diff options
Diffstat (limited to 'libaf')
34 files changed, 1324 insertions, 1324 deletions
diff --git a/libaf/af.c b/libaf/af.c index 987b7e548f..86b00a3e09 100644 --- a/libaf/af.c +++ b/libaf/af.c @@ -51,7 +51,7 @@ extern af_info_t af_info_scaletempo; extern af_info_t af_info_stats; extern af_info_t af_info_bs2b; -static af_info_t* filter_list[]={ +static af_info_t* filter_list[]={ &af_info_dummy, &af_info_delay, &af_info_channels, @@ -88,7 +88,7 @@ static af_info_t* filter_list[]={ #ifdef CONFIG_LIBBS2B &af_info_bs2b, #endif - NULL + NULL }; // CPU speed @@ -106,13 +106,13 @@ static af_info_t* af_find(char*name) } 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; + af_instance_t* af=s->first; // Find the filter while(af != NULL){ if(!strcmp(af->info->name,name)) @@ -134,7 +134,7 @@ static af_instance_t* af_create(af_stream_t* s, const char* name_with_cmd) 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 @@ -148,29 +148,29 @@ static af_instance_t* af_create(af_stream_t* s, const char* name_with_cmd) 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); + 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) && + 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; + return new; } - + err_out: free(new); mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Couldn't create or open audio filter '%s'\n", - name); + name); free(name); return NULL; } @@ -228,8 +228,8 @@ 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); + // 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); @@ -244,7 +244,7 @@ void af_remove(af_stream_t* s, af_instance_t* af) else s->last=af->prev; - // Uninitialize af and free memory + // Uninitialize af and free memory af->uninit(af); free(af); } @@ -260,21 +260,21 @@ static int af_reinit(af_stream_t* s, af_instance_t* af) // Check if there are any filters left in the list if(NULL == af){ - if(!(af=af_append(s,s->first,"dummy"))) - return AF_UNKNOWN; + 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) + // 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: @@ -293,7 +293,7 @@ static int af_reinit(af_stream_t* s, af_instance_t* af) if(AF_OK != (rv = new->control(new,AF_CONTROL_CHANNELS,&in.nch))) return rv; // Initialize channels filter - if(!new->prev) + if(!new->prev) memcpy(&in,&(s->input),sizeof(af_data_t)); else memcpy(&in,new->prev->data,sizeof(af_data_t)); @@ -310,7 +310,7 @@ static int af_reinit(af_stream_t* s, af_instance_t* af) if(AF_OK != (rv = new->control(new,AF_CONTROL_FORMAT_FMT,&in.format))) return rv; // Initialize format filter - if(!new->prev) + if(!new->prev) memcpy(&in,&(s->input),sizeof(af_data_t)); else memcpy(&in,new->prev->data,sizeof(af_data_t)); @@ -318,7 +318,7 @@ static int af_reinit(af_stream_t* s, af_instance_t* af) return rv; } if(!new){ // Should _never_ happen - mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. " + mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. " "This error should never uccur, please send bugreport.\n"); return AF_ERROR; } @@ -344,7 +344,7 @@ static int af_reinit(af_stream_t* s, af_instance_t* af) break; } default: - mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Reinitialization did not work, audio" + 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; } @@ -387,8 +387,8 @@ int af_init(af_stream_t* s) 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; + if(!af_append(s,s->first,"dummy")) + return -1; } else{ while(s->cfg.list[i]){ @@ -398,7 +398,7 @@ int af_init(af_stream_t* s) } } - // Init filters + // Init filters if(AF_OK != af_reinit(s,s->first)) return -1; @@ -426,7 +426,7 @@ int af_init(af_stream_t* s) 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); @@ -452,8 +452,8 @@ int af_init(af_stream_t* s) } 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){ @@ -467,7 +467,7 @@ int af_init(af_stream_t* s) 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){ @@ -491,11 +491,11 @@ int af_init(af_stream_t* s) 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) || + 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 - mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to setup filter system can not" + // 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 bugreport. \n"); af_uninit(s); return -1; @@ -532,8 +532,8 @@ af_instance_t* af_add(af_stream_t* s, char* name){ // 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 + af_instance_t* af=s->first; + // Iterate through all filters do{ if (data->len <= 0) break; data=af->play(af,data); @@ -555,7 +555,7 @@ int af_lencalc(double mul, af_data_t* d) // 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; + af_instance_t* af=s->first; double mul = 1; // Iterate through all filters and calculate total multiplication factor do{ @@ -569,9 +569,9 @@ double af_calc_filter_multiplier(af_stream_t* s) /* Calculate the total delay [bytes output] caused by the filters */ double af_calc_delay(af_stream_t* s) { - af_instance_t* af=s->first; + af_instance_t* af=s->first; register double delay = 0.0; - // Iterate through all filters + // Iterate through all filters while(af){ delay += af->delay; delay *= af->mul; @@ -586,10 +586,10 @@ 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, " + 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 - if(af->data->audio) + if(af->data->audio) free(af->data->audio); // Create new buffer and check that it is OK af->data->audio = malloc(len); diff --git a/libaf/af.h b/libaf/af.h index bbcec0dc33..aaf9b730e8 100644 --- a/libaf/af.h +++ b/libaf/af.h @@ -33,7 +33,7 @@ #undef AF_INIT_TYPE #define AF_INIT_TYPE \ ((gCpuCaps.has3DNow || gCpuCaps.hasSSE)?AF_INIT_FAST:AF_INIT_SLOW) -#endif +#endif struct af_instance_s; @@ -59,8 +59,8 @@ typedef struct af_data_s #define AF_FLAGS_NOT_REENTRANT 0x00000001 /* Audio filter information not specific for current instance, but for - a specific filter */ -typedef struct af_info_s + a specific filter */ +typedef struct af_info_s { const char *info; const char *name; @@ -80,7 +80,7 @@ typedef struct af_instance_s void* setup; // setup data for this specific instance and filter af_data_t* data; // configuration for outgoing data stream struct af_instance_s* next; - struct af_instance_s* prev; + struct af_instance_s* prev; double delay; /* Delay caused by the filter, in units of bytes read without * corresponding output */ double mul; /* length multiplier: how much does this instance change @@ -100,7 +100,7 @@ extern int* af_cpu_speed; #define AF_INIT_FLOAT 0x00000004 #define AF_INIT_FORMAT_MASK 0x00000004 -// Default init type +// Default init type #ifndef AF_INIT_TYPE #if HAVE_SSE || HAVE_AMD3DNOW #define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_FAST) @@ -194,7 +194,7 @@ void af_remove(af_stream_t* s, af_instance_t* af); * \brief find filter in chain by name * \param name name of the filter to find * \return first filter with right name or NULL if not found - * + * * This function is used for finding already initialized filters */ af_instance_t* af_get(af_stream_t* s, char* name); @@ -288,7 +288,7 @@ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma); * \param rate sample rate * \return AF_ERROR on error, AF_OK otherwise */ -int af_to_ms(int n, int* in, float* out, int rate); +int af_to_ms(int n, int* in, float* out, int rate); /** * \brief test if output format matches @@ -317,7 +317,7 @@ void af_help(void); * \brief fill the missing parameters in the af_data_t structure * \param data structure to fill * \ingroup af_filter - * + * * Currently only sets bps based on format */ void af_fix_parameters(af_data_t *data); @@ -344,7 +344,7 @@ void af_fix_parameters(af_data_t *data); #endif #ifndef sign -#define sign(a) (((a)>0)?(1):(-1)) +#define sign(a) (((a)>0)?(1):(-1)) #endif #ifndef lrnd diff --git a/libaf/af_center.c b/libaf/af_center.c index 5023872447..d3c867c39e 100644 --- a/libaf/af_center.c +++ b/libaf/af_center.c @@ -27,7 +27,7 @@ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include "af.h" @@ -40,7 +40,7 @@ typedef struct af_center_s // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_center_t* s = af->setup; + af_center_t* s = af->setup; switch(cmd){ case AF_CONTROL_REINIT:{ @@ -75,7 +75,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -90,7 +90,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) af_data_t* c = data; // Current working data af_center_t* s = af->setup; // Setup for this instance float* a = c->audio; // Audio data - int len = c->len/4; // Number of samples in current audio block + int len = c->len/4; // Number of samples in current audio block int nch = c->nch; // Number of channels int ch = s->ch; // Channel in which to insert the center audio register int i; diff --git a/libaf/af_channels.c b/libaf/af_channels.c index ffed84bfb0..b42cde380a 100644 --- a/libaf/af_channels.c +++ b/libaf/af_channels.c @@ -108,7 +108,7 @@ static void copy(void* in, void* out, int ins, int inos,int outs, int outos, int break; } default: - mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] Unsupported number of bytes/sample: %i" + mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] Unsupported number of bytes/sample: %i" " please report this error on the MPlayer mailing list. \n",bps); } } @@ -118,11 +118,11 @@ static int check_routes(af_channels_t* s, int nin, int nout) { int i; if((s->nr < 1) || (s->nr > AF_NCH)){ - mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of routing pairs must be" + mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of routing pairs must be" " between 1 and %i. Current value is %i\n",AF_NCH,s->nr); return AF_ERROR; } - + for(i=0;i<s->nr;i++){ if((s->route[i][FR] >= nin) || (s->route[i][TO] >= nout)){ mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] Invalid routing in pair nr. %i.\n", i); @@ -142,7 +142,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) // Set default channel assignment if(!s->router){ int i; - // Make sure this filter isn't redundant + // Make sure this filter isn't redundant if(af->data->nch == ((af_data_t*)arg)->nch) return AF_DETACH; @@ -180,14 +180,14 @@ static int control(struct af_instance_s* af, int cmd, void* arg) int ch = 0; // Sanity check if((s->nr < 1) || (s->nr > AF_NCH)){ - mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of routing pairs must be" + mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of routing pairs must be" " between 1 and %i. Current value is %i\n",AF_NCH,s->nr); - } + } s->router = 1; // Scan for pairs on commandline while((*cp == ':') && (ch < s->nr)){ sscanf(cp, ":%i:%i%n" ,&s->route[ch][FR], &s->route[ch][TO], &n); - mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Routing from channel %i to" + mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Routing from channel %i to" " channel %i\n",s->route[ch][FR],s->route[ch][TO]); cp = &cp[n]; ch++; @@ -197,20 +197,20 @@ static int control(struct af_instance_s* af, int cmd, void* arg) if(AF_OK != af->control(af,AF_CONTROL_CHANNELS | AF_CONTROL_SET ,&nch)) return AF_ERROR; return AF_OK; - } - case AF_CONTROL_CHANNELS | AF_CONTROL_SET: + } + case AF_CONTROL_CHANNELS | AF_CONTROL_SET: // Reinit must be called after this function has been called - + // Sanity check if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH){ - mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of output channels must be" + mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of output channels must be" " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]); return AF_ERROR; } - af->data->nch=((int*)arg)[0]; + af->data->nch=((int*)arg)[0]; if(!s->router) - mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Changing number of channels" + mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Changing number of channels" " to %i\n",af->data->nch); return AF_OK; case AF_CONTROL_CHANNELS | AF_CONTROL_GET: @@ -246,7 +246,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { free(af->setup); @@ -262,18 +262,18 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) af_data_t* l = af->data; // Local data af_channels_t* s = af->setup; int i; - + if(AF_OK != RESIZE_LOCAL_BUFFER(af,data)) return NULL; // Reset unused channels memset(l->audio,0,c->len / c->nch * l->nch); - + if(AF_OK == check_routes(s,c->nch,l->nch)) for(i=0;i<s->nr;i++) copy(c->audio,l->audio,c->nch,s->route[i][FR], l->nch,s->route[i][TO],c->len,c->bps); - + // Set output data c->audio = l->audio; c->len = c->len / c->nch * l->nch; diff --git a/libaf/af_comp.c b/libaf/af_comp.c index 0eef76c2f4..4999612865 100644 --- a/libaf/af_comp.c +++ b/libaf/af_comp.c @@ -43,21 +43,21 @@ typedef struct af_comp_s // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_comp_t* s = (af_comp_t*)af->setup; + af_comp_t* s = (af_comp_t*)af->setup; int i; switch(cmd){ case AF_CONTROL_REINIT: // Sanity check if(!arg) return AF_ERROR; - + af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; // Time constant set to 0.1s - // s->alpha = (1.0/0.2)/(2.0*M_PI*(float)((af_data_t*)arg)->rate); + // s->alpha = (1.0/0.2)/(2.0*M_PI*(float)((af_data_t*)arg)->rate); return af_test_output(af,(af_data_t*)arg); case AF_CONTROL_COMMAND_LINE:{ /* float v=-10.0; */ @@ -76,10 +76,10 @@ static int control(struct af_instance_s* af, int cmd, void* arg) } case AF_CONTROL_COMP_ON_OFF | AF_CONTROL_SET: memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int)); - return AF_OK; + return AF_OK; case AF_CONTROL_COMP_ON_OFF | AF_CONTROL_GET: memcpy((int*)arg,s->enable,AF_NCH*sizeof(int)); - return AF_OK; + return AF_OK; case AF_CONTROL_COMP_THRESH | AF_CONTROL_SET: return af_from_dB(AF_NCH,(float*)arg,s->tresh,20.0,-60.0,-1.0); case AF_CONTROL_COMP_THRESH | AF_CONTROL_GET: @@ -93,18 +93,18 @@ static int control(struct af_instance_s* af, int cmd, void* arg) case AF_CONTROL_COMP_RELEASE | AF_CONTROL_GET: return af_to_ms(AF_NCH,s->release,(float*)arg,af->data->rate); case AF_CONTROL_COMP_RATIO | AF_CONTROL_SET: - for(i=0;i<AF_NCH;i++) + for(i=0;i<AF_NCH;i++) s->ratio[i] = clamp(((float*)arg)[i],1.0,10.0); return AF_OK; case AF_CONTROL_COMP_RATIO | AF_CONTROL_GET: - for(i=0;i<AF_NCH;i++) + for(i=0;i<AF_NCH;i++) ((float*)arg)[i] = s->ratio[i]; - return AF_OK; + return AF_OK; } return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -121,7 +121,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) float* a = (float*)c->audio; // Audio data int len = c->len/4; // Number of samples int ch = 0; // Channel counter - register int nch = c->nch; // Number of channels + register int nch = c->nch; // Number of channels register int i = 0; // Compress/expand @@ -130,8 +130,8 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) float t = 1.0 - s->time[ch]; for(i=ch;i<len;i+=nch){ register float x = a[i]; - register float pow = x*x; - s->pow[ch] = t*s->pow[ch] + + register float pow = x*x; + s->pow[ch] = t*s->pow[ch] + pow*s->time[ch]; // LP filter if(pow < s->pow[ch]){ ; diff --git a/libaf/af_delay.c b/libaf/af_delay.c index 4b94e00066..02327d0c53 100644 --- a/libaf/af_delay.c +++ b/libaf/af_delay.c @@ -37,7 +37,7 @@ typedef struct af_delay_s void* q[AF_NCH]; // Circular queues used for delaying audio signal int wi[AF_NCH]; // Write index int ri; // Read index - float d[AF_NCH]; // Delay [ms] + float d[AF_NCH]; // Delay [ms] }af_delay_t; // Initialization and runtime control @@ -108,7 +108,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { int i; @@ -134,7 +134,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) switch(c->bps){ case 1:{ int8_t* a = c->audio; - int8_t* q = s->q[ch]; + int8_t* q = s->q[ch]; int wi = s->wi[ch]; ri = s->ri; for(i=ch;i<len;i+=nch){ @@ -148,7 +148,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) } case 2:{ int16_t* a = c->audio; - int16_t* q = s->q[ch]; + int16_t* q = s->q[ch]; int wi = s->wi[ch]; ri = s->ri; for(i=ch;i<len;i+=nch){ @@ -162,7 +162,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) } case 4:{ int32_t* a = c->audio; - int32_t* q = s->q[ch]; + int32_t* q = s->q[ch]; int wi = s->wi[ch]; ri = s->ri; for(i=ch;i<len;i+=nch){ diff --git a/libaf/af_dummy.c b/libaf/af_dummy.c index 6e32116b99..2d375924e1 100644 --- a/libaf/af_dummy.c +++ b/libaf/af_dummy.c @@ -38,7 +38,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) diff --git a/libaf/af_equalizer.c b/libaf/af_equalizer.c index 519d9aba4c..cdc0011cb7 100644 --- a/libaf/af_equalizer.c +++ b/libaf/af_equalizer.c @@ -32,13 +32,13 @@ #include "af.h" #define L 2 // Storage for filter taps -#define KM 10 // Max number of bands +#define KM 10 // Max number of bands #define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2) gives 4dB suppression @ Fc*2 and Fc/2 */ /* Center frequencies for band-pass filters - The different frequency bands are: + The different frequency bands are: nr. center frequency 0 31.25 Hz 1 62.50 Hz @@ -55,7 +55,7 @@ // Maximum and minimum gain for the bands #define G_MAX +12.0 -#define G_MIN -12.0 +#define G_MIN -12.0 // Data for specific instances of this filter typedef struct af_equalizer_s @@ -76,7 +76,7 @@ static void bp2(float* a, float* b, float fc, float q){ a[0] = (1.0 + C) * cos(th); a[1] = -1 * C; - + b[0] = (1.0 - C)/2.0; b[1] = -1.0050; } @@ -84,30 +84,30 @@ static void bp2(float* a, float* b, float fc, float q){ // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_equalizer_t* s = (af_equalizer_t*)af->setup; + af_equalizer_t* s = (af_equalizer_t*)af->setup; switch(cmd){ case AF_CONTROL_REINIT:{ int k =0, i =0; float F[KM] = CF; - + s->gain_factor=0.0; // Sanity check if(!arg) return AF_ERROR; - + af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; - + // Calculate number of active filters s->K=KM; while(F[s->K-1] > (float)af->data->rate/2.2) s->K--; - + if(s->K != KM) - mp_msg(MSGT_AFILTER, MSGL_INFO, "[equalizer] Limiting the number of filters to" + mp_msg(MSGT_AFILTER, MSGL_INFO, "[equalizer] Limiting the number of filters to" " %i due to low sample rate.\n",s->K); // Generate filter taps @@ -116,7 +116,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) // Calculate how much this plugin adds to the overall time delay af->delay = 2 * af->data->nch * af->data->bps; - + // Calculate gain factor to prevent clipping at output for(k=0;k<AF_NCH;k++) { @@ -127,24 +127,24 @@ static int control(struct af_instance_s* af, int cmd, void* arg) } s->gain_factor=log10(s->gain_factor + 1.0) * 20.0; - + if(s->gain_factor > 0.0) { s->gain_factor=0.1+(s->gain_factor/12.0); }else{ s->gain_factor=1; } - + return af_test_output(af,arg); } case AF_CONTROL_COMMAND_LINE:{ float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0}; int i,j; - sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1], + sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1], &g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]); for(i=0;i<AF_NCH;i++){ for(j=0;j<KM;j++){ - ((af_equalizer_t*)af->setup)->g[i][j] = + ((af_equalizer_t*)af->setup)->g[i][j] = pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0; } } @@ -178,7 +178,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -191,12 +191,12 @@ static void uninit(struct af_instance_s* af) static af_data_t* play(struct af_instance_s* af, af_data_t* data) { af_data_t* c = data; // Current working data - af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup + af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup uint32_t ci = af->data->nch; // Index for channels uint32_t nch = af->data->nch; // Number of channels while(ci--){ - float* g = s->g[ci]; // Gain factor + float* g = s->g[ci]; // Gain factor float* in = ((float*)c->audio)+ci; float* out = ((float*)c->audio)+ci; float* end = in + c->len/4; // Block loop end @@ -205,7 +205,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) register int k = 0; // Frequency band index register float yt = *in; // Current input sample in+=nch; - + // Run the filters for(;k<s->K;k++){ // Pointer to circular buffer wq @@ -218,7 +218,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) wq[1] = wq[0]; wq[0] = w; } - // Calculate output + // Calculate output *out=yt*s->gain_factor; out+=nch; } diff --git a/libaf/af_export.c b/libaf/af_export.c index 59a6e8ec00..949f3f18ca 100644 --- a/libaf/af_export.c +++ b/libaf/af_export.c @@ -41,7 +41,7 @@ #include "get_path.h" #define DEF_SZ 512 // default buffer size (in samples) -#define SHARED_FILE "mplayer-af_export" /* default file name +#define SHARED_FILE "mplayer-af_export" /* default file name (relative to ~/.mplayer/ */ #define SIZE_HEADER (2 * sizeof(int) + sizeof(unsigned long long)) @@ -81,46 +81,46 @@ static int control(struct af_instance_s* af, int cmd, void* arg) munmap(s->mmap_area, SIZE_HEADER + (af->data->bps*s->sz*af->data->nch)); // close previous file descriptor if(s->fd) - close(s->fd); + close(s->fd); // Accept only int16_t as input format (which sucks) af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; - + // If buffer length isn't set, set it to the default value if(s->sz == 0) s->sz = DEF_SZ; - + // Allocate new buffers (as one continuous block) s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps); if(NULL == s->buf[0]) mp_msg(MSGT_AFILTER, MSGL_FATAL, "[export] Out of memory\n"); for(i = 1; i < af->data->nch; i++) s->buf[i] = s->buf[0] + i*s->sz*af->data->bps; - + // Init memory mapping s->fd = open(s->filename, O_RDWR | O_CREAT | O_TRUNC, 0640); mp_msg(MSGT_AFILTER, MSGL_INFO, "[export] Exporting to file: %s\n", s->filename); if(s->fd < 0) - mp_msg(MSGT_AFILTER, MSGL_FATAL, "[export] Could not open/create file: %s\n", + mp_msg(MSGT_AFILTER, MSGL_FATAL, "[export] Could not open/create file: %s\n", s->filename); - + // header + buffer mapsize = (SIZE_HEADER + (af->data->bps * s->sz * af->data->nch)); - + // grow file to needed size for(i = 0; i < mapsize; i++){ char null = 0; write(s->fd, (void*) &null, 1); } - + // mmap size s->mmap_area = mmap(0, mapsize, PROT_READ|PROT_WRITE,MAP_SHARED, s->fd, 0); if(s->mmap_area == NULL) mp_msg(MSGT_AFILTER, MSGL_FATAL, "[export] Could not mmap file %s\n", s->filename); - mp_msg(MSGT_AFILTER, MSGL_INFO, "[export] Memory mapped to file: %s (%p)\n", + mp_msg(MSGT_AFILTER, MSGL_INFO, "[export] Memory mapped to file: %s (%p)\n", s->filename, s->mmap_area); // Initialize header @@ -134,15 +134,15 @@ static int control(struct af_instance_s* af, int cmd, void* arg) case AF_CONTROL_COMMAND_LINE:{ int i=0; char *str = arg; - + if (!str){ - if(s->filename) + if(s->filename) free(s->filename); s->filename = get_path(SHARED_FILE); return AF_OK; } - + while((str[i]) && (str[i] != ':')) i++; @@ -152,9 +152,9 @@ static int control(struct af_instance_s* af, int cmd, void* arg) s->filename = calloc(i + 1, 1); memcpy(s->filename, str, i); s->filename[i] = 0; - + sscanf(str + i + 1, "%d", &(s->sz)); - + return af->control(af, AF_CONTROL_EXPORT_SZ | AF_CONTROL_SET, &s->sz); } case AF_CONTROL_EXPORT_SZ | AF_CONTROL_SET: @@ -167,7 +167,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) case AF_CONTROL_EXPORT_SZ | AF_CONTROL_GET: *(int*) arg = s->sz; return AF_OK; - + } return AF_UNKNOWN; } @@ -186,10 +186,10 @@ static void uninit( struct af_instance_s* af ) af_export_t* s = af->setup; if (s->buf && s->buf[0]) free(s->buf[0]); - + // Free mmaped area if(s->mmap_area) - munmap(s->mmap_area, sizeof(af_export_t)); + munmap(s->mmap_area, sizeof(af_export_t)); if(s->fd > -1) close(s->fd); @@ -215,15 +215,15 @@ static af_data_t* play( struct af_instance_s* af, af_data_t* data ) int len = c->len/c->bps; // Number of sample in data chunk int sz = s->sz; // buffer size (in samples) int flag = 0; // Set to 1 if buffer is filled - + int ch, i; // Fill all buffers for(ch = 0; ch < nch; ch++){ int wi = s->wi; // Reset write index - int16_t* b = s->buf[ch]; // Current buffer + int16_t* b = s->buf[ch]; // Current buffer - // Copy data to export buffers + // Copy data to export buffers for(i = ch; i < len; i += nch){ b[wi++] = a[i]; if(wi >= sz){ // Don't write outside the end of the buffer @@ -239,7 +239,7 @@ static af_data_t* play( struct af_instance_s* af, af_data_t* data ) // update buffer in mapped area memcpy(s->mmap_area + SIZE_HEADER, s->buf[0], sz * c->bps * nch); s->count++; // increment counter (to sync) - memcpy(s->mmap_area + SIZE_HEADER - sizeof(s->count), + memcpy(s->mmap_area + SIZE_HEADER - sizeof(s->count), &(s->count), sizeof(s->count)); } @@ -248,7 +248,7 @@ static af_data_t* play( struct af_instance_s* af, af_data_t* data ) } /* Allocate memory and set function pointers - af audio filter instance + af audio filter instance returns AF_OK or AF_ERROR */ static int af_open( af_instance_t* af ) diff --git a/libaf/af_extrastereo.c b/libaf/af_extrastereo.c index 88d2557fd2..f0af96ff06 100644 --- a/libaf/af_extrastereo.c +++ b/libaf/af_extrastereo.c @@ -20,7 +20,7 @@ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include <inttypes.h> #include <math.h> @@ -40,13 +40,13 @@ static af_data_t* play_float(struct af_instance_s* af, af_data_t* data); // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_extrastereo_t* s = (af_extrastereo_t*)af->setup; + af_extrastereo_t* s = (af_extrastereo_t*)af->setup; switch(cmd){ case AF_CONTROL_REINIT:{ // Sanity check if(!arg) return AF_ERROR; - + af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = 2; if (((af_data_t*)arg)->format == AF_FORMAT_FLOAT_NE) @@ -79,7 +79,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -96,11 +96,11 @@ static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data) int16_t *a = (int16_t*)data->audio; // Audio data int len = data->len/2; // Number of samples int avg, l, r; - + for (i = 0; i < len; i+=2) { avg = (a[i] + a[i + 1]) / 2; - + l = avg + (int)(s->mul * (a[i] - avg)); r = avg + (int)(s->mul * (a[i + 1] - avg)); @@ -118,14 +118,14 @@ static af_data_t* play_float(struct af_instance_s* af, af_data_t* data) float *a = (float*)data->audio; // Audio data int len = data->len/4; // Number of samples float avg, l, r; - + for (i = 0; i < len; i+=2) { avg = (a[i] + a[i + 1]) / 2; - + l = avg + (s->mul * (a[i] - avg)); r = avg + (s->mul * (a[i + 1] - avg)); - + a[i] = af_softclip(l); a[i + 1] = af_softclip(r); } diff --git a/libaf/af_format.c b/libaf/af_format.c index 8917a870a6..faeae02efc 100644 --- a/libaf/af_format.c +++ b/libaf/af_format.c @@ -36,7 +36,7 @@ format */ /* The below includes retrieves functions for converting to and from - ulaw and alaw */ + ulaw and alaw */ #include "af_format_ulaw.h" #include "af_format_alaw.h" @@ -62,7 +62,7 @@ static af_data_t* play_s16_float(struct af_instance_s* af, af_data_t* data); static int check_bps(int bps) { if(bps != 4 && bps != 3 && bps != 2 && bps != 1){ - mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] The number of bytes per sample" + mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] The number of bytes per sample" " must be 1, 2, 3 or 4. Current value is %i \n",bps); return AF_ERROR; } @@ -74,11 +74,11 @@ static int check_format(int format) { char buf[256]; switch(format & AF_FORMAT_SPECIAL_MASK){ - case(AF_FORMAT_IMA_ADPCM): - case(AF_FORMAT_MPEG2): + case(AF_FORMAT_IMA_ADPCM): + case(AF_FORMAT_MPEG2): case(AF_FORMAT_AC3): mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] Sample format %s not yet supported \n", - af_fmt2str(format,buf,256)); + af_fmt2str(format,buf,256)); return AF_ERROR; } return AF_OK; @@ -92,9 +92,9 @@ static int control(struct af_instance_s* af, int cmd, void* arg) char buf1[256]; char buf2[256]; af_data_t *data = arg; - - // Make sure this filter isn't redundant - if(af->data->format == data->format && + + // Make sure this filter isn't redundant + if(af->data->format == data->format && af->data->bps == data->bps) return AF_DETACH; @@ -112,9 +112,9 @@ static int control(struct af_instance_s* af, int cmd, void* arg) af->data->rate = data->rate; af->data->nch = data->nch; af->mul = (double)af->data->bps / data->bps; - + af->play = play; // set default - + // look whether only endianness differences are there if ((af->data->format & ~AF_FORMAT_END_MASK) == (data->format & ~AF_FORMAT_END_MASK)) @@ -164,13 +164,13 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if (af->data) free(af->data->audio); free(af->data); - af->setup = 0; + af->setup = 0; } static af_data_t* play_swapendian(struct af_instance_s* af, af_data_t* data) @@ -271,10 +271,10 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) } } else { // Input must be int - + // Change signed/unsigned if((c->format&AF_FORMAT_SIGN_MASK) != (l->format&AF_FORMAT_SIGN_MASK)){ - si2us(c->audio,len,c->bps); + si2us(c->audio,len,c->bps); } // Convert to special formats switch(l->format&(AF_FORMAT_SPECIAL_MASK|AF_FORMAT_POINT_MASK)){ @@ -297,7 +297,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) } } - // Switch from cpu native endian to the correct endianness + // Switch from cpu native endian to the correct endianness if((l->format&AF_FORMAT_END_MASK)!=AF_FORMAT_NE) endian(l->audio,l->audio,len,l->bps); @@ -450,7 +450,7 @@ static void change_bps(void* in, void* out, int len, int inbps, int outbps) ((uint32_t*)out)[i]=(uint32_t)load24bit(in, i); break; } - break; + break; case(4): switch(outbps){ case(1): @@ -466,7 +466,7 @@ static void change_bps(void* in, void* out, int len, int inbps, int outbps) store24bit(out, i, ((uint32_t*)in)[i]); break; } - break; + break; } } @@ -478,7 +478,7 @@ static void float2int(float* in, void* out, int len, int bps) for(i=0;i<len;i++) ((int8_t*)out)[i] = lrintf(127.0 * in[i]); break; - case(2): + case(2): for(i=0;i<len;i++) ((int16_t*)out)[i] = lrintf(32767.0 * in[i]); break; @@ -490,7 +490,7 @@ static void float2int(float* in, void* out, int len, int bps) for(i=0;i<len;i++) ((int32_t*)out)[i] = lrintf(2147483647.0 * in[i]); break; - } + } } static void int2float(void* in, float* out, int len, int bps) @@ -513,5 +513,5 @@ static void int2float(void* in, float* out, int len, int bps) for(i=0;i<len;i++) out[i]=(1.0/2147483648.0)*((int32_t*)in)[i]; break; - } + } } diff --git a/libaf/af_format_alaw.h b/libaf/af_format_alaw.h index bddf1a5374..d7c00884f7 100644 --- a/libaf/af_format_alaw.h +++ b/libaf/af_format_alaw.h @@ -30,195 +30,195 @@ // Conversion tables (the function are below) static short alaw_decode [128] = -{ -5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, - -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, - -2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368, - -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, - -22016, -20992, -24064, -23040, -17920, -16896, -19968, -18944, - -30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136, - -11008, -10496, -12032, -11520, -8960, -8448, -9984, -9472, - -15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568, - -344, -328, -376, -360, -280, -264, -312, -296, - -472, -456, -504, -488, -408, -392, -440, -424, - -88, -72, -120, -104, -24, -8, -56, -40, - -216, -200, -248, -232, -152, -136, -184, -168, - -1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184, - -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, - -688, -656, -752, -720, -560, -528, -624, -592, +{ -5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, + -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, + -2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368, + -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, + -22016, -20992, -24064, -23040, -17920, -16896, -19968, -18944, + -30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136, + -11008, -10496, -12032, -11520, -8960, -8448, -9984, -9472, + -15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568, + -344, -328, -376, -360, -280, -264, -312, -296, + -472, -456, -504, -488, -408, -392, -440, -424, + -88, -72, -120, -104, -24, -8, -56, -40, + -216, -200, -248, -232, -152, -136, -184, -168, + -1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184, + -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, + -688, -656, -752, -720, -560, -528, -624, -592, -944, -912, -1008, -976, -816, -784, -880, -848 } ; /* alaw_decode */ static unsigned char alaw_encode [2049] = -{ 0xD5, 0xD4, 0xD7, 0xD6, 0xD1, 0xD0, 0xD3, 0xD2, 0xDD, 0xDC, 0xDF, 0xDE, - 0xD9, 0xD8, 0xDB, 0xDA, 0xC5, 0xC4, 0xC7, 0xC6, 0xC1, 0xC0, 0xC3, 0xC2, - 0xCD, 0xCC, 0xCF, 0xCE, 0xC9, 0xC8, 0xCB, 0xCA, 0xF5, 0xF5, 0xF4, 0xF4, - 0xF7, 0xF7, 0xF6, 0xF6, 0xF1, 0xF1, 0xF0, 0xF0, 0xF3, 0xF3, 0xF2, 0xF2, - 0xFD, 0xFD, 0xFC, 0xFC, 0xFF, 0xFF, 0xFE, 0xFE, 0xF9, 0xF9, 0xF8, 0xF8, - 0xFB, 0xFB, 0xFA, 0xFA, 0xE5, 0xE5, 0xE5, 0xE5, 0xE4, 0xE4, 0xE4, 0xE4, - 0xE7, 0xE7, 0xE7, 0xE7, 0xE6, 0xE6, 0xE6, 0xE6, 0xE1, 0xE1, 0xE1, 0xE1, - 0xE0, 0xE0, 0xE0, 0xE0, 0xE3, 0xE3, 0xE3, 0xE3, 0xE2, 0xE2, 0xE2, 0xE2, - 0xED, 0xED, 0xED, 0xED, 0xEC, 0xEC, 0xEC, 0xEC, 0xEF, 0xEF, 0xEF, 0xEF, - 0xEE, 0xEE, 0xEE, 0xEE, 0xE9, 0xE9, 0xE9, 0xE9, 0xE8, 0xE8, 0xE8, 0xE8, - 0xEB, 0xEB, 0xEB, 0xEB, 0xEA, 0xEA, 0xEA, 0xEA, 0x95, 0x95, 0x95, 0x95, - 0x95, 0x95, 0x95, 0x95, 0x94, 0x94, 0x94, 0x94, 0x94, 0x94, 0x94, 0x94, - 0x97, 0x97, 0x97, 0x97, 0x97, 0x97, 0x97, 0x97, 0x96, 0x96, 0x96, 0x96, - 0x96, 0x96, 0x96, 0x96, 0x91, 0x91, 0x91, 0x91, 0x91, 0x91, 0x91, 0x91, - 0x90, 0x90, 0x90, 0x90, 0x90, 0x90, 0x90, 0x90, 0x93, 0x93, 0x93, 0x93, - 0x93, 0x93, 0x93, 0x93, 0x92, 0x92, 0x92, 0x92, 0x92, 0x92, 0x92, 0x92, - 0x9D, 0x9D, 0x9D, 0x9D, 0x9D, 0x9D, 0x9D, 0x9D, 0x9C, 0x9C, 0x9C, 0x9C, - 0x9C, 0x9C, 0x9C, 0x9C, 0x9F, 0x9F, 0x9F, 0x9F, 0x9F, 0x9F, 0x9F, 0x9F, - 0x9E, 0x9E, 0x9E, 0x9E, 0x9E, 0x9E, 0x9E, 0x9E, 0x99, 0x99, 0x99, 0x99, - 0x99, 0x99, 0x99, 0x99, 0x98, 0x98, 0x98, 0x98, 0x98, 0x98, 0x98, 0x98, - 0x9B, 0x9B, 0x9B, 0x9B, 0x9B, 0x9B, 0x9B, 0x9B, 0x9A, 0x9A, 0x9A, 0x9A, - 0x9A, 0x9A, 0x9A, 0x9A, 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, - 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, 0x84, 0x84, 0x84, 0x84, - 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, - 0x87, 0x87, 0x87, 0x87, 0x87, 0x87, 0x87, 0x87, 0x87, 0x87, 0x87, 0x87, - 0x87, 0x87, 0x87, 0x87, 0x86, 0x86, 0x86, 0x86, 0x86, 0x86, 0x86, 0x86, - 0x86, 0x86, 0x86, 0x86, 0x86, 0x86, 0x86, 0x86, 0x81, 0x81, 0x81, 0x81, - 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x83, 0x83, 0x83, 0x83, 0x83, 0x83, 0x83, 0x83, - 0x83, 0x83, 0x83, 0x83, 0x83, 0x83, 0x83, 0x83, 0x82, 0x82, 0x82, 0x82, - 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, - 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, 0x8D, - 0x8D, 0x8D, 0x8D, 0x8D, 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, - 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, 0x8C, 0x8F, 0x8F, 0x8F, 0x8F, - 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, 0x8F, - 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, 0x8E, - 0x8E, 0x8E, 0x8E, 0x8E, 0x89, 0x89, 0x89, 0x89, 0x89, 0x89, 0x89, 0x89, - 0x89, 0x89, 0x89, 0x89, 0x89, 0x89, 0x89, 0x89, 0x88, 0x88, 0x88, 0x88, - 0x88, 0x88, 0x88, 0x88, 0x88, 0x88, 0x88, 0x88, 0x88, 0x88, 0x88, 0x88, - 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, 0x8B, - 0x8B, 0x8B, 0x8B, 0x8B, 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, - 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, 0x8A, 0xB5, 0xB5, 0xB5, 0xB5, - 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, - 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, 0xB5, - 0xB5, 0xB5, 0xB5, 0xB5, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, - 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, - 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, 0xB4, - 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, - 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, - 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB7, 0xB6, 0xB6, 0xB6, 0xB6, - 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, - 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, 0xB6, - 0xB6, 0xB6, 0xB6, 0xB6, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, - 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, - 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, 0xB1, - 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, - 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, - 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB3, 0xB3, 0xB3, 0xB3, - 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, - 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, 0xB3, - 0xB3, 0xB3, 0xB3, 0xB3, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, - 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, - 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, 0xB2, - 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, - 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, - 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBD, 0xBC, 0xBC, 0xBC, 0xBC, - 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, - 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, 0xBC, - 0xBC, 0xBC, 0xBC, 0xBC, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, - 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, - 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, 0xBF, - 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, - 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, - 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xBE, 0xB9, 0xB9, 0xB9, 0xB9, - 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, - 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, 0xB9, - 0xB9, 0xB9, 0xB9, 0xB9, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, - 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, - 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, 0xB8, - 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, - 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, - 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBB, 0xBA, 0xBA, 0xBA, 0xBA, - 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, - 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, 0xBA, - 0xBA, 0xBA, 0xBA, 0xBA, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, - 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, - 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, - 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, - 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, - 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA5, 0xA4, 0xA4, 0xA4, 0xA4, - 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, - 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, - 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, - 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, - 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, 0xA4, - 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, - 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, - 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, - 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, - 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, 0xA7, - 0xA7, 0xA7, 0xA7, 0xA7, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, - 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, - 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, - 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, - 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, - 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA6, 0xA1, 0xA1, 0xA1, 0xA1, - 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, - 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, - 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, - 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, - 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, - 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, - 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, - 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, - 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, - 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, - 0xA0, 0xA0, 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0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, + 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, + 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, 0xA1, + 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, + 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, + 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, + 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, + 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, 0xA0, + 0xA0, 0xA0, 0xA0, 0xA0, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, + 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, + 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, + 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, + 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, + 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA3, 0xA2, 0xA2, 0xA2, 0xA2, + 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, + 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, + 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, + 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, + 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, 0xA2, + 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, + 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, + 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, + 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, + 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, 0xAD, + 0xAD, 0xAD, 0xAD, 0xAD, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, + 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, + 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, + 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, + 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, + 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAC, 0xAF, 0xAF, 0xAF, 0xAF, + 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, + 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, + 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, + 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, + 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, 0xAF, + 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, + 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, + 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, + 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, + 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, 0xAE, + 0xAE, 0xAE, 0xAE, 0xAE, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, + 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, + 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, + 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, + 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, + 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA9, 0xA8, 0xA8, 0xA8, 0xA8, + 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, + 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, + 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, + 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, + 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, 0xA8, + 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, + 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, + 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, + 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, + 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, 0xAB, + 0xAB, 0xAB, 0xAB, 0xAB, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, + 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, + 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, + 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, + 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0xAA, 0x2A } ; /* alaw_encode */ @@ -229,17 +229,17 @@ static int from_alaw(void* in, void* out, int len, int bps, int format) // Make sure the input parametrs are OK if(format & (AF_FORMAT_SPECIAL_MASK | AF_FORMAT_US)) return AF_ERROR; - + // Convert to int or to float if((format & AF_FORMAT_POINT_MASK) == AF_FORMAT_I){ switch(bps){ - case(1): + case(1): for(i=0;i<len;i++){ if(((int8_t*)in)[i] & 0x80) ((int8_t*)out)[i] = (-1 * alaw_decode[(((int8_t*)in)[i]) & 0x7F]) >> 8; else ((int8_t*)out)[i] = (alaw_decode[(((int8_t*)in)[i]) & 0x7F]) >> 8; - } + } break; case(2): for(i=0;i<len;i++){ @@ -247,7 +247,7 @@ static int from_alaw(void* in, void* out, int len, int bps, int format) ((int16_t*)out)[i] = -1 * alaw_decode[(((int8_t*)in)[i]) & 0x7F]; else ((int16_t*)out)[i] = alaw_decode[(((int8_t*)in)[i]) & 0x7F]; - } + } break; case(4): for(i=0;i<len;i++){ @@ -255,7 +255,7 @@ static int from_alaw(void* in, void* out, int len, int bps, int format) ((int32_t*)out)[i] = (-1 * alaw_decode[(((int8_t*)in)[i]) & 0x7F]) << 16; else ((int32_t*)out)[i] = (alaw_decode[(((int8_t*)in)[i]) & 0x7F]) << 16; - } + } break; default: return AF_ERROR; @@ -279,11 +279,11 @@ static int to_alaw(void* in, void* out, int len, int bps, int format) // Make sure the input parametrs are OK if(format & (AF_FORMAT_SPECIAL_MASK | AF_FORMAT_US)) return AF_ERROR; - + // Convert from int or to float if((format & AF_FORMAT_POINT_MASK) == AF_FORMAT_I){ switch(bps){ - case(1): + case(1): for(i=0;i<len;i++){ if(((int8_t*)in)[i] >= 0) ((int8_t*)out)[i] = alaw_encode[((int8_t*)in)[i] << 4]; @@ -291,7 +291,7 @@ static int to_alaw(void* in, void* out, int len, int bps, int format) ((int8_t*)out)[i] = 0x7F & alaw_encode[-((int8_t*)in)[i] << 4]; } break; - case(2): + case(2): for(i=0;i<len;i++){ if(((int16_t*)in)[i] >= 0) ((int8_t*)out)[i] = alaw_encode[((int16_t*)in)[i] / 16]; @@ -299,7 +299,7 @@ static int to_alaw(void* in, void* out, int len, int bps, int format) ((int8_t*)out)[i] = 0x7F & alaw_encode[((int16_t*)in)[i] / -16]; } break; - case(4): + case(4): for(i=0;i<len;i++){ if(((int32_t*)in)[i] >= 0) ((int8_t*)out)[i] = alaw_encode[((int32_t*)in)[i] >> (16 + 4)]; diff --git a/libaf/af_format_ulaw.h b/libaf/af_format_ulaw.h index ecf521398c..5167a1593a 100644 --- a/libaf/af_format_ulaw.h +++ b/libaf/af_format_ulaw.h @@ -29,707 +29,707 @@ #include "af.h" // Conversion tables (the function are below) static short ulaw_decode[128] = -{ -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, - -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, - -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, - -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, - -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, - -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, - -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, - -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, - -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, - -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, - -876, -844, -812, -780, -748, -716, -684, -652, - -620, -588, -556, -524, -492, -460, -428, -396, - -372, -356, -340, -324, -308, -292, -276, -260, - -244, -228, -212, -196, -180, -164, -148, -132, - -120, -112, -104, -96, -88, -80, -72, -64, - -56, -48, -40, -32, -24, -16, -8, 0, +{ -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, + -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, + -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, + -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, + -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, + -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, + -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, + -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, + -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, + -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, + -876, -844, -812, -780, -748, -716, -684, -652, + -620, -588, -556, -524, -492, -460, -428, -396, + -372, -356, -340, -324, -308, -292, -276, -260, + -244, -228, -212, -196, -180, -164, -148, -132, + -120, -112, -104, -96, -88, -80, -72, -64, + -56, -48, -40, -32, -24, -16, -8, 0, } ; static unsigned char ulaw_encode[8193] = -{ 0xFF, 0xFE, 0xFE, 0xFD, 0xFD, 0xFC, 0xFC, 0xFB, 0xFB, 0xFA, 0xFA, 0xF9, - 0xF9, 0xF8, 0xF8, 0xF7, 0xF7, 0xF6, 0xF6, 0xF5, 0xF5, 0xF4, 0xF4, 0xF3, - 0xF3, 0xF2, 0xF2, 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int bps, int format) // Make sure the input parametrs are OK if(format & (AF_FORMAT_SPECIAL_MASK | AF_FORMAT_US)) return AF_ERROR; - + // Convert to int or to float if((format & AF_FORMAT_POINT_MASK) == AF_FORMAT_I){ switch(bps){ - case(1): + case(1): for(i=0;i<len;i++){ if(((int8_t*)in)[i] & 0x80) ((int8_t*)out)[i] = (-1 * ulaw_decode[(((int8_t*)in)[i]) & 0x7F]) >> 8; else ((int8_t*)out)[i] = (ulaw_decode[(((int8_t*)in)[i]) & 0x7F]) >> 8; - } + } break; case(2): for(i=0;i<len;i++){ @@ -759,7 +759,7 @@ static int from_ulaw(void* in, void* out, int len, int bps, int format) ((int16_t*)out)[i] = -1 * ulaw_decode[(((int8_t*)in)[i]) & 0x7F]; else ((int16_t*)out)[i] = ulaw_decode[(((int8_t*)in)[i]) & 0x7F]; - } + } break; case(4): for(i=0;i<len;i++){ @@ -767,7 +767,7 @@ static int from_ulaw(void* in, void* out, int len, int bps, int format) ((int32_t*)out)[i] = (-1 * ulaw_decode[(((int8_t*)in)[i]) & 0x7F]) << 16; else ((int32_t*)out)[i] = (ulaw_decode[(((int8_t*)in)[i]) & 0x7F]) << 16; - } + } break; default: return AF_ERROR; @@ -791,11 +791,11 @@ static int to_ulaw(void* in, void* out, int len, int bps, int format) // Make sure the input parametrs are OK if(format & (AF_FORMAT_SPECIAL_MASK | AF_FORMAT_US)) return AF_ERROR; - + // Convert from int or to float if((format & AF_FORMAT_POINT_MASK) == AF_FORMAT_I){ switch(bps){ - case(1): + case(1): for(i=0;i<len;i++){ if(((int8_t*)in)[i] >= 0) ((int8_t*)out)[i] = ulaw_encode[((int8_t*)in)[i] << 6]; @@ -803,7 +803,7 @@ static int to_ulaw(void* in, void* out, int len, int bps, int format) ((int8_t*)out)[i] = 0x7F & ulaw_encode[-((int8_t*)in)[i] << 6]; } break; - case(2): + case(2): for(i=0;i<len;i++){ if(((int16_t*)in)[i] >= 0) ((int8_t*)out)[i] = ulaw_encode[((int16_t*)in)[i] / 4]; @@ -811,7 +811,7 @@ static int to_ulaw(void* in, void* out, int len, int bps, int format) ((int8_t*)out)[i] = 0x7F & ulaw_encode[((int16_t*)in)[i] / -4]; } break; - case(4): + case(4): for(i=0;i<len;i++){ if(((int32_t*)in)[i] >= 0) ((int8_t*)out)[i] = ulaw_encode[((int32_t*)in)[i] >> (16 + 2)]; diff --git a/libaf/af_gate.c b/libaf/af_gate.c index 26c7e8ad03..38db9ff055 100644 --- a/libaf/af_gate.c +++ b/libaf/af_gate.c @@ -20,7 +20,7 @@ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include <inttypes.h> #include <math.h> @@ -43,19 +43,19 @@ typedef struct af_gate_s // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_gate_t* s = (af_gate_t*)af->setup; + af_gate_t* s = (af_gate_t*)af->setup; switch(cmd){ case AF_CONTROL_REINIT: // Sanity check if(!arg) return AF_ERROR; - + af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; // Time constant set to 0.1s - // s->alpha = (1.0/0.2)/(2.0*M_PI*(float)((af_data_t*)arg)->rate); + // s->alpha = (1.0/0.2)/(2.0*M_PI*(float)((af_data_t*)arg)->rate); return af_test_output(af,(af_data_t*)arg); case AF_CONTROL_COMMAND_LINE:{ /* float v=-10.0; */ @@ -75,10 +75,10 @@ static int control(struct af_instance_s* af, int cmd, void* arg) case AF_CONTROL_GATE_ON_OFF | AF_CONTROL_SET: memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int)); - return AF_OK; + return AF_OK; case AF_CONTROL_GATE_ON_OFF | AF_CONTROL_GET: memcpy((int*)arg,s->enable,AF_NCH*sizeof(int)); - return AF_OK; + return AF_OK; case AF_CONTROL_GATE_THRESH | AF_CONTROL_SET: return af_from_dB(AF_NCH,(float*)arg,s->tresh,20.0,-60.0,-1.0); case AF_CONTROL_GATE_THRESH | AF_CONTROL_GET: @@ -99,7 +99,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -116,7 +116,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) float* a = (float*)c->audio; // Audio data int len = c->len/4; // Number of samples int ch = 0; // Channel counter - register int nch = c->nch; // Number of channels + register int nch = c->nch; // Number of channels register int i = 0; @@ -126,8 +126,8 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) float t = 1.0 - s->time[ch]; for(i=ch;i<len;i+=nch){ register float x = a[i]; - register float pow = x*x; - s->pow[ch] = t*s->pow[ch] + + register float pow = x*x; + s->pow[ch] = t*s->pow[ch] + pow*s->time[ch]; // LP filter if(pow < s->pow[ch]){ ; diff --git a/libaf/af_hrtf.c b/libaf/af_hrtf.c index 8f86f5d854..30af96ad35 100644 --- a/libaf/af_hrtf.c +++ b/libaf/af_hrtf.c @@ -75,7 +75,7 @@ typedef struct af_hrtf_s { * nk: length of the convolution kernel * sx: ring buffer * sk: convolution kernel - * offset: offset on the ring buffer, can be + * offset: offset on the ring buffer, can be */ static float conv(const int nx, const int nk, const float *sx, const float *sk, const int offset) @@ -338,7 +338,7 @@ static int control(struct af_instance_s *af, int cmd, void* arg) } s->print_flag = 1; return AF_OK; - } + } return AF_UNKNOWN; } @@ -429,7 +429,7 @@ static af_data_t* play(struct af_instance_s *af, af_data_t *data) "[hrtf] bogus decode_mode: %d\n", s->decode_mode); break; } - + if(s->matrix_mode) mp_msg(MSGT_AFILTER, MSGL_INFO, "[hrtf] Using active matrix to decode rear center " @@ -439,20 +439,20 @@ static af_data_t* play(struct af_instance_s *af, af_data_t *data) out = af->data->audio; /* MPlayer's 5 channel layout (notation for the variable): - * + * * 0: L (LF), 1: R (RF), 2: Ls (LR), 3: Rs (RR), 4: C (CF), matrix * encoded: Cs (CR) - * + * * or: L = left, C = center, R = right, F = front, R = rear - * + * * Filter notation: - * + * * CF * OF AF * Ear-> * OR AR * CR - * + * * or: C = center, A = same side, O = opposite, F = front, R = rear */ @@ -667,7 +667,7 @@ static int af_open(af_instance_t* af) } for(i = 0; i < s->basslen; i++) s->ba_ir[i] *= BASSGAIN; - + return AF_OK; } diff --git a/libaf/af_karaoke.c b/libaf/af_karaoke.c index 5239c74721..e0f0d42471 100644 --- a/libaf/af_karaoke.c +++ b/libaf/af_karaoke.c @@ -23,7 +23,7 @@ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include "af.h" @@ -43,7 +43,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -55,22 +55,22 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) { af_data_t* c = data; // Current working data float* a = c->audio; // Audio data - int len = c->len/4; // Number of samples in current audio block + int len = c->len/4; // Number of samples in current audio block int nch = c->nch; // Number of channels register int i; - /* - FIXME1 add a low band pass filter to avoid suppressing + /* + FIXME1 add a low band pass filter to avoid suppressing centered bass/drums FIXME2 better calculated* attenuation factor */ - + for(i=0;i<len;i+=nch) { a[i] = (a[i] - a[i+1]) * 0.7; a[i+1]=a[i]; } - + return c; } @@ -84,7 +84,7 @@ static int af_open(af_instance_t* af){ if(af->data == NULL) return AF_ERROR; - + return AF_OK; } diff --git a/libaf/af_ladspa.c b/libaf/af_ladspa.c index 34cf0a6d19..e343d1ded7 100644 --- a/libaf/af_ladspa.c +++ b/libaf/af_ladspa.c @@ -212,7 +212,7 @@ static int af_ladspa_parse_plugin(af_ladspa_t *setup) { } if (setup->ninputs == 0) { - mp_msg(MSGT_AFILTER, MSGL_WARN, "%s: %s\n", setup->myname, + mp_msg(MSGT_AFILTER, MSGL_WARN, "%s: %s\n", setup->myname, MSGTR_AF_LADSPA_WarnNoInputs); } else if (setup->ninputs == 1) { mp_msg(MSGT_AFILTER, MSGL_V, "%s: this is a mono effect\n", setup->myname); @@ -281,7 +281,7 @@ static int af_ladspa_parse_plugin(af_ladspa_t *setup) { * filename (with or without the addition of the .so extension) in various * directories specified by the LADSPA_PATH environment variable. If all fails * it tries the filename directly as an absolute path to the library. - * + * * \param filename filename of the library to load. * \param flag see dlopen(3) for a description of the flags. * @@ -411,7 +411,7 @@ static int af_ladspa_load_plugin(af_ladspa_t *setup) { /* if label == help, list all labels in library and exit */ if (strcmp(setup->label, "help") == 0) { - mp_msg(MSGT_AFILTER, MSGL_INFO, "%s: %s %s:\n", setup->myname, + mp_msg(MSGT_AFILTER, MSGL_INFO, "%s: %s %s:\n", setup->myname, MSGTR_AF_LADSPA_AvailableLabels, setup->file); for (i=0; ; i++) { ladspa_descriptor = descriptor_function(i); @@ -558,7 +558,7 @@ static int control(struct af_instance_s *af, int cmd, void *arg) { mp_msg(MSGT_AFILTER, MSGL_V, "%s: label --> %s\n", setup->myname, setup->label); /* if (*(char*)arg != '0') arg++; */ /* read ':' */ - + free(buf); /* no longer needed */ /* set new setup->myname */ @@ -716,7 +716,7 @@ static void uninit(struct af_instance_s *af) { /* ------------------------------------------------------------------------- */ /** \brief Process chunk of audio data through the selected LADSPA Plugin. - * + * * \param af Pointer to audio filter instance * \param data Pointer to chunk of audio data * @@ -730,7 +730,7 @@ static af_data_t* play(struct af_instance_s *af, af_data_t *data) { int nsamples = data->len/4; /* /4 because it's 32-bit float */ int nch = data->nch; int rate = data->rate; - int i, p; + int i, p; if (setup->status !=AF_OK) return data; @@ -845,7 +845,7 @@ static af_data_t* play(struct af_instance_s *af, af_data_t *data) { * and right. connect it to the second port. */ - for (p = i; p % setup->ninputs; p++) { + for (p = i; p % setup->ninputs; p++) { pdes->connect_port(setup->chhandles[i-1], setup->inputs[p % setup->ninputs], setup->inbufs[i-1]); diff --git a/libaf/af_lavcresample.c b/libaf/af_lavcresample.c index 2dbdf786b9..49a284bba6 100644 --- a/libaf/af_lavcresample.c +++ b/libaf/af_lavcresample.c @@ -34,7 +34,7 @@ typedef struct af_resample_s{ int16_t *in[AF_NCH]; int in_alloc; int index; - + int filter_length; int linear; int phase_shift; @@ -45,7 +45,7 @@ typedef struct af_resample_s{ // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_resample_t* s = (af_resample_t*)af->setup; + af_resample_t* s = (af_resample_t*)af->setup; af_data_t *data= (af_data_t*)arg; int out_rate, test_output_res; // helpers for checking input format @@ -83,7 +83,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -101,7 +101,7 @@ static void uninit(struct af_instance_s* af) // Filter data through filter static af_data_t* play(struct af_instance_s* af, af_data_t* data) -{ +{ af_resample_t *s = af->setup; int i, j, consumed, ret; int16_t *in = (int16_t*)data->audio; @@ -110,14 +110,14 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) int in_len = data->len/(2*chans); int out_len = in_len * af->mul + 10; int16_t tmp[AF_NCH][out_len]; - + if(AF_OK != RESIZE_LOCAL_BUFFER(af,data)) return NULL; - + out= (int16_t*)af->data->audio; - + out_len= min(out_len, af->data->len/(2*chans)); - + if(s->in_alloc < in_len + s->index){ s->in_alloc= in_len + s->index; for(i=0; i<chans; i++){ @@ -145,7 +145,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans); } out_len= ret; - + s->index= in_len - consumed; for(i=0; i<chans; i++){ memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t)); diff --git a/libaf/af_pan.c b/libaf/af_pan.c index 2c1a41dd87..9a3e520446 100644 --- a/libaf/af_pan.c +++ b/libaf/af_pan.c @@ -37,7 +37,7 @@ typedef struct af_pan_s // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_pan_t* s = af->setup; + af_pan_t* s = af->setup; switch(cmd){ case AF_CONTROL_REINIT: @@ -50,7 +50,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) af->data->nch = s->nch ? s->nch: ((af_data_t*)arg)->nch; af->mul = (double)af->data->nch / ((af_data_t*)arg)->nch; - if((af->data->format != ((af_data_t*)arg)->format) || + if((af->data->format != ((af_data_t*)arg)->format) || (af->data->bps != ((af_data_t*)arg)->bps)){ ((af_data_t*)arg)->format = af->data->format; ((af_data_t*)arg)->bps = af->data->bps; @@ -72,12 +72,12 @@ static int control(struct af_instance_s* af, int cmd, void* arg) j = 0; k = 0; while((*cp == ':') && (k < AF_NCH)){ sscanf(cp, ":%f%n" , &s->level[j][k], &n); - mp_msg(MSGT_AFILTER, MSGL_V, "[pan] Pan level from channel %i to" + mp_msg(MSGT_AFILTER, MSGL_V, "[pan] Pan level from channel %i to" " channel %i = %f\n",k,j,s->level[j][k]); cp =&cp[n]; j++; if(j>=nch){ - j = 0; + j = 0; k++; } } @@ -108,7 +108,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) // Sanity check if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH){ - mp_msg(MSGT_AFILTER, MSGL_ERR, "[pan] The number of output channels must be" + mp_msg(MSGT_AFILTER, MSGL_ERR, "[pan] The number of output channels must be" " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]); return AF_ERROR; } @@ -138,7 +138,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -165,7 +165,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) return NULL; out = l->audio; - // Execute panning + // Execute panning // FIXME: Too slow while(in < end){ for(j=0;j<ncho;j++){ diff --git a/libaf/af_resample.c b/libaf/af_resample.c index 03ad4e3d42..ac39f36faf 100644 --- a/libaf/af_resample.c +++ b/libaf/af_resample.c @@ -34,11 +34,11 @@ length of the filter. This definition affects the computational complexity (see play()), the performance (see filter.h) and the memory usage. The filter length is chosen to 8 if the machine is - slow and to 16 if the machine is fast and has MMX. + slow and to 16 if the machine is fast and has MMX. */ #if !HAVE_MMX // This machine is slow -#define L8 +#define L8 #else #define L16 #endif @@ -47,7 +47,7 @@ // Filtering types #define RSMP_LIN (0<<0) // Linear interpolation -#define RSMP_INT (1<<0) // 16 bit integer +#define RSMP_INT (1<<0) // 16 bit integer #define RSMP_FLOAT (2<<0) // 32 bit floating point #define RSMP_MASK (3<<0) @@ -66,9 +66,9 @@ typedef struct af_resample_s void** xq; // Circular buffers uint32_t xi; // Index for circular buffers uint32_t wi; // Index for w - uint32_t i; // Number of new samples to put in x queue + uint32_t i; // Number of new samples to put in x queue uint32_t dn; // Down sampling factor - uint32_t up; // Up sampling factor + uint32_t up; // Up sampling factor uint64_t step; // Step size for linear interpolation uint64_t pt; // Pointer remainder for linear interpolation int setup; // Setup parameters cmdline or through postcreate @@ -79,7 +79,7 @@ static int linint(af_data_t* c,af_data_t* l, af_resample_t* s) { uint32_t len = 0; // Number of input samples uint32_t nch = l->nch; // Words pre transfer - uint64_t step = s->step; + uint64_t step = s->step; int16_t* in16 = ((int16_t*)c->audio); int16_t* out16 = ((int16_t*)l->audio); int32_t* in32 = ((int32_t*)c->audio); @@ -87,35 +87,35 @@ static int linint(af_data_t* c,af_data_t* l, af_resample_t* s) uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY); uint64_t pt = s->pt; uint16_t tmp; - + switch (nch){ case 1: while(pt < end){ - out16[len++]=in16[pt>>STEPACCURACY]; + out16[len++]=in16[pt>>STEPACCURACY]; pt+=step; } s->pt=pt & ((1LL<<STEPACCURACY)-1); - break; + break; case 2: end/=2; while(pt < end){ - out32[len++]=in32[pt>>STEPACCURACY]; + out32[len++]=in32[pt>>STEPACCURACY]; pt+=step; } len=(len<<1); s->pt=pt & ((1LL<<STEPACCURACY)-1); break; - default: + default: end /=nch; while(pt < end){ tmp=nch; - do { - tmp--; - out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch]; + do { + tmp--; + out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch]; } while (tmp); len+=nch; pt+=step; - } + } s->pt=pt & ((1LL<<STEPACCURACY)-1); } return len; @@ -128,13 +128,13 @@ static int set_types(struct af_instance_s* af, af_data_t* data) int rv = AF_OK; float rd = 0; - // Make sure this filter isn't redundant + // Make sure this filter isn't redundant if((af->data->rate == data->rate) || (af->data->rate == 0)) return AF_DETACH; /* If sloppy and small resampling difference (2%) */ rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate; - if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) && - (data->format != (AF_FORMAT_FLOAT_NE))) || + if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) && + (data->format != (AF_FORMAT_FLOAT_NE))) || ((s->setup & RSMP_MASK) == RSMP_LIN)){ s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN; af->data->format = AF_FORMAT_S16_NE; @@ -144,7 +144,7 @@ static int set_types(struct af_instance_s* af, af_data_t* data) else{ /* If the input format is float or if float is explicitly selected use float, otherwise use int */ - if((data->format == (AF_FORMAT_FLOAT_NE)) || + if((data->format == (AF_FORMAT_FLOAT_NE)) || ((s->setup & RSMP_MASK) == RSMP_FLOAT)){ s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT; af->data->format = AF_FORMAT_FLOAT_NE; @@ -174,7 +174,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) { switch(cmd){ case AF_CONTROL_REINIT:{ - af_resample_t* s = af->setup; + af_resample_t* s = af->setup; af_data_t* n = arg; // New configuration int i,d = 0; int rv = AF_OK; @@ -188,8 +188,8 @@ static int control(struct af_instance_s* af, int cmd, void* arg) if(AF_DETACH == (rv = set_types(af,n))) return AF_DETACH; - - // If linear interpolation + + // If linear interpolation if((s->setup & RSMP_MASK) == RSMP_LIN){ s->pt=0LL; s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL; @@ -227,11 +227,11 @@ static int control(struct af_instance_s* af, int cmd, void* arg) float* wt; float fc; int j; - s->up = af->data->rate/d; + s->up = af->data->rate/d; s->dn = n->rate/d; s->wi = 0; s->i = 0; - + // Calculate cutoff frequency for filter fc = 1/(float)(max(s->up,s->dn)); // Allocate space for polyphase filter bank and prototype filter @@ -241,7 +241,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) s->w = malloc(L*s->up*af->data->bps); // Design prototype filter type using Kaiser window with beta = 10 - if(NULL == w || NULL == s->w || + if(NULL == w || NULL == s->w || -1 == af_filter_design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){ mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] Unable to design prototype filter.\n"); return AF_ERROR; @@ -270,39 +270,39 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return rv; } case AF_CONTROL_COMMAND_LINE:{ - af_resample_t* s = af->setup; + af_resample_t* s = af->setup; int rate=0; int type=RSMP_INT; int sloppy=1; sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type); - s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) | + s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) | (clamp(type,RSMP_LIN,RSMP_FLOAT)); return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate); } - case AF_CONTROL_POST_CREATE: + case AF_CONTROL_POST_CREATE: if((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) ((af_resample_t*)af->setup)->setup = RSMP_FLOAT; return AF_OK; - case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET: + case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET: // Reinit must be called after this function has been called - + // Sanity check if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){ - mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] The output sample frequency " + mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] The output sample frequency " "must be between 8kHz and 192kHz. Current value is %i \n", ((int*)arg)[0]); return AF_ERROR; } - af->data->rate=((int*)arg)[0]; - mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Changing sample rate " + af->data->rate=((int*)arg)[0]; + mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Changing sample rate " "to %iHz\n",af->data->rate); return AF_OK; } return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { af_resample_t *s = af->setup; @@ -335,7 +335,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) if(s->up>s->dn){ # define UP # include "af_resample_template.c" -# undef UP +# undef UP } else{ # define DN @@ -349,7 +349,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) if(s->up>s->dn){ # define UP # include "af_resample_template.c" -# undef UP +# undef UP } else{ # define DN @@ -366,7 +366,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) c->audio = l->audio; c->len = len*l->bps; c->rate = l->rate; - + return c; } diff --git a/libaf/af_resample_template.c b/libaf/af_resample_template.c index 03b0f1a78b..4d4c5922ca 100644 --- a/libaf/af_resample_template.c +++ b/libaf/af_resample_template.c @@ -21,8 +21,8 @@ /* This file contains the resampling engine, the sample format is controlled by the FORMAT parameter, the filter length by the L parameter and the resampling type by UP and DN. This file should - only be included by af_resample.c -*/ + only be included by af_resample.c +*/ #undef L #undef SHIFT @@ -46,16 +46,16 @@ #if defined(FORMAT_I) #define SHIFT >>16 #define FORMAT int16_t -#else +#else #define SHIFT #define FORMAT float #endif // Short filter -#if defined(L8) +#if defined(L8) #define L 8 // Filter length -// Unrolled loop to speed up execution +// Unrolled loop to speed up execution #define FIR(x,w,y) \ (y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \ + w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) SHIFT @@ -65,7 +65,7 @@ #else /* L8/L16 */ #define L 16 -// Unrolled loop to speed up execution +// Unrolled loop to speed up execution #define FIR(x,w,y) \ y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \ + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \ @@ -74,7 +74,7 @@ #endif /* L8/L16 */ -// Macro to add data to circular que +// Macro to add data to circular que #define ADDQUE(xi,xq,in)\ xq[xi]=xq[(xi)+L]=*(in);\ xi=((xi)-1)&(L-1); @@ -83,15 +83,15 @@ uint32_t ci = l->nch; // Index for channels uint32_t nch = l->nch; // Number of channels - uint32_t inc = s->up/s->dn; - uint32_t level = s->up%s->dn; + uint32_t inc = s->up/s->dn; + uint32_t level = s->up%s->dn; uint32_t up = s->up; uint32_t dn = s->dn; uint32_t ns = c->len/l->bps; register FORMAT* w = s->w; register uint32_t wi = 0; - register uint32_t xi = 0; + register uint32_t xi = 0; // Index current channel while(ci--){ @@ -126,8 +126,8 @@ #if defined(DN) /* DN */ uint32_t ci = l->nch; // Index for channels uint32_t nch = l->nch; // Number of channels - uint32_t inc = s->dn/s->up; - uint32_t level = s->dn%s->up; + uint32_t inc = s->dn/s->up; + uint32_t level = s->dn%s->up; uint32_t up = s->up; uint32_t dn = s->dn; uint32_t ns = c->len/l->bps; @@ -136,7 +136,7 @@ register int32_t i = 0; register uint32_t wi = 0; register uint32_t xi = 0; - + // Index current channel while(ci--){ // Temporary pointers @@ -156,7 +156,7 @@ len++; out+=nch; // Update wi to point at the correct polyphase component - wi=(wi+dn)%up; + wi=(wi+dn)%up; // Insert i number of new samples in queue i = inc; diff --git a/libaf/af_sinesuppress.c b/libaf/af_sinesuppress.c index b7ed277d7b..56f6076b5c 100644 --- a/libaf/af_sinesuppress.c +++ b/libaf/af_sinesuppress.c @@ -22,7 +22,7 @@ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include <inttypes.h> #include <math.h> @@ -47,13 +47,13 @@ static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data); // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_sinesuppress_t* s = (af_sinesuppress_t*)af->setup; + af_sinesuppress_t* s = (af_sinesuppress_t*)af->setup; switch(cmd){ case AF_CONTROL_REINIT:{ // Sanity check if(!arg) return AF_ERROR; - + af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = 1; #if 0 @@ -95,7 +95,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -111,7 +111,7 @@ static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data) register int i = 0; int16_t *a = (int16_t*)data->audio; // Audio data int len = data->len/2; // Number of samples - + for (i = 0; i < len; i++) { double co= cos(s->pos); @@ -143,14 +143,14 @@ static af_data_t* play_float(struct af_instance_s* af, af_data_t* data) float *a = (float*)data->audio; // Audio data int len = data->len/4; // Number of samples float avg, l, r; - + for (i = 0; i < len; i+=2) { avg = (a[i] + a[i + 1]) / 2; - + /* l = avg + (s->mul * (a[i] - avg)); r = avg + (s->mul * (a[i + 1] - avg));*/ - + a[i] = af_softclip(l); a[i + 1] = af_softclip(r); } diff --git a/libaf/af_sub.c b/libaf/af_sub.c index 294c6f788e..ada1950649 100644 --- a/libaf/af_sub.c +++ b/libaf/af_sub.c @@ -30,7 +30,7 @@ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include "af.h" #include "dsp.h" @@ -38,7 +38,7 @@ // Q value for low-pass filter #define Q 1.0 -// Analog domain biquad section +// Analog domain biquad section typedef struct{ float a[3]; // Numerator coefficients float b[3]; // Denominator coefficients @@ -56,13 +56,13 @@ typedef struct af_sub_s float fc; // Cutoff frequency [Hz] for low-pass filter float k; // Filter gain; int ch; // Channel number which to insert the filtered data - + }af_sub_t; // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_sub_t* s = af->setup; + af_sub_t* s = af->setup; switch(cmd){ case AF_CONTROL_REINIT:{ @@ -120,7 +120,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -146,7 +146,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) af_data_t* c = data; // Current working data af_sub_t* s = af->setup; // Setup for this instance float* a = c->audio; // Audio data - int len = c->len/4; // Number of samples in current audio block + int len = c->len/4; // Number of samples in current audio block int nch = c->nch; // Number of channels int ch = s->ch; // Channel in which to insert the sub audio register int i; diff --git a/libaf/af_surround.c b/libaf/af_surround.c index a81eb40427..098234b364 100644 --- a/libaf/af_surround.c +++ b/libaf/af_surround.c @@ -118,14 +118,14 @@ static int control(struct af_instance_s* af, int cmd, void* arg) s->dr = calloc(LD,af->data->bps); if((NULL == s->dl) || (NULL == s->dr)) mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n"); - + // Initialize delay queue index if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0)) return AF_ERROR; // printf("%i\n",s->wi); s->ri = 0; - if((af->data->format != ((af_data_t*)arg)->format) || + if((af->data->format != ((af_data_t*)arg)->format) || (af->data->bps != ((af_data_t*)arg)->bps)){ ((af_data_t*)arg)->format = af->data->format; ((af_data_t*)arg)->bps = af->data->bps; @@ -171,7 +171,7 @@ static float steering_matrix[][12] = { // Filter data through filter static af_data_t* play(struct af_instance_s* af, af_data_t* data){ af_surround_t* s = (af_surround_t*)af->setup; - float* m = steering_matrix[0]; + float* m = steering_matrix[0]; float* in = data->audio; // Input audio data float* out = NULL; // Output audio data float* end = in + data->len / sizeof(float); // Loop end @@ -236,10 +236,10 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data){ #endif // Next sample... - in = &in[data->nch]; + in = &in[data->nch]; out = &out[af->data->nch]; } - + // Save indexes s->i = i; s->ri = ri; s->wi = wi; diff --git a/libaf/af_sweep.c b/libaf/af_sweep.c index dab6895b0f..782bf7ec46 100644 --- a/libaf/af_sweep.c +++ b/libaf/af_sweep.c @@ -36,7 +36,7 @@ typedef struct af_sweep_s{ // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_sweept* s = (af_sweept*)af->setup; + af_sweept* s = (af_sweept*)af->setup; af_data_t *data= (af_data_t*)arg; switch(cmd){ @@ -57,7 +57,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -70,7 +70,7 @@ static void uninit(struct af_instance_s* af) // Filter data through filter static af_data_t* play(struct af_instance_s* af, af_data_t* data) -{ +{ af_sweept *s = af->setup; int i, j; int16_t *in = (int16_t*)data->audio; diff --git a/libaf/af_tools.c b/libaf/af_tools.c index 556429f4d9..82e0940fb2 100644 --- a/libaf/af_tools.c +++ b/libaf/af_tools.c @@ -17,16 +17,16 @@ */ #include <math.h> -#include <string.h> +#include <string.h> #include "af.h" /* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if fail */ int af_from_dB(int n, float* in, float* out, float k, float mi, float ma) { - int i = 0; + int i = 0; // Sanity check - if(!in || !out) + if(!in || !out) return AF_ERROR; for(i=0;i<n;i++){ @@ -42,9 +42,9 @@ int af_from_dB(int n, float* in, float* out, float k, float mi, float ma) fail */ int af_to_dB(int n, float* in, float* out, float k) { - int i = 0; + int i = 0; // Sanity check - if(!in || !out) + if(!in || !out) return AF_ERROR; for(i=0;i<n;i++){ @@ -59,9 +59,9 @@ int af_to_dB(int n, float* in, float* out, float k) /* Convert from ms to sample time */ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma) { - int i = 0; + int i = 0; // Sanity check - if(!in || !out) + if(!in || !out) return AF_ERROR; for(i=0;i<n;i++) @@ -73,21 +73,21 @@ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma) /* Convert from sample time to ms */ int af_to_ms(int n, int* in, float* out, int rate) { - int i = 0; + int i = 0; // Sanity check - if(!in || !out || !rate) + if(!in || !out || !rate) return AF_ERROR; for(i=0;i<n;i++) out[i]=1000.0 * (float)in[i]/((float)rate); - + return AF_OK; } /* Helper function for testing the output format */ int af_test_output(struct af_instance_s* af, af_data_t* out) { - if((af->data->format != out->format) || + if((af->data->format != out->format) || (af->data->bps != out->bps) || (af->data->rate != out->rate) || (af->data->nch != out->nch)){ diff --git a/libaf/af_volnorm.c b/libaf/af_volnorm.c index 76cf9b1dd1..9bfaef6220 100644 --- a/libaf/af_volnorm.c +++ b/libaf/af_volnorm.c @@ -20,7 +20,7 @@ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include <inttypes.h> #include <math.h> @@ -80,16 +80,16 @@ typedef struct af_volume_s // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_volnorm_t* s = (af_volnorm_t*)af->setup; + af_volnorm_t* s = (af_volnorm_t*)af->setup; switch(cmd){ case AF_CONTROL_REINIT: // Sanity check if(!arg) return AF_ERROR; - + af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - + if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){ af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; @@ -113,7 +113,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -129,26 +129,26 @@ static void method1_int16(af_volnorm_t *s, af_data_t *c) int len = c->len/2; // Number of samples float curavg = 0.0, newavg, neededmul; int tmp; - + for (i = 0; i < len; i++) { tmp = data[i]; curavg += tmp * tmp; } curavg = sqrt(curavg / (float) len); - + // Evaluate an adequate 'mul' coefficient based on previous state, current // samples level, etc - + if (curavg > SIL_S16) { neededmul = s->mid_s16 / (curavg * s->mul); s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; - + // clamp the mul coefficient s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); } - + // Scale & clamp the samples for (i = 0; i < len; i++) { @@ -156,10 +156,10 @@ static void method1_int16(af_volnorm_t *s, af_data_t *c) tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); data[i] = tmp; } - + // Evaulation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; - + // Stores computed values for future smoothing s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; } @@ -170,33 +170,33 @@ static void method1_float(af_volnorm_t *s, af_data_t *c) float *data = (float*)c->audio; // Audio data int len = c->len/4; // Number of samples float curavg = 0.0, newavg, neededmul, tmp; - + for (i = 0; i < len; i++) { tmp = data[i]; curavg += tmp * tmp; } curavg = sqrt(curavg / (float) len); - + // Evaluate an adequate 'mul' coefficient based on previous state, current // samples level, etc - + if (curavg > SIL_FLOAT) // FIXME { neededmul = s->mid_float / (curavg * s->mul); s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; - + // clamp the mul coefficient s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); } - + // Scale & clamp the samples for (i = 0; i < len; i++) data[i] *= s->mul; - + // Evaulation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; - + // Stores computed values for future smoothing s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; } @@ -208,14 +208,14 @@ static void method2_int16(af_volnorm_t *s, af_data_t *c) int len = c->len/2; // Number of samples float curavg = 0.0, newavg, avg = 0.0; int tmp, totallen = 0; - + for (i = 0; i < len; i++) { tmp = data[i]; curavg += tmp * tmp; } curavg = sqrt(curavg / (float) len); - + // Evaluate an adequate 'mul' coefficient based on previous state, current // samples level, etc for (i = 0; i < NSAMPLES; i++) @@ -223,7 +223,7 @@ static void method2_int16(af_volnorm_t *s, af_data_t *c) avg += s->mem[i].avg * (float)s->mem[i].len; totallen += s->mem[i].len; } - + if (totallen > MIN_SAMPLE_SIZE) { avg /= (float)totallen; @@ -233,7 +233,7 @@ static void method2_int16(af_volnorm_t *s, af_data_t *c) s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); } } - + // Scale & clamp the samples for (i = 0; i < len; i++) { @@ -241,10 +241,10 @@ static void method2_int16(af_volnorm_t *s, af_data_t *c) tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); data[i] = tmp; } - + // Evaulation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; - + // Stores computed values for future smoothing s->mem[s->idx].len = len; s->mem[s->idx].avg = newavg; @@ -258,14 +258,14 @@ static void method2_float(af_volnorm_t *s, af_data_t *c) int len = c->len/4; // Number of samples float curavg = 0.0, newavg, avg = 0.0, tmp; int totallen = 0; - + for (i = 0; i < len; i++) { tmp = data[i]; curavg += tmp * tmp; } curavg = sqrt(curavg / (float) len); - + // Evaluate an adequate 'mul' coefficient based on previous state, current // samples level, etc for (i = 0; i < NSAMPLES; i++) @@ -273,7 +273,7 @@ static void method2_float(af_volnorm_t *s, af_data_t *c) avg += s->mem[i].avg * (float)s->mem[i].len; totallen += s->mem[i].len; } - + if (totallen > MIN_SAMPLE_SIZE) { avg /= (float)totallen; @@ -283,14 +283,14 @@ static void method2_float(af_volnorm_t *s, af_data_t *c) s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); } } - + // Scale & clamp the samples for (i = 0; i < len; i++) data[i] *= s->mul; - + // Evaulation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; - + // Stores computed values for future smoothing s->mem[s->idx].len = len; s->mem[s->idx].avg = newavg; @@ -310,7 +310,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) method1_int16(s, data); } else if(af->data->format == (AF_FORMAT_FLOAT_NE)) - { + { if (s->method) method2_float(s, data); else diff --git a/libaf/af_volume.c b/libaf/af_volume.c index 127db7b37a..59cdc5973e 100644 --- a/libaf/af_volume.c +++ b/libaf/af_volume.c @@ -31,12 +31,12 @@ probing is enable by AF_CONTROL_VOLUME_PROBE_ON_OFF and is done on a per channel basis. The result from the probing is obtained using AF_CONTROL_VOLUME_PROBE_GET and AF_CONTROL_VOLUME_PROBE_GET_MAX. The - probed values are calculated in dB. + probed values are calculated in dB. */ #include <stdio.h> #include <stdlib.h> -#include <string.h> +#include <string.h> #include <inttypes.h> #include <math.h> @@ -59,16 +59,16 @@ typedef struct af_volume_s // Initialization and runtime control static int control(struct af_instance_s* af, int cmd, void* arg) { - af_volume_t* s = (af_volume_t*)af->setup; + af_volume_t* s = (af_volume_t*)af->setup; switch(cmd){ case AF_CONTROL_REINIT: // Sanity check if(!arg) return AF_ERROR; - + af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - + if(s->fast && (((af_data_t*)arg)->format != (AF_FORMAT_FLOAT_NE))){ af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; @@ -91,22 +91,22 @@ static int control(struct af_instance_s* af, int cmd, void* arg) for(i=0;i<AF_NCH;i++) vol[i]=v; return control(af,AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET, vol); } - case AF_CONTROL_POST_CREATE: - s->fast = ((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == + case AF_CONTROL_POST_CREATE: + s->fast = ((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) ? 0 : 1; return AF_OK; case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_SET: memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int)); - return AF_OK; + return AF_OK; case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_GET: memcpy((int*)arg,s->enable,AF_NCH*sizeof(int)); - return AF_OK; + return AF_OK; case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_SET: s->soft = *(int*)arg; - return AF_OK; + return AF_OK; case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_GET: *(int*)arg = s->soft; - return AF_OK; + return AF_OK; case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET: return af_from_dB(AF_NCH,(float*)arg,s->level,20.0,-200.0,60.0); case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_GET: @@ -130,7 +130,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg) return AF_UNKNOWN; } -// Deallocate memory +// Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) @@ -145,7 +145,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) af_data_t* c = data; // Current working data af_volume_t* s = (af_volume_t*)af->setup; // Setup for this instance int ch = 0; // Channel counter - register int nch = c->nch; // Number of channels + register int nch = c->nch; // Number of channels register int i = 0; // Basic operation volume control only (used on slow machines) @@ -154,7 +154,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) int len = c->len/2; // Number of samples for(ch = 0; ch < nch ; ch++){ if(s->enable[ch]){ - register int vol = (int)(255.0 * s->level[ch]); + register int vol = (int)(255.0 * s->level[ch]); for(i=ch;i<len;i+=nch){ register int x = (a[i] * vol) >> 8; a[i]=clamp(x,SHRT_MIN,SHRT_MAX); @@ -163,7 +163,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) } } // Machine is fast and data is floating point - else if(af->data->format == (AF_FORMAT_FLOAT_NE)){ + else if(af->data->format == (AF_FORMAT_FLOAT_NE)){ float* a = (float*)c->audio; // Audio data int len = c->len/4; // Number of samples for(ch = 0; ch < nch ; ch++){ @@ -172,7 +172,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data) float t = 1.0 - s->time; for(i=ch;i<len;i+=nch){ register float x = a[i]; - register float pow = x*x; + register float pow = x*x; // Check maximum power value if(pow > s->max[ch]) s->max[ch] = pow; diff --git a/libaf/control.h b/libaf/control.h index 1d657b4a21..b6a1ca8262 100644 --- a/libaf/control.h +++ b/libaf/control.h @@ -22,12 +22,12 @@ #include <sys/types.h> /********************************************* -// Control info struct. +// Control info struct. // // This struct is the argument in a info call to a filter. */ -// Argument types +// Argument types #define AF_CONTROL_TYPE_BOOL (0x0<<0) #define AF_CONTROL_TYPE_CHAR (0x1<<0) #define AF_CONTROL_TYPE_INT (0x2<<0) @@ -52,9 +52,9 @@ typedef struct af_control_info_s{ int def; // Control enumrification char* name; // Name of argument char* info; // Description of what it does - int flags; // Flags as defined above - float max; // Max and min value - float min; // (only aplicable on float and int) + int flags; // Flags as defined above + float max; // Max and min value + float min; // (only aplicable on float and int) int xdim; // 1st dimension int ydim; // 2nd dimension (=0 for everything except matrix) size_t sz; // Size of argument in bytes @@ -73,10 +73,10 @@ typedef struct af_control_ext_s{ }af_control_ext_t; /********************************************* -// Control parameters +// Control parameters */ -/* The control system is divided into 3 levels +/* The control system is divided into 3 levels mandatory calls - all filters must answer to all of these optional calls - are optional filter specific calls - applies only to some filters @@ -115,12 +115,12 @@ typedef struct af_control_ext_s{ // FILTER SPECIFIC CALLS // Basic operations: These can be ored with any of the below calls -// Set argument +// Set argument #define AF_CONTROL_SET 0x00000000 // Get argument #define AF_CONTROL_GET 0x00000001 // Get info about the control, i.e fill in everything except argument -#define AF_CONTROL_INFO 0x00000002 +#define AF_CONTROL_INFO 0x00000002 // Resample @@ -153,8 +153,8 @@ typedef struct af_control_ext_s{ // Set make af_channels into a router #define AF_CONTROL_CHANNELS_ROUTER 0x00000A00 | AF_CONTROL_FILTER_SPECIFIC - -// Volume + +// Volume // Turn volume control on and off, arg is int* #define AF_CONTROL_VOLUME_ON_OFF 0x00000B00 | AF_CONTROL_FILTER_SPECIFIC @@ -165,10 +165,10 @@ typedef struct af_control_ext_s{ // Set volume level, arg is a float* with the volume for all the channels #define AF_CONTROL_VOLUME_LEVEL 0x00000D00 | AF_CONTROL_FILTER_SPECIFIC -// Probed power level for all channels, arg is a float* +// Probed power level for all channels, arg is a float* #define AF_CONTROL_VOLUME_PROBE 0x00000E00 | AF_CONTROL_FILTER_SPECIFIC -// Maximum probed power level for all channels, arg is a float* +// Maximum probed power level for all channels, arg is a float* #define AF_CONTROL_VOLUME_PROBE_MAX 0x00000F00 | AF_CONTROL_FILTER_SPECIFIC // Compressor/expander @@ -193,13 +193,13 @@ typedef struct af_control_ext_s{ // Turn noise gate on an off #define AF_CONTROL_GATE_ON_OFF 0x00001500 | AF_CONTROL_FILTER_SPECIFIC -// Noise gate threshold [dB] +// Noise gate threshold [dB] #define AF_CONTROL_GATE_THRESH 0x00001600 | AF_CONTROL_FILTER_SPECIFIC // Noise gate attack time [ms] #define AF_CONTROL_GATE_ATTACK 0x00001700 | AF_CONTROL_FILTER_SPECIFIC -// Noise gate release time [ms] +// Noise gate release time [ms] #define AF_CONTROL_GATE_RELEASE 0x00001800 | AF_CONTROL_FILTER_SPECIFIC // Noise gate release range level [dB] @@ -207,16 +207,16 @@ typedef struct af_control_ext_s{ // Pan -// Pan levels, arg is a control_ext with a float* +// Pan levels, arg is a control_ext with a float* #define AF_CONTROL_PAN_LEVEL 0x00001A00 | AF_CONTROL_FILTER_SPECIFIC // Number of outputs from pan, arg is int* #define AF_CONTROL_PAN_NOUT 0x00001B00 | AF_CONTROL_FILTER_SPECIFIC - + // Balance, arg is float*; range -1 (left) to 1 (right), 0 center #define AF_CONTROL_PAN_BALANCE 0x00002500 | AF_CONTROL_FILTER_SPECIFIC -// Set equalizer gain, arg is a control_ext with a float* +// Set equalizer gain, arg is a control_ext with a float* #define AF_CONTROL_EQUALIZER_GAIN 0x00001C00 | AF_CONTROL_FILTER_SPECIFIC diff --git a/libaf/equalizer.h b/libaf/equalizer.h index 8700ad2585..4935401413 100644 --- a/libaf/equalizer.h +++ b/libaf/equalizer.h @@ -26,8 +26,8 @@ typedef struct equalizer_s { - float gain; // Gain in dB -15 - 15 - int channel; // Channel number 0 - 5 + float gain; // Gain in dB -15 - 15 + int channel; // Channel number 0 - 5 int band; // Frequency band 0 - 9 }equalizer_t; diff --git a/libaf/filter.c b/libaf/filter.c index 78d5e86e2d..9ea4359a32 100644 --- a/libaf/filter.c +++ b/libaf/filter.c @@ -32,13 +32,13 @@ n number of filter taps, where mod(n,4)==0 w filter taps - x input signal must be a circular buffer which is indexed backwards + x input signal must be a circular buffer which is indexed backwards */ inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w, const FLOAT_TYPE* x) { register FLOAT_TYPE y; // Output - y = 0.0; + y = 0.0; do{ n--; y+=w[n]*x[n]; @@ -52,7 +52,7 @@ inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w, d number of filters xi current index in xq w filter taps k by n big - x input signal must be a circular buffers which are indexed backwards + x input signal must be a circular buffers which are indexed backwards y output buffer s output buffer stride */ @@ -82,7 +82,7 @@ int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, { register FLOAT_TYPE* txq = *xq + xi; register int nt = n*2; - + while(d-- >0){ *txq= *(txq+n) = *in; txq+=nt; @@ -99,13 +99,13 @@ int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, n filter length must be odd for HP and BS filters w buffer for the filter taps (must be n long) - fc cutoff frequencies (1 for LP and HP, 2 for BP and BS) + fc cutoff frequencies (1 for LP and HP, 2 for BP and BS) 0 < fc < 1 where 1 <=> Fs/2 flags window and filter type as defined in filter.h - variables are ored together: i.e. LP|HAMMING will give a - low pass filter designed using a hamming window + variables are ored together: i.e. LP|HAMMING will give a + low pass filter designed using a hamming window opt beta constant used only when designing using kaiser windows - + returns 0 if OK, -1 if fail */ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, @@ -142,10 +142,10 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, case(KAISER): af_window_kaiser(n,w,opt); break; default: - return -1; + return -1; } - if(flags & (LP | HP)){ + if(flags & (LP | HP)){ fc1=*fc; // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25; @@ -154,7 +154,7 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, if(flags & LP){ // Low pass filter // If the filter length is odd, there is one point which is exactly - // in the middle. The value at this point is 2*fCutoff*sin(x)/x, + // in the middle. The value at this point is 2*fCutoff*sin(x)/x, // where x is zero. To make sure nothing strange happens, we set this // value separately. if (o){ @@ -206,9 +206,9 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2 t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1 g += w[end-i-1] * (t3 + t2); // Total gain in filter - w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); + w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); } - } + } else{ // Band stop if (!o) // Band stop filters must have odd length return -1; @@ -220,7 +220,7 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, t1 = (FLOAT_TYPE)(i+1); t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1 t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2 - w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); + w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); g += 2*w[end-i-1]; // Total gain in filter } } @@ -228,9 +228,9 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, // Normalize gain g=1/g; - for (i=0; i<n; i++) + for (i=0; i<n; i++) w[i] *= g; - + return 0; } @@ -239,7 +239,7 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, n length of prototype filter k number of polyphase components w prototype filter taps - pw Parallel FIR filter + pw Parallel FIR filter g Filter gain flags FWD forward indexing REW reverse indexing @@ -254,7 +254,7 @@ int af_filter_design_pfir(unsigned int n, unsigned int k, const FLOAT_TYPE* w, int i; // Counters int j; FLOAT_TYPE t; // g * w[i] - + // Sanity check if(l<1 || k<1 || !w || !pw) return -1; @@ -287,7 +287,7 @@ int af_filter_design_pfir(unsigned int n, unsigned int k, const FLOAT_TYPE* w, /* Pre-warp the coefficients of a numerator or denominator. Note that a0 is assumed to be 1, so there is no wrapping - of it. + of it. */ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs) { @@ -299,7 +299,7 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs) /* Transform the numerator and denominator coefficients of s-domain biquad section into corresponding z-domain coefficients. - + The transfer function for z-domain is: 1 + alpha1 * z^(-1) + alpha2 * z^(-2) @@ -310,7 +310,7 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs) order: beta1, beta2 (denominator) alpha1, alpha2 (numerator) - + Arguments: a - s-domain numerator coefficients b - s-domain denominator coefficients @@ -318,10 +318,10 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs) biquad section in such a way, as to make it the coefficient by which to multiply the overall filter gain in order to achieve a desired overall filter gain, - specified in initial value of k. + specified in initial value of k. fs - sampling rate (Hz) coef - array of z-domain coefficients to be filled in. - + Return: On return, set coef z-domain coefficients and k to the gain required to maintain overall gain = 1.0; */ @@ -352,26 +352,26 @@ static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_T /* IIR filter design using bilinear transform and prewarp. Transforms 2nd order s domain analog filter into a digital IIR biquad link. To create a filter fill in a, b, Q and fs and make space for coef and k. - - Example Butterworth design: + + Example Butterworth design: Below are Butterworth polynomials, arranged as a series of 2nd order sections: Note: n is filter order. - + n Polynomials ------------------------------------------------------------------- 2 s^2 + 1.4142s + 1 4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1) 6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1) - + For n=4 we have following equation for the filter transfer function: 1 1 T(s) = --------------------------- * ---------------------------- s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1 - + The filter consists of two 2nd order sections since highest s power is 2. Now we can take the coefficients, or the numbers by which s is multiplied and plug them into a standard formula to be used by @@ -414,7 +414,7 @@ static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_T biquad section in such a way, as to make it the coefficient by which to multiply the overall filter gain in order to achieve a desired overall filter gain, - specified in initial value of k. + specified in initial value of k. fs - sampling rate (Hz) coef - array of z-domain coefficients to be filled in. @@ -432,7 +432,7 @@ int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, FL FLOAT_TYPE at[3]; FLOAT_TYPE bt[3]; - if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0)) + if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0)) return -1; memcpy(at,a,3*sizeof(FLOAT_TYPE)); diff --git a/libaf/filter.h b/libaf/filter.h index 8b9c7e8822..a7c8a71c6a 100644 --- a/libaf/filter.h +++ b/libaf/filter.h @@ -26,7 +26,7 @@ #define MPLAYER_FILTER_H -// Design and implementation of different types of digital filters +// Design and implementation of different types of digital filters // Flags used for filter design @@ -79,7 +79,7 @@ int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, /* Add new data to circular queue designed to be used with a FIR filter. xq is the circular queue, in pointing at the new sample, xi current index for xq and n the length of the filter. xq must be n*2 - long. + long. */ #define af_filter_updateq(n,xi,xq,in)\ xq[xi]=(xq)[(xi)+(n)]=*(in);\ diff --git a/libaf/format.c b/libaf/format.c index e681cc2050..1427f232f1 100644 --- a/libaf/format.c +++ b/libaf/format.c @@ -37,7 +37,7 @@ int af_str2fmt(const char* str) else if(strstr(str,"le") || strstr(str,"LE")) format |= AF_FORMAT_LE; else - format |= AF_FORMAT_NE; + format |= AF_FORMAT_NE; // Scan for special formats if(strstr(str,"mulaw") || strstr(str,"MULAW")){ @@ -55,7 +55,7 @@ int af_str2fmt(const char* str) if(strstr(str,"imaadpcm") || strstr(str,"IMAADPCM")){ format |= AF_FORMAT_IMA_ADPCM; return format; } - + // Scan for int/float if(strstr(str,"float") || strstr(str,"FLOAT")){ format |= AF_FORMAT_F; return format; @@ -65,10 +65,10 @@ int af_str2fmt(const char* str) // Scan for signed/unsigned if(strstr(str,"unsigned") || strstr(str,"UNSIGNED")) - format |= AF_FORMAT_US; + format |= AF_FORMAT_US; else format |= AF_FORMAT_SI; - + return format; } @@ -94,7 +94,7 @@ int af_bits2fmt(int bits) return (bits/8 - 1) << 3; } -/* Convert format to str input str is a buffer for the +/* Convert format to str input str is a buffer for the converted string, size is the size of the buffer */ char* af_fmt2str(int format, char* str, int size) { @@ -109,21 +109,21 @@ char* af_fmt2str(int format, char* str, int size) i+=snprintf(str,size-i,"little-endian "); else i+=snprintf(str,size-i,"big-endian "); - + if(format & AF_FORMAT_SPECIAL_MASK){ switch(format & AF_FORMAT_SPECIAL_MASK){ - case(AF_FORMAT_MU_LAW): + case(AF_FORMAT_MU_LAW): i+=snprintf(&str[i],size-i,"mu-law "); break; - case(AF_FORMAT_A_LAW): + case(AF_FORMAT_A_LAW): i+=snprintf(&str[i],size-i,"A-law "); break; - case(AF_FORMAT_MPEG2): + case(AF_FORMAT_MPEG2): i+=snprintf(&str[i],size-i,"MPEG-2 "); break; - case(AF_FORMAT_AC3): + case(AF_FORMAT_AC3): i+=snprintf(&str[i],size-i,"AC3 "); break; - case(AF_FORMAT_IMA_ADPCM): + case(AF_FORMAT_IMA_ADPCM): i+=snprintf(&str[i],size-i,"IMA-ADPCM "); break; default: - i+=snprintf(&str[i],size-i,MSGTR_AF_FORMAT_UnknownFormat); + i+=snprintf(&str[i],size-i,MSGTR_AF_FORMAT_UnknownFormat); } } else{ @@ -183,7 +183,7 @@ static struct { { "floatle", AF_FORMAT_FLOAT_LE }, { "floatbe", AF_FORMAT_FLOAT_BE }, { "floatne", AF_FORMAT_FLOAT_NE }, - + { NULL, 0 } }; diff --git a/libaf/window.c b/libaf/window.c index 0230c33673..2b2923afc9 100644 --- a/libaf/window.c +++ b/libaf/window.c @@ -46,7 +46,7 @@ void af_window_boxcar(int n, FLOAT_TYPE* w) /* // Triang a.k.a Bartlett // -// | (N-1)| +// | (N-1)| // 2 * |k - -----| // | 2 | // w = 1.0 - --------------- @@ -60,7 +60,7 @@ void af_window_triang(int n, FLOAT_TYPE* w) FLOAT_TYPE k2 = 1/((FLOAT_TYPE)n + k1); int end = (n + 1) >> 1; int i; - + // Calculate window coefficients for (i=0 ; i<end ; i++) w[i] = w[n-i-1] = (2.0*((FLOAT_TYPE)(i+1))-(1.0-k1))*k2; @@ -79,7 +79,7 @@ void af_window_hanning(int n, FLOAT_TYPE* w) { int i; FLOAT_TYPE k = 2*M_PI/((FLOAT_TYPE)(n+1)); // 2*pi/(N+1) - + // Calculate window coefficients for (i=0; i<n; i++) *w++ = 0.5*(1.0 - cos(k*(FLOAT_TYPE)(i+1))); @@ -138,23 +138,23 @@ void af_window_flattop(int n,FLOAT_TYPE* w) int i; FLOAT_TYPE k1 = 2*M_PI/((FLOAT_TYPE)(n-1)); // 2*pi/(N-1) FLOAT_TYPE k2 = 2*k1; // 4*pi/(N-1) - + // Calculate window coefficients for (i=0; i<n; i++) *w++ = 0.2810638602 - 0.5208971735*cos(k1*(FLOAT_TYPE)i) + 0.1980389663*cos(k2*(FLOAT_TYPE)i); } -/* Computes the 0th order modified Bessel function of the first kind. -// (Needed to compute Kaiser window) -// +/* Computes the 0th order modified Bessel function of the first kind. +// (Needed to compute Kaiser window) +// // y = sum( (x/(2*n))^2 ) // n */ -#define BIZ_EPSILON 1E-21 // Max error acceptable +#define BIZ_EPSILON 1E-21 // Max error acceptable static FLOAT_TYPE besselizero(FLOAT_TYPE x) -{ +{ FLOAT_TYPE temp; FLOAT_TYPE sum = 1.0; FLOAT_TYPE u = 1.0; @@ -183,10 +183,10 @@ static FLOAT_TYPE besselizero(FLOAT_TYPE x) // Gold (Theory and Application of DSP) under Kaiser windows for more // about Beta. The following table from Rabiner and Gold gives some // feel for the effect of Beta: -// +// // All ripples in dB, width of transition band = D*N where N = window // length -// +// // BETA D PB RIP SB RIP // 2.120 1.50 +-0.27 -30 // 3.384 2.23 0.0864 -40 @@ -203,8 +203,8 @@ void af_window_kaiser(int n, FLOAT_TYPE* w, FLOAT_TYPE b) FLOAT_TYPE k1 = 1.0/besselizero(b); int k2 = 1 - (n & 1); int end = (n + 1) >> 1; - int i; - + int i; + // Calculate window coefficients for (i=0 ; i<end ; i++){ tmp = (FLOAT_TYPE)(2*i + k2) / ((FLOAT_TYPE)n - 1.0); |