mirror of
https://github.com/mpv-player/mpv
synced 2024-12-21 06:14:32 +00:00
7deec05ea0
Change the audio filters to use a double instead of rationals for the ratio of output to input size. The rationals could overflow when calculating the overall ratio of a filter chain and gave no real advantage compared to doubles. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24916 b3059339-0415-0410-9bf9-f77b7e298cf2
161 lines
4.6 KiB
C
161 lines
4.6 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU General Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
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//
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//=============================================================================
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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 <math.h>
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#include <limits.h>
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#include "af.h"
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// Data for specific instances of this filter
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typedef struct af_comp_s
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{
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int enable[AF_NCH]; // Enable/disable / channel
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float time[AF_NCH]; // Forgetting factor for power estimate
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float pow[AF_NCH]; // Estimated power level [dB]
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float tresh[AF_NCH]; // Threshold [dB]
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int attack[AF_NCH]; // Attack time [ms]
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int release[AF_NCH]; // Release time [ms]
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float ratio[AF_NCH]; // Compression ratio
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}af_comp_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_comp_t* s = (af_comp_t*)af->setup;
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int i;
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switch(cmd){
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case AF_CONTROL_REINIT:
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// Sanity check
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if(!arg) return AF_ERROR;
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af->data->rate = ((af_data_t*)arg)->rate;
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af->data->nch = ((af_data_t*)arg)->nch;
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af->data->format = AF_FORMAT_FLOAT_NE;
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af->data->bps = 4;
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// Time constant set to 0.1s
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// s->alpha = (1.0/0.2)/(2.0*M_PI*(float)((af_data_t*)arg)->rate);
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return af_test_output(af,(af_data_t*)arg);
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case AF_CONTROL_COMMAND_LINE:{
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/* float v=-10.0; */
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/* float vol[AF_NCH]; */
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/* float s=0.0; */
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/* float clipp[AF_NCH]; */
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/* int i; */
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/* sscanf((char*)arg,"%f:%f", &v, &s); */
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/* for(i=0;i<AF_NCH;i++){ */
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/* vol[i]=v; */
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/* clipp[i]=s; */
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/* } */
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/* if(AF_OK != control(af,AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_SET, clipp)) */
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/* return AF_ERROR; */
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/* return control(af,AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET, vol); */
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}
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case AF_CONTROL_COMP_ON_OFF | AF_CONTROL_SET:
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memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int));
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return AF_OK;
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case AF_CONTROL_COMP_ON_OFF | AF_CONTROL_GET:
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memcpy((int*)arg,s->enable,AF_NCH*sizeof(int));
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return AF_OK;
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case AF_CONTROL_COMP_THRESH | AF_CONTROL_SET:
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return af_from_dB(AF_NCH,(float*)arg,s->tresh,20.0,-60.0,-1.0);
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case AF_CONTROL_COMP_THRESH | AF_CONTROL_GET:
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return af_to_dB(AF_NCH,s->tresh,(float*)arg,10.0);
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case AF_CONTROL_COMP_ATTACK | AF_CONTROL_SET:
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return af_from_ms(AF_NCH,(float*)arg,s->attack,af->data->rate,500.0,0.1);
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case AF_CONTROL_COMP_ATTACK | AF_CONTROL_GET:
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return af_to_ms(AF_NCH,s->attack,(float*)arg,af->data->rate);
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case AF_CONTROL_COMP_RELEASE | AF_CONTROL_SET:
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return af_from_ms(AF_NCH,(float*)arg,s->release,af->data->rate,3000.0,10.0);
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case AF_CONTROL_COMP_RELEASE | AF_CONTROL_GET:
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return af_to_ms(AF_NCH,s->release,(float*)arg,af->data->rate);
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case AF_CONTROL_COMP_RATIO | AF_CONTROL_SET:
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for(i=0;i<AF_NCH;i++)
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s->ratio[i] = clamp(((float*)arg)[i],1.0,10.0);
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return AF_OK;
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case AF_CONTROL_COMP_RATIO | AF_CONTROL_GET:
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for(i=0;i<AF_NCH;i++)
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((float*)arg)[i] = s->ratio[i];
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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);
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if(af->setup)
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free(af->setup);
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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_data_t* c = data; // Current working data
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af_comp_t* s = (af_comp_t*)af->setup; // Setup for this instance
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float* a = (float*)c->audio; // Audio data
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int len = c->len/4; // Number of samples
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int ch = 0; // Channel counter
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register int nch = c->nch; // Number of channels
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register int i = 0;
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// Compress/expand
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for(ch = 0; ch < nch ; ch++){
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if(s->enable[ch]){
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float t = 1.0 - s->time[ch];
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for(i=ch;i<len;i+=nch){
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register float x = a[i];
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register float pow = x*x;
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s->pow[ch] = t*s->pow[ch] +
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pow*s->time[ch]; // LP filter
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if(pow < s->pow[ch]){
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;
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}
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else{
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;
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}
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a[i] = x;
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}
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}
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}
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return c;
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}
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// Allocate memory and set function pointers
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static int af_open(af_instance_t* af){
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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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af->setup=calloc(1,sizeof(af_comp_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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return AF_OK;
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}
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// Description of this filter
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af_info_t af_info_comp = {
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"Compressor/expander audio filter",
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"comp",
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"Anders",
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"",
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AF_FLAGS_NOT_REENTRANT,
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af_open
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};
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