Mercurial > mplayer.hg
view libaf/af_lavcresample.c @ 35605:71f19d352561
Fix bug with wrong video window size after end of fullscreen playback.
(This is a regression introduced in r35572.)
The normal size the window should have (media size or configured size)
depends on the current state of playback. In order to influence the
size, add a boolean parameter. This usually is True unless playback has
stopped.
author | ib |
---|---|
date | Wed, 19 Dec 2012 13:58:06 +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 };