mirror of
https://github.com/mpv-player/mpv
synced 2024-12-22 06:42:03 +00:00
0b4bfff65c
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@26772 b3059339-0415-0410-9bf9-f77b7e298cf2
198 lines
5.4 KiB
C
198 lines
5.4 KiB
C
/*
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <inttypes.h>
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#include "config.h"
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#include "af.h"
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#include "libavcodec/avcodec.h"
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#include "libavutil/rational.h"
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// Data for specific instances of this filter
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typedef struct af_resample_s{
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struct AVResampleContext *avrctx;
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int16_t *in[AF_NCH];
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int in_alloc;
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int index;
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int filter_length;
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int linear;
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int phase_shift;
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double cutoff;
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}af_resample_t;
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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af_resample_t* s = (af_resample_t*)af->setup;
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af_data_t *data= (af_data_t*)arg;
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int out_rate, test_output_res; // helpers for checking input format
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switch(cmd){
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case AF_CONTROL_REINIT:
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if((af->data->rate == data->rate) || (af->data->rate == 0))
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return AF_DETACH;
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af->data->nch = data->nch;
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if (af->data->nch > AF_NCH) af->data->nch = AF_NCH;
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af->data->format = AF_FORMAT_S16_NE;
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af->data->bps = 2;
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af->mul = (double)af->data->rate / data->rate;
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af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
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if(s->avrctx) av_resample_close(s->avrctx);
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s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff);
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// hack to make af_test_output ignore the samplerate change
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out_rate = af->data->rate;
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af->data->rate = data->rate;
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test_output_res = af_test_output(af, (af_data_t*)arg);
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af->data->rate = out_rate;
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return test_output_res;
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case AF_CONTROL_COMMAND_LINE:{
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s->cutoff= 0.0;
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sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff);
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if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
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return AF_OK;
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}
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case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
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af->data->rate = *(int*)arg;
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return AF_OK;
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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if(af->data)
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free(af->data->audio);
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free(af->data);
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if(af->setup){
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int i;
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af_resample_t *s = af->setup;
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if(s->avrctx) av_resample_close(s->avrctx);
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for (i=0; i < AF_NCH; i++)
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free(s->in[i]);
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free(s);
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}
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}
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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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{
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af_resample_t *s = af->setup;
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int i, j, consumed, ret;
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int16_t *in = (int16_t*)data->audio;
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int16_t *out;
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int chans = data->nch;
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int in_len = data->len/(2*chans);
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int out_len = in_len * af->mul + 10;
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int16_t tmp[AF_NCH][out_len];
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if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
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return NULL;
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out= (int16_t*)af->data->audio;
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out_len= min(out_len, af->data->len/(2*chans));
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if(s->in_alloc < in_len + s->index){
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s->in_alloc= in_len + s->index;
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for(i=0; i<chans; i++){
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s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t));
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}
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}
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if(chans==1){
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memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
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}else if(chans==2){
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for(j=0; j<in_len; j++){
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s->in[0][j + s->index]= *(in++);
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s->in[1][j + s->index]= *(in++);
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}
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}else{
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for(j=0; j<in_len; j++){
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for(i=0; i<chans; i++){
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s->in[i][j + s->index]= *(in++);
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}
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}
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}
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in_len += s->index;
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for(i=0; i<chans; i++){
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ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans);
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}
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out_len= ret;
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s->index= in_len - consumed;
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for(i=0; i<chans; i++){
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memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
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}
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if(chans==1){
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memcpy(out, tmp[0], out_len*sizeof(int16_t));
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}else if(chans==2){
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for(j=0; j<out_len; j++){
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*(out++)= tmp[0][j];
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*(out++)= tmp[1][j];
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}
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}else{
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for(j=0; j<out_len; j++){
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for(i=0; i<chans; i++){
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*(out++)= tmp[i][j];
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}
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}
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}
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data->audio = af->data->audio;
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data->len = out_len*chans*2;
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data->rate = af->data->rate;
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return data;
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}
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static int af_open(af_instance_t* af){
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af_resample_t *s = calloc(1,sizeof(af_resample_t));
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af->control=control;
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af->uninit=uninit;
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af->play=play;
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af->mul=1;
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af->data=calloc(1,sizeof(af_data_t));
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s->filter_length= 16;
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s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
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s->phase_shift= 10;
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// s->setup = RSMP_INT | FREQ_SLOPPY;
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af->setup=s;
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return AF_OK;
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}
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af_info_t af_info_lavcresample = {
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"Sample frequency conversion using libavcodec",
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"lavcresample",
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"Michael Niedermayer",
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"",
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AF_FLAGS_REENTRANT,
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af_open
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};
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