mirror of
https://github.com/mpv-player/mpv
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fe3c4810e1
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32624 b3059339-0415-0410-9bf9-f77b7e298cf2
189 lines
5.5 KiB
C
189 lines
5.5 KiB
C
/*
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* Copyright (C) 2002 Anders Johansson ajh@watri.uwa.edu.au
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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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/* This filter adds a sub-woofer channels to the audio stream by
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averaging the left and right channel and low-pass filter them. The
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low-pass filter is implemented as a 4th order IIR Butterworth
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filter, with a variable cutoff frequency between 10 and 300 Hz. The
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filter gives 24dB/octave attenuation. There are two runtime
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controls one for setting which channel to insert the sub-audio into
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called AF_CONTROL_SUB_CH and one for setting the cutoff frequency
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called AF_CONTROL_SUB_FC.
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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 "af.h"
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#include "dsp.h"
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// Q value for low-pass filter
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#define Q 1.0
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// Analog domain biquad section
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typedef struct{
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float a[3]; // Numerator coefficients
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float b[3]; // Denominator coefficients
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} biquad_t;
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// S-parameters for designing 4th order Butterworth filter
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static biquad_t sp[2] = {{{1.0,0.0,0.0},{1.0,0.765367,1.0}},
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{{1.0,0.0,0.0},{1.0,1.847759,1.0}}};
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// Data for specific instances of this filter
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typedef struct af_sub_s
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{
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float w[2][4]; // Filter taps for low-pass filter
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float q[2][2]; // Circular queues
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float fc; // Cutoff frequency [Hz] for low-pass filter
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float k; // Filter gain;
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int ch; // Channel number which to insert the filtered data
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}af_sub_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_sub_t* s = af->setup;
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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 = max(s->ch+1,((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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// Design low-pass filter
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s->k = 1.0;
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if((-1 == af_filter_szxform(sp[0].a, sp[0].b, Q, s->fc,
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(float)af->data->rate, &s->k, s->w[0])) ||
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(-1 == af_filter_szxform(sp[1].a, sp[1].b, Q, s->fc,
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(float)af->data->rate, &s->k, s->w[1])))
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return AF_ERROR;
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return af_test_output(af,(af_data_t*)arg);
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}
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case AF_CONTROL_COMMAND_LINE:{
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int ch=5;
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float fc=60.0;
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sscanf(arg,"%f:%i", &fc , &ch);
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if(AF_OK != control(af,AF_CONTROL_SUB_CH | AF_CONTROL_SET, &ch))
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return AF_ERROR;
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return control(af,AF_CONTROL_SUB_FC | AF_CONTROL_SET, &fc);
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}
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case AF_CONTROL_SUB_CH | AF_CONTROL_SET: // Requires reinit
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// Sanity check
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if((*(int*)arg >= AF_NCH) || (*(int*)arg < 0)){
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mp_msg(MSGT_AFILTER, MSGL_ERR, "[sub] Subwoofer channel number must be between "
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" 0 and %i current value is %i\n", AF_NCH-1, *(int*)arg);
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return AF_ERROR;
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}
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s->ch = *(int*)arg;
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return AF_OK;
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case AF_CONTROL_SUB_CH | AF_CONTROL_GET:
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*(int*)arg = s->ch;
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return AF_OK;
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case AF_CONTROL_SUB_FC | AF_CONTROL_SET: // Requires reinit
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// Sanity check
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if((*(float*)arg > 300) || (*(float*)arg < 20)){
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mp_msg(MSGT_AFILTER, MSGL_ERR, "[sub] Cutoff frequency must be between 20Hz and"
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" 300Hz current value is %0.2f",*(float*)arg);
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return AF_ERROR;
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}
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// Set cutoff frequency
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s->fc = *(float*)arg;
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return AF_OK;
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case AF_CONTROL_SUB_FC | AF_CONTROL_GET:
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*(float*)arg = s->fc;
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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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free(af->data);
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free(af->setup);
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}
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#ifndef IIR
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#define IIR(in,w,q,out) { \
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float h0 = (q)[0]; \
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float h1 = (q)[1]; \
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float hn = (in) - h0 * (w)[0] - h1 * (w)[1]; \
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out = hn + h0 * (w)[2] + h1 * (w)[3]; \
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(q)[1] = h0; \
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(q)[0] = hn; \
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}
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#endif
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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_sub_t* s = af->setup; // Setup for this instance
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float* a = c->audio; // Audio data
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int len = c->len/4; // Number of samples in current audio block
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int nch = c->nch; // Number of channels
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int ch = s->ch; // Channel in which to insert the sub audio
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register int i;
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// Run filter
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for(i=0;i<len;i+=nch){
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// Average left and right
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register float x = 0.5 * (a[i] + a[i+1]);
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IIR(x * s->k, s->w[0], s->q[0], x);
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IIR(x , s->w[1], s->q[1], a[i+ch]);
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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_sub_t* s;
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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=s=calloc(1,sizeof(af_sub_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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// Set default values
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s->ch = 5; // Channel nr 6
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s->fc = 60; // Cutoff frequency 60Hz
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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_sub = {
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"Audio filter for adding a sub-base channel",
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"sub",
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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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