Mercurial > mplayer.hg
view libaf/af_lavcresample.c @ 25372:9850c8b67ccc
Always enable largefile support by default
The largefile configure option was disabled by default, but the
enabled-by-default dvdread and dvdcss options force in on (dvdnav
code also tries to enable it, but sets the variable too late so that
is has no effect). Change configure to enable largefile support
independently of other options unless explicitly disabled.
author | uau |
---|---|
date | Fri, 14 Dec 2007 22:02:48 +0000 |
parents | 9079c9745ff9 |
children | 867ee1c2114b |
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// Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> // #inlcude <GPL_v2.h> #include <stdio.h> #include <stdlib.h> #include <string.h> #include <inttypes.h> #include "config.h" #include "af.h" #ifdef USE_LIBAVCODEC_SO #include <ffmpeg/avcodec.h> #include <ffmpeg/rational.h> #else #include "avcodec.h" #include "rational.h" #endif // Data for specific instances of this filter typedef struct af_resample_s{ struct AVResampleContext *avrctx; int16_t *in[AF_NCH]; int in_alloc; int index; int filter_length; int linear; int phase_shift; double cutoff; }af_resample_t; // 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_data_t *data= (af_data_t*)arg; int out_rate, test_output_res; // helpers for checking input format switch(cmd){ case AF_CONTROL_REINIT: if((af->data->rate == data->rate) || (af->data->rate == 0)) return AF_DETACH; af->data->nch = data->nch; if (af->data->nch > AF_NCH) af->data->nch = AF_NCH; af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; af->mul = (double)af->data->rate / data->rate; af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5 if(s->avrctx) av_resample_close(s->avrctx); s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff); // hack to make af_test_output ignore the samplerate change out_rate = af->data->rate; af->data->rate = data->rate; test_output_res = af_test_output(af, (af_data_t*)arg); af->data->rate = out_rate; return test_output_res; case AF_CONTROL_COMMAND_LINE:{ s->cutoff= 0.0; sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff); if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80); return AF_OK; } case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET: af->data->rate = *(int*)arg; return AF_OK; } return AF_UNKNOWN; } // Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) free(af->data->audio); free(af->data); if(af->setup){ int i; af_resample_t *s = af->setup; if(s->avrctx) av_resample_close(s->avrctx); for (i=0; i < AF_NCH; i++) free(s->in[i]); free(s); } } // 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; int16_t *out; int chans = data->nch; 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++){ s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t)); } } if(chans==1){ memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t)); }else if(chans==2){ for(j=0; j<in_len; j++){ s->in[0][j + s->index]= *(in++); s->in[1][j + s->index]= *(in++); } }else{ for(j=0; j<in_len; j++){ for(i=0; i<chans; i++){ s->in[i][j + s->index]= *(in++); } } } in_len += s->index; for(i=0; i<chans; i++){ 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)); } if(chans==1){ memcpy(out, tmp[0], out_len*sizeof(int16_t)); }else if(chans==2){ for(j=0; j<out_len; j++){ *(out++)= tmp[0][j]; *(out++)= tmp[1][j]; } }else{ for(j=0; j<out_len; j++){ for(i=0; i<chans; i++){ *(out++)= tmp[i][j]; } } } data->audio = af->data->audio; data->len = out_len*chans*2; data->rate = af->data->rate; return data; } static int af_open(af_instance_t* af){ af_resample_t *s = calloc(1,sizeof(af_resample_t)); af->control=control; af->uninit=uninit; af->play=play; af->mul=1; af->data=calloc(1,sizeof(af_data_t)); s->filter_length= 16; s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80); s->phase_shift= 10; // s->setup = RSMP_INT | FREQ_SLOPPY; af->setup=s; return AF_OK; } af_info_t af_info_lavcresample = { "Sample frequency conversion using libavcodec", "lavcresample", "Michael Niedermayer", "", AF_FLAGS_REENTRANT, af_open };