Mercurial > mplayer.hg
view libaf/af_lavcresample.c @ 32714:5248e989612a
Remove libswscale subdirectory.
libswscale has been moved to FFmpeg since some time and only kept in the
MPlayer repository for historical reasons and difficulty in moving code
complete with history between Subversion repositories.
FFmpeg now contains libswscale directly without indirection through an
svn:externals property, so no reason to keep libswscale in MPlayer remains.
author | diego |
---|---|
date | Thu, 20 Jan 2011 09:41:43 +0000 |
parents | 3af732078f52 |
children | 2b9bc3c2933d |
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/* * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> * * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include <stdio.h> #include <stdlib.h> #include <string.h> #include <inttypes.h> #include "config.h" #include "af.h" #include "libavcodec/avcodec.h" #include "libavutil/rational.h" // 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; int ctx_out_rate; int ctx_in_rate; int ctx_filter_size; int ctx_phase_shift; int ctx_linear; double ctx_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->ctx_out_rate != af->data->rate || s->ctx_in_rate != data->rate || s->ctx_filter_size != s->filter_length || s->ctx_phase_shift != s->phase_shift || s->ctx_linear != s->linear || s->ctx_cutoff != s->cutoff) { 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); s->ctx_out_rate = af->data->rate; s->ctx_in_rate = data->rate; s->ctx_filter_size = s->filter_length; s->ctx_phase_shift = s->phase_shift; s->ctx_linear = s->linear; s->ctx_cutoff = 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 };