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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;
}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
};
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