diff options
Diffstat (limited to 'src/post/audio')
| -rw-r--r-- | src/post/audio/audio_filters.c | 10 | ||||
| -rw-r--r-- | src/post/audio/audio_filters.h | 8 | ||||
| -rw-r--r-- | src/post/audio/dsp.h | 4 | ||||
| -rw-r--r-- | src/post/audio/filter.c | 66 | ||||
| -rw-r--r-- | src/post/audio/filter.h | 6 | ||||
| -rw-r--r-- | src/post/audio/stretch.c | 194 | ||||
| -rw-r--r-- | src/post/audio/upmix.c | 44 | ||||
| -rw-r--r-- | src/post/audio/upmix_mono.c | 56 | ||||
| -rw-r--r-- | src/post/audio/volnorm.c | 44 | ||||
| -rw-r--r-- | src/post/audio/window.c | 26 | ||||
| -rw-r--r-- | src/post/audio/window.h | 2 |
11 files changed, 230 insertions, 230 deletions
diff --git a/src/post/audio/audio_filters.c b/src/post/audio/audio_filters.c index 448353b52..78f495ca6 100644 --- a/src/post/audio/audio_filters.c +++ b/src/post/audio/audio_filters.c @@ -1,18 +1,18 @@ /* * Copyright (C) 2000-2004 the xine project - * + * * This file is part of xine, a free video player. - * + * * xine is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * xine is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA @@ -38,7 +38,7 @@ static const post_info_t volnorm_special_info = { XINE_POST_TYPE_AUDIO_FILTER const plugin_info_t xine_plugin_info[] EXPORTED = { - /* type, API, "name", version, special_info, init_function */ + /* type, API, "name", version, special_info, init_function */ { PLUGIN_POST, 10, "upmix", XINE_VERSION_CODE, &upmix_special_info, &upmix_init_plugin }, { PLUGIN_POST, 10, "upmix_mono", XINE_VERSION_CODE, &upmix_mono_special_info, &upmix_mono_init_plugin }, { PLUGIN_POST, 10, "stretch", XINE_VERSION_CODE, &stretch_special_info, &stretch_init_plugin }, diff --git a/src/post/audio/audio_filters.h b/src/post/audio/audio_filters.h index 8a0202751..144e85836 100644 --- a/src/post/audio/audio_filters.h +++ b/src/post/audio/audio_filters.h @@ -1,18 +1,18 @@ /* * Copyright (C) 2000-2004 the xine project - * + * * This file is part of xine, a free video player. - * + * * xine is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * xine is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA diff --git a/src/post/audio/dsp.h b/src/post/audio/dsp.h index 237640283..8dabe4f36 100644 --- a/src/post/audio/dsp.h +++ b/src/post/audio/dsp.h @@ -1,5 +1,5 @@ /*============================================================================= - * + * * This software has been released under the terms of the GNU Public * license. See http://www.gnu.org/copyleft/gpl.html for details. * @@ -9,7 +9,7 @@ */ #ifndef _DSP_H -#define _DSP_H 1 +#define _DSP_H 1 /* Implementation of routines used for DSP */ diff --git a/src/post/audio/filter.c b/src/post/audio/filter.c index 55d75e1e4..f41d880fc 100644 --- a/src/post/audio/filter.c +++ b/src/post/audio/filter.c @@ -1,5 +1,5 @@ /*============================================================================= - * + * * This software has been released under the terms of the GNU Public * license. See http://www.gnu.org/copyleft/gpl.html for details. * @@ -25,12 +25,12 @@ * * 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 _ftype_t fir(register unsigned int n, _ftype_t* w, _ftype_t* x) { register _ftype_t y; /* Output */ - y = 0.0; + y = 0.0; do{ n--; y+=w[n]*x[n]; @@ -44,7 +44,7 @@ inline _ftype_t fir(register unsigned int n, _ftype_t* w, _ftype_t* x) * 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 */ @@ -67,11 +67,11 @@ inline _ftype_t* pfir(unsigned int n, unsigned int d, unsigned int xi, _ftype_t* at the new samples, xi current index in xq and n the length of the filter. xq must be n*2 by k big, s is the index for in. */ -inline int updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s) +inline int updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s) { register _ftype_t* txq = *xq + xi; register int nt = n*2; - + while(d-- >0){ *txq= *(txq+n) = *in; txq+=nt; @@ -88,26 +88,26 @@ inline int updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** 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 design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _ftype_t opt) { - unsigned int o = n & 1; /* Indicator for odd filter length */ + unsigned int o = n & 1; /* Indicator for odd filter length */ unsigned int end = ((n + 1) >> 1) - o; /* Loop end */ unsigned int i; /* Loop index */ _ftype_t k1 = 2 * M_PI; /* 2*pi*fc1 */ _ftype_t k2 = 0.5 * (_ftype_t)(1 - o);/* Constant used if the filter has even length */ _ftype_t k3; /* 2*pi*fc2 Constant used in BP and BS design */ - _ftype_t g = 0.0; /* Gain */ - _ftype_t t1,t2,t3; /* Temporary variables */ + _ftype_t g = 0.0; /* Gain */ + _ftype_t t1,t2,t3; /* Temporary variables */ _ftype_t fc1,fc2; /* Cutoff frequencies */ /* Sanity check */ @@ -130,10 +130,10 @@ int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _f case(KAISER): 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; @@ -143,7 +143,7 @@ int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _f /* * 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. */ @@ -196,9 +196,9 @@ int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _f 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; @@ -210,7 +210,7 @@ int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _f t1 = (_ftype_t)(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 */ } } @@ -218,9 +218,9 @@ int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _f /* Normalize gain */ g=1/g; - for (i=0; i<n; i++) + for (i=0; i<n; i++) w[i] *= g; - + return 0; } @@ -229,7 +229,7 @@ int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _f * 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 @@ -240,10 +240,10 @@ int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _f int design_pfir(unsigned int n, unsigned int k, _ftype_t* w, _ftype_t** pw, _ftype_t g, unsigned int flags) { int l = (int)n/k; /* Length of individual FIR filters */ - int i; /* Counters */ + int i; /* Counters */ int j; _ftype_t t; /* g * w[i] */ - + /* Sanity check */ if(l<1 || k<1 || !w || !pw) return -1; @@ -276,7 +276,7 @@ int design_pfir(unsigned int n, unsigned int k, _ftype_t* w, _ftype_t** pw, _fty /* 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. */ void prewarp(_ftype_t* a, _ftype_t fc, _ftype_t fs) { @@ -303,12 +303,12 @@ void prewarp(_ftype_t* a, _ftype_t fc, _ftype_t fs) * Arguments: * a - s-domain numerator coefficients * b - s-domain denominator coefficients - * k - filter gain factor. Initially set to 1 and modified by each + * k - filter gain factor. Initially set to 1 and modified by each * 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. - * fs - sampling rate (Hz) + * 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 @@ -342,7 +342,7 @@ void bilinear(_ftype_t* a, _ftype_t* b, _ftype_t* k, _ftype_t fs, _ftype_t *coef * 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: @@ -398,12 +398,12 @@ void bilinear(_ftype_t* a, _ftype_t* b, _ftype_t* k, _ftype_t fs, _ftype_t *coef * a - s-domain numerator coefficients, a[1] is always assumed to be 1.0 * b - s-domain denominator coefficients * Q - Q value for the filter - * k - filter gain factor. Initially set to 1 and modified by each + * k - filter gain factor. Initially set to 1 and modified by each * 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. - * fs - sampling rate (Hz) + * specified in initial value of k. + * fs - sampling rate (Hz) * coef - array of z-domain coefficients to be filled in. * * Note: Upon return from each call, the k argument will be set to a @@ -419,7 +419,7 @@ int szxform(const _ftype_t* a, const _ftype_t* b, _ftype_t Q, _ftype_t fc, _ftyp _ftype_t at[3]; _ftype_t 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(_ftype_t)); diff --git a/src/post/audio/filter.h b/src/post/audio/filter.h index 0e08aa2b9..4d756a618 100644 --- a/src/post/audio/filter.h +++ b/src/post/audio/filter.h @@ -1,5 +1,5 @@ /*============================================================================= - * + * * This software has been released under the terms of the GNU Public * license. See http://www.gnu.org/copyleft/gpl.html for details. * @@ -37,7 +37,7 @@ #define WINDOW_MASK 0x0000001F /* Parallel filter design */ -#define FWD 0x00000001 /* Forward indexing of polyphase filter */ +#define FWD 0x00000001 /* Forward indexing of polyphase filter */ #define REW 0x00000002 /* Reverse indexing of polyphase filter */ #define ODD 0x00000010 /* Make filter HP */ @@ -60,7 +60,7 @@ extern int szxform(const _ftype_t* a, const _ftype_t* b, _ftype_t Q, _ftype_t fc /* 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 updateq(n,xi,xq,in)\ xq[xi]=(xq)[(xi)+(n)]=*(in);\ diff --git a/src/post/audio/stretch.c b/src/post/audio/stretch.c index 36ac18dd6..7397844b4 100644 --- a/src/post/audio/stretch.c +++ b/src/post/audio/stretch.c @@ -1,18 +1,18 @@ /* * Copyright (C) 2000-2004 the xine project - * + * * This file is part of xine, a free video player. - * + * * xine is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * xine is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA @@ -70,19 +70,19 @@ static void stretchscr_set_pivot (stretchscr_t *this) { struct timeval tv; int64_t pts; - double pts_calc; + double pts_calc; xine_monotonic_clock(&tv, NULL); pts_calc = (tv.tv_sec - this->cur_time.tv_sec) * this->speed_factor; pts_calc += (tv.tv_usec - this->cur_time.tv_usec) * this->speed_factor / 1e6; pts = this->cur_pts + pts_calc; -/* This next part introduces a one off inaccuracy - * to the scr due to rounding tv to pts. +/* This next part introduces a one off inaccuracy + * to the scr due to rounding tv to pts. */ this->cur_time.tv_sec=tv.tv_sec; this->cur_time.tv_usec=tv.tv_usec; - this->cur_pts=pts; + this->cur_pts=pts; return ; } @@ -94,7 +94,7 @@ static int stretchscr_set_speed (scr_plugin_t *scr, int speed) { stretchscr_set_pivot( this ); this->xine_speed = speed; - this->speed_factor = (double) speed * 90000.0 / XINE_FINE_SPEED_NORMAL / + this->speed_factor = (double) speed * 90000.0 / XINE_FINE_SPEED_NORMAL / (*this->stretch_factor); pthread_mutex_unlock (&this->lock); @@ -125,7 +125,7 @@ static void stretchscr_start (scr_plugin_t *scr, int64_t start_vpts) { this->cur_pts = start_vpts; pthread_mutex_unlock (&this->lock); - + stretchscr_set_speed (&this->scr, XINE_FINE_SPEED_NORMAL); } @@ -134,16 +134,16 @@ static int64_t stretchscr_get_current (scr_plugin_t *scr) { struct timeval tv; int64_t pts; - double pts_calc; + double pts_calc; pthread_mutex_lock (&this->lock); xine_monotonic_clock(&tv, NULL); - + pts_calc = (tv.tv_sec - this->cur_time.tv_sec) * this->speed_factor; pts_calc += (tv.tv_usec - this->cur_time.tv_usec) * this->speed_factor / 1e6; pts = this->cur_pts + pts_calc; - + pthread_mutex_unlock (&this->lock); return pts; @@ -168,9 +168,9 @@ static stretchscr_t *XINE_MALLOC stretchscr_init (double *stretch_factor) { this->scr.start = stretchscr_start; this->scr.get_current = stretchscr_get_current; this->scr.exit = stretchscr_exit; - + pthread_mutex_init (&this->lock, NULL); - + this->stretch_factor = stretch_factor; stretchscr_set_speed (&this->scr, XINE_SPEED_PAUSE); @@ -202,21 +202,21 @@ typedef struct stretch_parameters_s { START_PARAM_DESCR( stretch_parameters_t ) PARAM_ITEM( POST_PARAM_TYPE_BOOL, preserve_pitch, NULL, 0, 1, 0, "Preserve pitch" ) -PARAM_ITEM( POST_PARAM_TYPE_DOUBLE, factor, NULL, 0.5, 1.5, 0, +PARAM_ITEM( POST_PARAM_TYPE_DOUBLE, factor, NULL, 0.5, 1.5, 0, "Time stretch factor (<1.0 shorten duration)" ) END_PARAM_DESCR( param_descr ) /* plugin structure */ struct post_plugin_stretch_s { post_plugin_t post; - + stretchscr_t* scr; /* private data */ stretch_parameters_t params; xine_post_in_t params_input; int params_changed; - + int channels; int bytes_per_frame; @@ -228,9 +228,9 @@ struct post_plugin_stretch_s { int num_frames; /* current # of frames on audiofrag */ int16_t last_sample[RESAMPLE_MAX_CHANNELS]; - + int64_t pts; /* pts for audiofrag */ - + pthread_mutex_t lock; }; @@ -290,16 +290,16 @@ static int stretch_port_open(xine_audio_port_t *port_gen, xine_stream_t *stream, post_audio_port_t *port = (post_audio_port_t *)port_gen; post_plugin_stretch_t *this = (post_plugin_stretch_t *)port->post; int64_t time; - + _x_post_rewire(&this->post); _x_post_inc_usage(port); - + port->stream = stream; port->bits = bits; port->rate = rate; port->mode = mode; - - /* register our own scr provider */ + + /* register our own scr provider */ time = port->stream->xine->clock->get_current_time(port->stream->xine->clock); this->scr = stretchscr_init(&this->params.factor); this->scr->scr.start(&this->scr->scr, time); @@ -307,7 +307,7 @@ static int stretch_port_open(xine_audio_port_t *port_gen, xine_stream_t *stream, /* force updating on stretch_port_put_buffer */ this->params_changed = 1; - + return (port->original_port->open) (port->original_port, stream, bits, rate, mode); } @@ -320,34 +320,34 @@ static void stretch_port_close(xine_audio_port_t *port_gen, xine_stream_t *strea port->stream->xine->clock->unregister_scr(port->stream->xine->clock, &this->scr->scr); this->scr->scr.exit(&this->scr->scr); } - + if(this->audiofrag) { free(this->audiofrag); this->audiofrag = NULL; } - + if(this->outfrag) { free(this->outfrag); this->outfrag = NULL; } - + if(this->w) { free(this->w); this->w = NULL; } - + port->stream = NULL; port->original_port->close(port->original_port, stream ); - + _x_post_dec_usage(port); } -static void stretch_process_fragment( post_audio_port_t *port, +static void stretch_process_fragment( post_audio_port_t *port, xine_stream_t *stream, extra_info_t *extra_info ) { post_plugin_stretch_t *this = (post_plugin_stretch_t *)port->post; - + audio_buffer_t *outbuf; int16_t *data_out = this->outfrag; int num_frames_in = this->num_frames; @@ -374,52 +374,52 @@ static void stretch_process_fragment( post_audio_port_t *port, * input chunk has two halves, A and B. * output chunk is composed as follow: * - some frames copied directly from A - * - some frames copied from A merged with frames from B + * - some frames copied from A merged with frames from B * weighted by an increasing factor (0 -> 1.0) * - frames from A weighted by a decreasing factor (1.0 -> 0) * merged with frames copied from B * - some frames copied directly from B */ - + int merge_frames = num_frames_in - num_frames_out; int copy_frames; int16_t *src = this->audiofrag; int16_t *dst = this->outfrag; int i, j; - + if( merge_frames > num_frames_out ) merge_frames = num_frames_out; copy_frames = num_frames_out - merge_frames; - + memcpy(dst, src, copy_frames/2 * this->bytes_per_frame); dst += copy_frames/2 * this->channels; src += copy_frames/2 * this->channels; - + for( i = 0; i < merge_frames/2; i++ ) { for( j = 0; j < this->channels; j++, src++, dst++ ) { - + int32_t s = (int32_t) ((_ftype_t) src[0] + src[merge_frames * this->channels] * this->w[i]); *dst = CLIP_INT16(s); } } - + for( ; i < merge_frames; i++ ) { for( j = 0; j < this->channels; j++, src++, dst++ ) { - + int32_t s = (int32_t) ((_ftype_t) src[0] * this->w[i] + src[merge_frames * this->channels]); *dst = CLIP_INT16(s); } } - + src += merge_frames * this->channels; - + memcpy(dst, src, (copy_frames - copy_frames/2) * this->bytes_per_frame); - + } else { /* * time expansion strategy @@ -427,13 +427,13 @@ static void stretch_process_fragment( post_audio_port_t *port, * output chunk is composed of two versions of the * input chunk: * - first part copied directly from input, and then - * merged with the second (delayed) part using a + * merged with the second (delayed) part using a * decreasing factor (1.0 -> 0) * - the delayed version of the input is merged with - * an increasing factor (0 -> 1.0) and then (when - * factor reaches 1.0) just copied until the end. + * an increasing factor (0 -> 1.0) and then (when + * factor reaches 1.0) just copied until the end. */ - + int merge_frames = num_frames_out - num_frames_in; int copy_frames = num_frames_out - merge_frames; int16_t *src1 = this->audiofrag; @@ -449,67 +449,67 @@ static void stretch_process_fragment( post_audio_port_t *port, for( i = 0; i < merge_frames/2; i++ ) { for( j = 0; j < this->channels; j++, src1++, src2++, dst++ ) { - + int32_t s = (int32_t) ((_ftype_t) *src1 + *src2 * this->w[i]); *dst = CLIP_INT16(s); } } - + for( ; i < merge_frames; i++ ) { for( j = 0; j < this->channels; j++, src1++, src2++, dst++ ) { - + int32_t s = (int32_t) ((_ftype_t) *src1 * this->w[i] + *src2); *dst = CLIP_INT16(s); } } - + memcpy(dst, src2, (copy_frames - copy_frames/2) * this->bytes_per_frame); } } - + /* copy processed fragment into multiple audio buffers, if needed */ while( num_frames_out ) { - outbuf = port->original_port->get_buffer(port->original_port); - + outbuf = port->original_port->get_buffer(port->original_port); + outbuf->num_frames = outbuf->mem_size / this->bytes_per_frame; if( outbuf->num_frames > num_frames_out ) outbuf->num_frames = num_frames_out; - - memcpy( outbuf->mem, data_out, + + memcpy( outbuf->mem, data_out, outbuf->num_frames * this->bytes_per_frame ); num_frames_out -= outbuf->num_frames; data_out = (uint16_t *)((uint8_t *)data_out + outbuf->num_frames * this->bytes_per_frame); - + outbuf->vpts = this->pts; this->pts = 0; outbuf->stream = stream; outbuf->format.bits = port->bits; outbuf->format.rate = port->rate; outbuf->format.mode = port->mode; - + _x_extra_info_merge( outbuf->extra_info, extra_info ); - - port->original_port->put_buffer(port->original_port, outbuf, stream ); + + port->original_port->put_buffer(port->original_port, outbuf, stream ); } - + this->num_frames = 0; } -static void stretch_port_put_buffer (xine_audio_port_t *port_gen, +static void stretch_port_put_buffer (xine_audio_port_t *port_gen, audio_buffer_t *buf, xine_stream_t *stream) { - + post_audio_port_t *port = (post_audio_port_t *)port_gen; post_plugin_stretch_t *this = (post_plugin_stretch_t *)port->post; int16_t *data_in; - + pthread_mutex_lock (&this->lock); - + if( this->params_changed ) { int64_t audio_step; @@ -517,40 +517,40 @@ static void stretch_port_put_buffer (xine_audio_port_t *port_gen, /* output whatever we have before changing parameters */ stretch_process_fragment( port, stream, buf->extra_info ); } - + this->channels = _x_ao_mode2channels(port->mode); this->bytes_per_frame = port->bits / 8 * this->channels; - + audio_step = ((int64_t)90000 * (int64_t)32768) / (int64_t)port->rate; audio_step = (int64_t) ((double)audio_step / this->params.factor); stream->metronom->set_audio_rate(stream->metronom, audio_step); stretchscr_set_speed(&this->scr->scr, this->scr->xine_speed); - + if(this->audiofrag) { free(this->audiofrag); this->audiofrag = NULL; } - + if(this->outfrag) { free(this->outfrag); this->outfrag = NULL; } - + if(this->w) { free(this->w); this->w = NULL; } - + this->frames_per_frag = port->rate * AUDIO_FRAGMENT; this->frames_per_outfrag = (int) ((double)this->params.factor * this->frames_per_frag); if( this->frames_per_frag != this->frames_per_outfrag ) { int wsize; - - this->audiofrag = malloc( this->frames_per_frag * this->bytes_per_frame ); - this->outfrag = malloc( this->frames_per_outfrag * this->bytes_per_frame ); - + + this->audiofrag = malloc( this->frames_per_frag * this->bytes_per_frame ); + this->outfrag = malloc( this->frames_per_outfrag * this->bytes_per_frame ); + if( this->frames_per_frag > this->frames_per_outfrag ) wsize = this->frames_per_frag - this->frames_per_outfrag; else @@ -559,41 +559,41 @@ static void stretch_port_put_buffer (xine_audio_port_t *port_gen, this->w = (_ftype_t*) malloc( wsize * sizeof(_ftype_t) ); triang(wsize, this->w); } - + this->num_frames = 0; this->pts = 0; - + this->params_changed = 0; } - + pthread_mutex_unlock (&this->lock); /* just pass data through if we have nothing to do */ if( this->frames_per_frag == this->frames_per_outfrag || /* FIXME: we only handle 1 or 2 channels, 16 bits for now */ - (this->channels != 1 && this->channels != 2) || + (this->channels != 1 && this->channels != 2) || port->bits != 16 ) { - - port->original_port->put_buffer(port->original_port, buf, stream ); - + + port->original_port->put_buffer(port->original_port, buf, stream ); + return; } - + /* update pts for our current audio fragment */ if( buf->vpts ) this->pts = buf->vpts - (this->num_frames * 90000 / port->rate); - + data_in = buf->mem; while( buf->num_frames ) { int frames_to_copy = this->frames_per_frag - this->num_frames; - + if( frames_to_copy > buf->num_frames ) frames_to_copy = buf->num_frames; /* copy up to one fragment from input buf to our buffer */ memcpy( (uint8_t *)this->audiofrag + this->num_frames * this->bytes_per_frame, data_in, frames_to_copy * this->bytes_per_frame ); - + data_in = (uint16_t *)((uint8_t *)data_in + frames_to_copy * this->bytes_per_frame); this->num_frames += frames_to_copy; buf->num_frames -= frames_to_copy; @@ -603,10 +603,10 @@ static void stretch_port_put_buffer (xine_audio_port_t *port_gen, stretch_process_fragment( port, stream, buf->extra_info ); } } - + buf->num_frames=0; /* UNDOCUMENTED, but hey, it works! Force old audio_out buffer free. */ - port->original_port->put_buffer(port->original_port, buf, stream ); - + port->original_port->put_buffer(port->original_port, buf, stream ); + return; } @@ -630,21 +630,21 @@ static post_plugin_t *stretch_open_plugin(post_class_t *class_gen, int inputs, xine_post_in_t *input_api; post_audio_port_t *port; stretch_parameters_t init_params; - + if (!this || !audio_target || !audio_target[0] ) { free(this); return NULL; } - + _x_post_init(&this->post, 1, 0); init_params.preserve_pitch = 1; init_params.factor = 0.80; - + pthread_mutex_init (&this->lock, NULL); set_parameters ((xine_post_t *)&this->post, &init_params); - + port = _x_post_intercept_audio_port(&this->post, audio_target[0], &input, &output); port->new_port.open = stretch_port_open; port->new_port.close = stretch_port_close; @@ -667,16 +667,16 @@ static post_plugin_t *stretch_open_plugin(post_class_t *class_gen, int inputs, void *stretch_init_plugin(xine_t *xine, void *data) { post_class_stretch_t *class = (post_class_stretch_t *)xine_xmalloc(sizeof(post_class_stretch_t)); - + if (!class) return NULL; - + class->post_class.open_plugin = stretch_open_plugin; class->post_class.identifier = "stretch"; class->post_class.description = N_("Time stretch by a given factor, optionally preserving pitch"); class->post_class.dispose = default_post_class_dispose; - + class->xine = xine; - + return class; } diff --git a/src/post/audio/upmix.c b/src/post/audio/upmix.c index 24df657d0..c8f7c0121 100644 --- a/src/post/audio/upmix.c +++ b/src/post/audio/upmix.c @@ -1,18 +1,18 @@ /* * Copyright (C) 2000-2004 the xine project - * + * * This file is part of xine, a free video player. - * + * * xine is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * xine is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA @@ -167,13 +167,13 @@ static int upmix_port_open(xine_audio_port_t *port_gen, xine_stream_t *stream, _x_post_rewire(&this->post); _x_post_inc_usage(port); - + port->stream = stream; port->bits = bits; port->rate = rate; port->mode = mode; capabilities = port->original_port->get_capabilities(port->original_port); - + this->channels = _x_ao_mode2channels(mode); /* FIXME: Handle all desired output formats */ if ((capabilities & AO_CAP_MODE_5_1CHANNEL) && (capabilities & AO_CAP_FLOAT32)) { @@ -232,7 +232,7 @@ static int upmix_frames_2to51_any_to_float( uint8_t *dst8, uint8_t *src8, int nu float right; float sum; int frame; - int src_units_per_sample=1; + int src_units_per_sample=1; if (step_channel_in == 3) src_units_per_sample=step_channel_in; /* Special handling for 24 bit 3byte input */ for (frame=0;frame < num_frames; frame++) { @@ -297,9 +297,9 @@ static int upmix_frames_2to51_any_to_float( uint8_t *dst8, uint8_t *src8, int nu return frame; } -static void upmix_port_put_buffer (xine_audio_port_t *port_gen, +static void upmix_port_put_buffer (xine_audio_port_t *port_gen, audio_buffer_t *buf, xine_stream_t *stream) { - + post_audio_port_t *port = (post_audio_port_t *)port_gen; post_plugin_upmix_t *this = (post_plugin_upmix_t *)port->post; int src_step_frame; @@ -315,7 +315,7 @@ static void upmix_port_put_buffer (xine_audio_port_t *port_gen, if ((this->channels==2) && (this->channels_out==6)) { while (num_frames_processed < buf->num_frames) { - this->buf = port->original_port->get_buffer(port->original_port); + this->buf = port->original_port->get_buffer(port->original_port); /* this->buf->num_frames is handled after the upmix */ this->buf->vpts = buf->vpts; if (num_frames_processed != 0) this->buf->vpts = 0; @@ -327,7 +327,7 @@ static void upmix_port_put_buffer (xine_audio_port_t *port_gen, this->buf->format.bits = 32; /* Upmix to floats */ this->buf->format.rate = port->rate; this->buf->format.mode = AO_CAP_MODE_5_1CHANNEL; - _x_extra_info_merge( this->buf->extra_info, buf->extra_info); + _x_extra_info_merge( this->buf->extra_info, buf->extra_info); step_channel_in = port->bits>>3; step_channel_out = this->buf->format.bits>>3; dst_step_frame = this->channels_out*step_channel_out; @@ -353,18 +353,18 @@ static void upmix_port_put_buffer (xine_audio_port_t *port_gen, } } pthread_mutex_unlock (&this->lock); - + num_frames_done = upmix_frames_2to51_any_to_float(data8dst, data8src, num_frames, step_channel_in, this->sub); this->buf->num_frames = num_frames_done; num_frames_processed+= num_frames_done; /* pass data to original port */ - port->original_port->put_buffer(port->original_port, this->buf, stream ); + port->original_port->put_buffer(port->original_port, this->buf, stream ); } /* free data from origial buffer */ buf->num_frames=0; /* UNDOCUMENTED, but hey, it works! Force old audio_out buffer free. */ } - port->original_port->put_buffer(port->original_port, buf, stream ); - + port->original_port->put_buffer(port->original_port, buf, stream ); + return; } @@ -388,16 +388,16 @@ static post_plugin_t *upmix_open_plugin(post_class_t *class_gen, int inputs, post_out_t *output; xine_post_in_t *input_api; post_audio_port_t *port; - + if (!this || !audio_target || !audio_target[0] ) { free(this); return NULL; } - + _x_post_init(&this->post, 1, 0); this->params.cut_off_freq = 100; - + port = _x_post_intercept_audio_port(&this->post, audio_target[0], &input, &output); port->new_port.open = upmix_port_open; #if 0 @@ -422,16 +422,16 @@ static post_plugin_t *upmix_open_plugin(post_class_t *class_gen, int inputs, void *upmix_init_plugin(xine_t *xine, void *data) { post_class_upmix_t *class = (post_class_upmix_t *)xine_xmalloc(sizeof(post_class_upmix_t)); - + if (!class) return NULL; - + class->post_class.open_plugin = upmix_open_plugin; class->post_class.identifier = "upmix"; class->post_class.description = N_("upmix"); class->post_class.dispose = default_post_class_dispose; - + class->xine = xine; - + return class; } diff --git a/src/post/audio/upmix_mono.c b/src/post/audio/upmix_mono.c index c59638be4..499856e69 100644 --- a/src/post/audio/upmix_mono.c +++ b/src/post/audio/upmix_mono.c @@ -1,18 +1,18 @@ /* * Copyright (C) 2000-2004 the xine project - * + * * This file is part of xine, a free video player. - * + * * xine is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * xine is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA @@ -68,11 +68,11 @@ struct post_plugin_upmix_mono_s { /* private data */ int channels; - + upmix_mono_parameters_t params; xine_post_in_t params_input; int params_changed; - + pthread_mutex_t lock; }; @@ -134,7 +134,7 @@ static int upmix_mono_port_open(xine_audio_port_t *port_gen, xine_stream_t *stre _x_post_rewire(&this->post); _x_post_inc_usage(port); - + port->stream = stream; port->bits = bits; port->rate = rate; @@ -142,7 +142,7 @@ static int upmix_mono_port_open(xine_audio_port_t *port_gen, xine_stream_t *stre this->channels = _x_ao_mode2channels(mode); capabilities = port->original_port->get_capabilities(port->original_port); - + if (this->channels == 1 && (capabilities & AO_CAP_MODE_STEREO)) { xprintf(stream->xine, XINE_VERBOSITY_LOG, _(LOG_MODULE ": upmixing Mono to Stereo.\n")); @@ -151,7 +151,7 @@ static int upmix_mono_port_open(xine_audio_port_t *port_gen, xine_stream_t *stre if ( this->channels != 1) xprintf(stream->xine, XINE_VERBOSITY_LOG, ngettext(LOG_MODULE ": upmixing a single channel from original %d channel stream.\n", - LOG_MODULE ": upmixing a single channel from original %d channels stream.\n", + LOG_MODULE ": upmixing a single channel from original %d channels stream.\n", this->channels), this->channels); else { xprintf(stream->xine, XINE_VERBOSITY_LOG, @@ -163,15 +163,15 @@ static int upmix_mono_port_open(xine_audio_port_t *port_gen, xine_stream_t *stre return (port->original_port->open) (port->original_port, stream, bits, rate, mode); } -static void upmix_mono_port_put_buffer(xine_audio_port_t *port_gen, +static void upmix_mono_port_put_buffer(xine_audio_port_t *port_gen, audio_buffer_t *buf, xine_stream_t *stream) { - + post_audio_port_t *port = (post_audio_port_t *)port_gen; post_plugin_upmix_mono_t *this = (post_plugin_upmix_mono_t *)port->post; - + pthread_mutex_lock (&this->lock); - if (this->channels == 1) + if (this->channels == 1) { audio_buffer_t *buf0 = port->original_port->get_buffer(port->original_port); audio_buffer_t *buf1 = port->original_port->get_buffer(port->original_port); @@ -232,7 +232,7 @@ static void upmix_mono_port_put_buffer(xine_audio_port_t *port_gen, /* free data from origial buffer */ buf->num_frames = 0; /* UNDOCUMENTED, but hey, it works! Force old audio_out buffer free. */ } - else if (this->channels && this->params.channel >= 0) + else if (this->channels && this->params.channel >= 0) { audio_buffer_t *buf0 = port->original_port->get_buffer(port->original_port); buf0->num_frames = buf->num_frames; @@ -253,12 +253,12 @@ static void upmix_mono_port_put_buffer(xine_audio_port_t *port_gen, uint8_t *dst0 = (uint8_t *)buf0->mem; int cur_channel = this->params.channel; int i, j; - + if( cur_channel >= this->channels ) cur_channel = this->channels-1; src += cur_channel * step; - + for (i = 0; i < buf->num_frames; i++) { for (j = 0; j < this->channels; j++ ) @@ -276,11 +276,11 @@ static void upmix_mono_port_put_buffer(xine_audio_port_t *port_gen, /* free data from origial buffer */ buf->num_frames = 0; /* UNDOCUMENTED, but hey, it works! Force old audio_out buffer free. */ } - + pthread_mutex_unlock (&this->lock); - + port->original_port->put_buffer(port->original_port, buf, stream); - + return; } @@ -303,24 +303,24 @@ static post_plugin_t *upmix_mono_open_plugin(post_class_t *class_gen, int inputs xine_post_in_t *input_api; post_audio_port_t *port; upmix_mono_parameters_t init_params; - + if (!this || !audio_target || !audio_target[0]) { free(this); return NULL; } - + _x_post_init(&this->post, 1, 0); init_params.channel = -1; - + pthread_mutex_init (&this->lock, NULL); set_parameters ((xine_post_t *)&this->post, &init_params); - + port = _x_post_intercept_audio_port(&this->post, audio_target[0], &input, &output); port->new_port.open = upmix_mono_port_open; port->new_port.put_buffer = upmix_mono_port_put_buffer; - + input_api = &this->params_input; input_api->name = "parameters"; input_api->type = XINE_POST_DATA_PARAMETERS; @@ -337,17 +337,17 @@ static post_plugin_t *upmix_mono_open_plugin(post_class_t *class_gen, int inputs void *upmix_mono_init_plugin(xine_t *xine, void *data) { post_class_upmix_mono_t *class = (post_class_upmix_mono_t *)xine_xmalloc(sizeof(post_class_upmix_mono_t)); - + if (!class) return NULL; - + class->post_class.open_plugin = upmix_mono_open_plugin; class->post_class.identifier = "upmix_mono"; class->post_class.description = N_("converts Mono into Stereo"); class->post_class.dispose = default_post_class_dispose; - + class->xine = xine; - + return class; } diff --git a/src/post/audio/volnorm.c b/src/post/audio/volnorm.c index e473d6a09..dda38fa5f 100644 --- a/src/post/audio/volnorm.c +++ b/src/post/audio/volnorm.c @@ -1,24 +1,24 @@ /* * Copyright (C) 2000-2008 the xine project - * + * * This file is part of xine, a free video player. - * + * * xine is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * xine is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA * * Volume normalization audio filter for xine. Ported by Jason Tackaberry - * from MPlayer's af_volnorm, which is copyright 2004 by Alex Beregszaszi + * from MPlayer's af_volnorm, which is copyright 2004 by Alex Beregszaszi * & Pierre Lombard. */ @@ -117,7 +117,7 @@ struct post_plugin_volnorm_s { /************************************************************************** * volnorm parameters functions *************************************************************************/ -static int set_parameters (xine_post_t *this_gen, void *param_gen) +static int set_parameters (xine_post_t *this_gen, void *param_gen) { post_plugin_volnorm_t *this = (post_plugin_volnorm_t *)this_gen; volnorm_parameters_t *param = (volnorm_parameters_t *)param_gen; @@ -129,7 +129,7 @@ static int set_parameters (xine_post_t *this_gen, void *param_gen) return 1; } -static int get_parameters (xine_post_t *this_gen, void *param_gen) +static int get_parameters (xine_post_t *this_gen, void *param_gen) { post_plugin_volnorm_t *this = (post_plugin_volnorm_t *)this_gen; volnorm_parameters_t *param = (volnorm_parameters_t *)param_gen; @@ -141,12 +141,12 @@ static int get_parameters (xine_post_t *this_gen, void *param_gen) return 1; } -static xine_post_api_descr_t * get_param_descr (void) +static xine_post_api_descr_t * get_param_descr (void) { return ¶m_descr; } -static char * get_help (void) +static char * get_help (void) { return _("Normalizes audio by maximizing the volume without distorting " "the sound.\n" @@ -172,14 +172,14 @@ static xine_post_api_t post_api = { *************************************************************************/ static int volnorm_port_open(xine_audio_port_t *port_gen, xine_stream_t *stream, - uint32_t bits, uint32_t rate, int mode) + uint32_t bits, uint32_t rate, int mode) { post_audio_port_t *port = (post_audio_port_t *)port_gen; post_plugin_volnorm_t *this = (post_plugin_volnorm_t *)port->post; _x_post_rewire(&this->post); _x_post_inc_usage(port); - + port->stream = stream; port->bits = bits; port->rate = rate; @@ -373,10 +373,10 @@ static void method2_float(post_plugin_volnorm_t *this, audio_buffer_t *buf) } -static void volnorm_port_put_buffer (xine_audio_port_t *port_gen, - audio_buffer_t *buf, xine_stream_t *stream) +static void volnorm_port_put_buffer (xine_audio_port_t *port_gen, + audio_buffer_t *buf, xine_stream_t *stream) { - + post_audio_port_t *port = (post_audio_port_t *)port_gen; post_plugin_volnorm_t *this = (post_plugin_volnorm_t *)port->post; @@ -391,8 +391,8 @@ static void volnorm_port_put_buffer (xine_audio_port_t *port_gen, else if (buf->format.bits == 32) method2_float(this, buf); } - port->original_port->put_buffer(port->original_port, buf, stream ); - + port->original_port->put_buffer(port->original_port, buf, stream ); + return; } @@ -416,12 +416,12 @@ static post_plugin_t *volnorm_open_plugin(post_class_t *class_gen, int inputs, post_out_t *output; xine_post_in_t *input_api; post_audio_port_t *port; - + if (!this || !audio_target || !audio_target[0] ) { free(this); return NULL; } - + _x_post_init(&this->post, 1, 0); pthread_mutex_init (&this->lock, NULL); @@ -453,16 +453,16 @@ static post_plugin_t *volnorm_open_plugin(post_class_t *class_gen, int inputs, void *volnorm_init_plugin(xine_t *xine, void *data) { post_class_volnorm_t *class = (post_class_volnorm_t *)xine_xmalloc(sizeof(post_class_volnorm_t)); - + if (!class) return NULL; - + class->post_class.open_plugin = volnorm_open_plugin; class->post_class.identifier = "volnorm"; class->post_class.description = N_("Normalize volume"); class->post_class.dispose = default_post_class_dispose; - + class->xine = xine; - + return class; } diff --git a/src/post/audio/window.c b/src/post/audio/window.c index fdea90590..ca58152d5 100644 --- a/src/post/audio/window.c +++ b/src/post/audio/window.c @@ -1,5 +1,5 @@ /*============================================================================= - * + * * This software has been released under the terms of the GNU Public * license. See http://www.gnu.org/copyleft/gpl.html for details. * @@ -39,7 +39,7 @@ void boxcar(int n, _ftype_t* w) /* * Triang a.k.a Bartlett * - * | (N-1)| + * | (N-1)| * 2 * |k - -----| * | 2 | * w = 1.0 - --------------- @@ -53,7 +53,7 @@ void triang(int n, _ftype_t* w) _ftype_t k2 = 1/((_ftype_t)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*((_ftype_t)(i+1))-(1.0-k1))*k2; @@ -72,7 +72,7 @@ void hanning(int n, _ftype_t* w) { int i; _ftype_t k = 2*M_PI/((_ftype_t)(n+1)); /* 2*pi/(N+1) */ - + /* Calculate window coefficients */ for (i=0; i<n; i++) *w++ = 0.5*(1.0 - cos(k*(_ftype_t)(i+1))); @@ -131,22 +131,22 @@ void flattop(int n,_ftype_t* w) int i; _ftype_t k1 = 2*M_PI/((_ftype_t)(n-1)); /* 2*pi/(N-1) */ _ftype_t k2 = 2*k1; /* 4*pi/(N-1) */ - + /* Calculate window coefficients */ for (i=0; i<n; i++) *w++ = 0.2810638602 - 0.5208971735*cos(k1*(_ftype_t)i) + 0.1980389663*cos(k2*(_ftype_t)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 */ static _ftype_t besselizero(_ftype_t x) -{ +{ _ftype_t temp; _ftype_t sum = 1.0; _ftype_t u = 1.0; @@ -175,10 +175,10 @@ static _ftype_t besselizero(_ftype_t 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 @@ -195,8 +195,8 @@ void kaiser(int n, _ftype_t* w, _ftype_t b) _ftype_t 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 = (_ftype_t)(2*i + k2) / ((_ftype_t)n - 1.0); diff --git a/src/post/audio/window.h b/src/post/audio/window.h index d0a7446eb..638707809 100644 --- a/src/post/audio/window.h +++ b/src/post/audio/window.h @@ -1,5 +1,5 @@ /*============================================================================= - * + * * This software has been released under the terms of the GNU Public * license. See http://www.gnu.org/copyleft/gpl.html for details. * |