Mercurial > mplayer.hg
view libaf/af_lavcresample.c @ 20022:580fa198e8de
faacopts :object is in the range 1..4, not 0..3
author | nicodvb |
---|---|
date | Wed, 04 Oct 2006 09:04:14 +0000 |
parents | 2408715522a7 |
children | c4d9550c9faf |
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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 #define CHANS 6 int64_t ff_gcd(int64_t a, int64_t b); // Data for specific instances of this filter typedef struct af_resample_s{ struct AVResampleContext *avrctx; int16_t *in[CHANS]; 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 > CHANS) af->data->nch = CHANS; af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; af->mul.n = af->data->rate; af->mul.d = data->rate; af_frac_cancel(&af->mul); af->delay = 500*s->filter_length/(double)min(af->data->rate, data->rate); if(s->avrctx) av_resample_close(s->avrctx); s->avrctx= av_resample_init(af->mul.n, /*in_rate*/af->mul.d, 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:{ 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 - 1.0/s->filter_length, 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); if(af->setup){ af_resample_t *s = af->setup; if(s->avrctx) av_resample_close(s->avrctx); 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.n) / af->mul.d + 10; int16_t tmp[CHANS][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)); //FIXME free this maybe ;) } } 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 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.n=1; af->mul.d=1; af->data=calloc(1,sizeof(af_data_t)); s->filter_length= 16; s->cutoff= max(1.0 - 1.0/s->filter_length, 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, open };