/* * Filter to do simple decoding of matrixed surround sound. * This will provide a (basic) surround-sound effect from * audio encoded for Dolby Surround, Pro Logic etc. * * original author: Steve Davies * * 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. */ /* The principle: Make rear channels by extracting anti-phase data from the front channels, delay by 20ms and feed to rear in anti-phase */ /* SPLITREAR: Define to decode two distinct rear channels - this doesn't work so well in practice because separation in a passive matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so dialogue leaks to the rear. Still - give it a try and send feedback. Comment this define for old behavior of a single surround sent to rear in anti-phase */ #define SPLITREAR 1 #include #include #include #include "af.h" #include "dsp.h" #define L 32 // Length of fir filter #define LD 65536 // Length of delay buffer // 32 Tap fir filter loop unrolled #define FIR(x,w,y) \ y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \ + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \ + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \ + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \ + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \ + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \ + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \ + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31]) // Add to circular queue macro + update index #ifdef SPLITREAR #define ADDQUE(qi,rq,lq,r,l)\ lq[qi]=lq[qi+L]=(l);\ rq[qi]=rq[qi+L]=(r);\ qi=(qi-1)&(L-1); #else #define ADDQUE(qi,lq,l)\ lq[qi]=lq[qi+L]=(l);\ qi=(qi-1)&(L-1); #endif // Macro for updating queue index in delay queues #define UPDATEQI(qi) qi=(qi+1)&(LD-1) // instance data typedef struct af_surround_s { float lq[2*L]; // Circular queue for filtering left rear channel float rq[2*L]; // Circular queue for filtering right rear channel float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass float* dr; // Delay queue right rear channel float* dl; // Delay queue left rear channel float d; // Delay time int i; // Position in circular buffer int wi; // Write index for delay queue int ri; // Read index for delay queue }af_surround_t; // Initialization and runtime control static int control(struct af_instance* af, int cmd, void* arg) { af_surround_t *s = af->setup; switch(cmd){ case AF_CONTROL_REINIT:{ float fc; mp_audio_copy_config(af->data, (struct mp_audio*)arg); mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch*2); mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); if (af->data->nch != 4){ mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Only stereo input is supported.\n"); return AF_DETACH; } // Surround filer coefficients fc = 2.0 * 7000.0/(float)af->data->rate; if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){ mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Unable to design low-pass filter.\n"); return AF_ERROR; } // Free previous delay queues free(s->dl); free(s->dr); // Allocate new delay queues s->dl = calloc(LD,af->data->bps); s->dr = calloc(LD,af->data->bps); if((NULL == s->dl) || (NULL == s->dr)) mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n"); // Initialize delay queue index if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0)) return AF_ERROR; // printf("%i\n",s->wi); s->ri = 0; if((af->data->format != ((struct mp_audio*)arg)->format) || (af->data->bps != ((struct mp_audio*)arg)->bps)){ mp_audio_set_format((struct mp_audio*)arg, af->data->format); return AF_FALSE; } return AF_OK; } case AF_CONTROL_COMMAND_LINE:{ float d = 0; sscanf((char*)arg,"%f",&d); if ((d < 0) || (d > 1000)){ mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Invalid delay time, valid time values" " are 0ms to 1000ms current value is %0.3f ms\n",d); return AF_ERROR; } s->d = d; return AF_OK; } } return AF_UNKNOWN; } // Deallocate memory static void uninit(struct af_instance* af) { if(af->data) free(af->data->audio); free(af->data); free(af->setup); } // The beginnings of an active matrix... static float steering_matrix[][12] = { // LL RL LR RR LS RS // LLs RLs LRs RRs LC RC {.707, .0, .0, .707, .5, -.5, .5878, -.3928, .3928, -.5878, .5, .5}, }; // Experimental moving average dominance //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0; // Filter data through filter static struct mp_audio* play(struct af_instance* af, struct mp_audio* data){ af_surround_t* s = (af_surround_t*)af->setup; float* m = steering_matrix[0]; float* in = data->audio; // Input audio data float* out = NULL; // Output audio data float* end = in + data->len / sizeof(float); // Loop end int i = s->i; // Filter queue index int ri = s->ri; // Read index for delay queue int wi = s->wi; // Write index for delay queue if (AF_OK != RESIZE_LOCAL_BUFFER(af, data)) return NULL; out = af->data->audio; while(in < end){ /* Dominance: abs(in[0]) abs(in[1]); abs(in[0]+in[1]) abs(in[0]-in[1]); 10 * log( abs(in[0]) / (abs(in[1])|1) ); 10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */ /* About volume balancing... Surround encoding does the following: Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S So S should be extracted as: (Lt-Rt) But we are splitting the S to two output channels, so we must take 3dB off as we split it: Ls=Rs=.707*(Lt-Rt) Trouble is, Lt could be +1, Rt -1, so possibility that S will overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by 6dB (/2). This keeps the overall balance, but guarantees no overflow. */ // Output front left and right out[0] = m[0]*in[0] + m[1]*in[1]; out[1] = m[2]*in[0] + m[3]*in[1]; // Low-pass output @ 7kHz FIR((&s->lq[i]), s->w, s->dl[wi]); // Delay output by d ms out[2] = s->dl[ri]; #ifdef SPLITREAR // Low-pass output @ 7kHz FIR((&s->rq[i]), s->w, s->dr[wi]); // Delay output by d ms out[3] = s->dr[ri]; #else out[3] = -out[2]; #endif // Update delay queues indexes UPDATEQI(ri); UPDATEQI(wi); // Calculate and save surround in circular queue #ifdef SPLITREAR ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]); #else ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]); #endif // Next sample... in = &in[data->nch]; out = &out[af->data->nch]; } // Save indexes s->i = i; s->ri = ri; s->wi = wi; // Set output data data->audio = af->data->audio; data->len *= 2; mp_audio_set_channels_old(data, af->data->nch); return data; } static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; af->mul=2; af->data=calloc(1,sizeof(struct mp_audio)); af->setup=calloc(1,sizeof(af_surround_t)); if(af->data == NULL || af->setup == NULL) return AF_ERROR; ((af_surround_t*)af->setup)->d = 20; return AF_OK; } struct af_info af_info_surround = { "Surround decoder filter", "surround", "Steve Davies ", "", AF_FLAGS_NOT_REENTRANT, af_open };