/* * Copyright (C) 2002 Anders Johansson ajh@watri.uwa.edu.au * * 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. */ /* This filter adds a sub-woofer channels to the audio stream by averaging the left and right channel and low-pass filter them. The low-pass filter is implemented as a 4th order IIR Butterworth filter, with a variable cutoff frequency between 10 and 300 Hz. The filter gives 24dB/octave attenuation. There are two runtime controls one for setting which channel to insert the sub-audio into called AF_CONTROL_SUB_CH and one for setting the cutoff frequency called AF_CONTROL_SUB_FC. */ #include #include #include #include "common/common.h" #include "af.h" #include "dsp.h" // Q value for low-pass filter #define Q 1.0 // Analog domain biquad section typedef struct{ float a[3]; // Numerator coefficients float b[3]; // Denominator coefficients } biquad_t; // S-parameters for designing 4th order Butterworth filter static const biquad_t sp[2] = {{{1.0,0.0,0.0},{1.0,0.765367,1.0}}, {{1.0,0.0,0.0},{1.0,1.847759,1.0}}}; // Data for specific instances of this filter typedef struct af_sub_s { float w[2][4]; // Filter taps for low-pass filter float q[2][2]; // Circular queues float fc; // Cutoff frequency [Hz] for low-pass filter float k; // Filter gain; int ch; // Channel number which to insert the filtered data }af_sub_t; // Initialization and runtime control static int control(struct af_instance* af, int cmd, void* arg) { af_sub_t* s = af->priv; switch(cmd){ case AF_CONTROL_REINIT:{ // Sanity check if(!arg) return AF_ERROR; af->data->rate = ((struct mp_audio*)arg)->rate; mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch)); mp_audio_set_format(af->data, AF_FORMAT_FLOAT); // Design low-pass filter s->k = 1.0; if((-1 == af_filter_szxform(sp[0].a, sp[0].b, Q, s->fc, (float)af->data->rate, &s->k, s->w[0])) || (-1 == af_filter_szxform(sp[1].a, sp[1].b, Q, s->fc, (float)af->data->rate, &s->k, s->w[1]))) return AF_ERROR; return af_test_output(af,(struct mp_audio*)arg); } } return AF_UNKNOWN; } #ifndef IIR #define IIR(in,w,q,out) { \ float h0 = (q)[0]; \ float h1 = (q)[1]; \ float hn = (in) - h0 * (w)[0] - h1 * (w)[1]; \ out = hn + h0 * (w)[2] + h1 * (w)[3]; \ (q)[1] = h0; \ (q)[0] = hn; \ } #endif // Filter data through filter static int filter(struct af_instance* af, struct mp_audio* data, int flags) { struct mp_audio* c = data; // Current working data af_sub_t* s = af->priv; // Setup for this instance float* a = c->planes[0]; // Audio data int len = c->samples*c->nch; // Number of samples in current audio block int nch = c->nch; // Number of channels int ch = s->ch; // Channel in which to insert the sub audio register int i; // Run filter for(i=0;ik, s->w[0], s->q[0], x); IIR(x , s->w[1], s->q[1], a[i+ch]); } return 0; } // Allocate memory and set function pointers static int af_open(struct af_instance* af){ af->control=control; af->filter=filter; return AF_OK; } #define OPT_BASE_STRUCT af_sub_t const struct af_info af_info_sub = { .info = "Audio filter for adding a sub-base channel", .name = "sub", .flags = AF_FLAGS_NOT_REENTRANT, .open = af_open, .priv_size = sizeof(af_sub_t), .options = (const struct m_option[]) { OPT_FLOATRANGE("fc", fc, 0, 20, 300, OPTDEF_FLOAT(60.0)), OPT_INTRANGE("ch", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(5)), {0} }, };