Mercurial > mplayer.hg
view libaf/af_surround.c @ 26096:e6a565ec1a3b
New S3 VIDIX driver.
Provides support for S3 Trio and S3 Virge chipsets.
This deprecates the old Savage driver that worked with latest chips only.
(synchronized with vidix.sf.net r326 and r327)
author | ben |
---|---|
date | Fri, 29 Feb 2008 20:01:28 +0000 |
parents | b2402b4f0afa |
children | 089d5341f9bc |
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/* This is an libaf 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. * This program 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. * * This program 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 this program; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. Original author: Steve Davies <steve@daviesfam.org> */ /* 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 <stdio.h> #include <stdlib.h> #include <string.h> #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_s* af, int cmd, void* arg) { af_surround_t *s = af->setup; switch(cmd){ case AF_CONTROL_REINIT:{ float fc; af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch*2; af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; if (af->data->nch != 4){ af_msg(AF_MSG_ERROR,"[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)){ af_msg(AF_MSG_ERROR,"[surround] Unable to design low-pass filter.\n"); return AF_ERROR; } // Free previous delay queues if(s->dl) free(s->dl); if(s->dr) 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)) af_msg(AF_MSG_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 != ((af_data_t*)arg)->format) || (af->data->bps != ((af_data_t*)arg)->bps)){ ((af_data_t*)arg)->format = af->data->format; ((af_data_t*)arg)->bps = af->data->bps; return AF_FALSE; } return AF_OK; } case AF_CONTROL_COMMAND_LINE:{ float d = 0; sscanf((char*)arg,"%f",&d); if ((d < 0) || (d > 1000)){ af_msg(AF_MSG_ERROR,"[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_s* 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 af_data_t* play(struct af_instance_s* af, af_data_t* 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; data->nch = af->data->nch; return data; } static int af_open(af_instance_t* af){ af->control=control; af->uninit=uninit; af->play=play; af->mul=2; af->data=calloc(1,sizeof(af_data_t)); 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; } af_info_t af_info_surround = { "Surround decoder filter", "surround", "Steve Davies <steve@daviesfam.org>", "", AF_FLAGS_NOT_REENTRANT, af_open };