Mercurial > mplayer.hg
view libaf/af.c @ 35410:1c991ec0e6da
Change declaration to use proper type to avoid a ridiculous
number of casts.
author | reimar |
---|---|
date | Wed, 28 Nov 2012 19:43:03 +0000 |
parents | b192c23ea7f1 |
children | f5b48e700de4 |
line wrap: on
line source
/* * 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 "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; }