space for the new filter and reset all pointers af_instance_t* new=malloc(sizeof(af_instance_t)); if(!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)) return new; } else return new; } err_out: free(new); af_msg(AF_MSG_ERROR,"[libaf] Couldn't create or open audio filter '%s'\n", 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 USE_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 USE_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); return data; } /* Helper function used to calculate the exact buffer length needed when buffers are resized. The returned length is >= than what is needed */ inline int af_lencalc(frac_t mul, af_data_t* d){ register int t = d->bps*d->nch; return t*(((d->len/t)*mul.n)/mul.d + 1); } /* Calculate how long the output from the filters will be given the input length "len". The calculated length is >= the actual length. */ int af_outputlen(af_stream_t* s, int len) { int t = s->input.bps*s->input.nch; af_instance_t* af=s->first; frac_t mul = {1,1}; // Iterate through all filters do{ af_frac_mul(&mul, &af->mul); af=af->next; }while(af); return t * (((len/t)*mul.n + 1)/mul.d); } /* Calculate how long the input to the filters should be to produce a certain output length, i.e. the return value of this function is the input length required to produce the output length "len". The calculated length is <= the actual length */ int af_inputlen(af_stream_t* s, int len) { int t = s->input.bps*s->input.nch; af_instance_t* af=s->first; frac_t mul = {1,1}; // Iterate through all filters do{ af_frac_mul(&mul, &af->mul); af=af->next; }while(af); return t * (((len/t) * mul.d - 1)/mul.n); } /* Calculate how long the input IN to the filters should be to produce a certain output length OUT but with the following three constraints: 1. IN <= max_insize, where max_insize is the maximum possible input block length 2. OUT <= max_outsize, where max_outsize is the maximum possible output block length 3. If possible OUT >= len. Return -1 in case of error */ int af_calc_insize_constrained(af_stream_t* s, int len, int max_outsize,int max_insize) { int t = s->input.bps*s->input.nch; int in = 0; int out = 0; af_instance_t* af=s->first; frac_t mul = {1,1}; // Iterate through all filters and calculate total multiplication factor do{ af_frac_mul(&mul, &af->mul); af=af->next; }while(af); // Sanity check if(!mul.n || !mul.d) return -1; in = t * (((len/t) * mul.d - 1)/mul.n); if(in>max_insize) in=t*(max_insize/t); // Try to meet constraint nr 3. while((out=t * (((in/t+1)*mul.n - 1)/mul.d)) <= max_outsize && in<=max_insize){ if( (t * (((in/t)*mul.n))/mul.d) >= len) return in; in+=t; } // Could no meet constraint nr 3. while(out > max_outsize || in > max_insize){ in-=t; if(in<t) return -1; // Input parameters are probably incorrect out = t * (((in/t)*mul.n + 1)/mul.d); } return in; } /* Calculate the total delay [ms] 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; af=af->next; } return delay; } /* Helper function called by the macro with the same name this function should not be called directly */ inline 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; } /** * \brief calculate greatest common divisior of a and b. * \ingroup af_filter * * If both are 0 the result is 1. */ int af_gcd(register int a, register int b) { while (b != 0) { a %= b; if (a == 0) break; b %= a; } // the result is either in a or b. As the other one is 0 just add them. a += b; if (!a) return 1; return a; } /** * \brief cancel down a fraction f * \param f fraction to cancel down * \ingroup af_filter */ void af_frac_cancel(frac_t *f) { int gcd = af_gcd(f->n, f->d); f->n /= gcd; f->d /= gcd; } /** * \brief multiply out by in and store result in out. * \param out [inout] fraction to multiply by in * \param in [in] fraction to multiply out by * \ingroup af_filter * * the resulting fraction will be cancelled down * if in and out were. */ void af_frac_mul(frac_t *out, const frac_t *in) { int gcd1 = af_gcd(out->n, in->d); int gcd2 = af_gcd(in->n, out->d); out->n = (out->n / gcd1) * (in->n / gcd2); out->d = (out->d / gcd2) * (in->d / gcd1); } 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; }