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mpv/libao2/pl_surround.c
alex d65c8518de 64bit libao2 fix by Jens Axboe <mplayer-dev@kernel.dk>
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9634 b3059339-0415-0410-9bf9-f77b7e298cf2
2003-03-21 16:42:50 +00:00

252 lines
8.9 KiB
C

/*
This is an ao2 plugin 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., 675 Mass Ave, Cambridge, MA 02139, USA.
Original author: Steve Davies <steve@daviesfam.org>
*/
/* The principle: Make rear channels by extracting anti-phase data
from the front channels, delay by 20msec 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 behaviour of a single
// surround sent to rear in anti-phase
#define SPLITREAR
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
#include "remez.h"
#include "firfilter.c"
static ao_info_t info =
{
"Surround decoder plugin",
"surround",
"Steve Davies <steve@daviesfam.org>",
""
};
LIBAO_PLUGIN_EXTERN(surround)
// local data
typedef struct pl_surround_s
{
int passthrough; // Just be a "NO-OP"
int msecs; // Rear channel delay in milliseconds
int16_t* databuf; // Output audio buffer
int16_t* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
int16_t* Rs_delaybuf; // circular buffer to be used for delaying Rs audio
int delaybuf_len; // delaybuf buffer length in samples
int delaybuf_pos; // offset in buffer where we are reading/writing
double* filter_coefs_surround; // FIR filter coefficients for surround sound 7kHz lowpass
int rate; // input data rate
int format; // input format
int input_channels; // input channels
} pl_surround_t;
static pl_surround_t pl_surround={0,20,NULL,NULL,NULL,0,0,NULL,0,0,0};
// to set/get/query special features/parameters
static int control(int cmd,void *arg){
switch(cmd){
case AOCONTROL_PLUGIN_SET_LEN:
if (pl_surround.passthrough) return CONTROL_OK;
//fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg);
//fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len);
// Allocate an output buffer
if (pl_surround.databuf != NULL) {
free(pl_surround.databuf); pl_surround.databuf = NULL;
}
// Allocate output buffer
pl_surround.databuf = calloc(ao_plugin_data.len, 1);
// Return back smaller len so we don't get overflowed...
ao_plugin_data.len /= 2;
return CONTROL_OK;
}
return -1;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(){
fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels);
if (ao_plugin_data.channels != 2) {
fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n");
pl_surround.passthrough = 1;
return 1;
}
if (ao_plugin_data.format != AFMT_S16_NE) {
fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_NE audio format, using passthrough mode\n");
pl_surround.passthrough = 1;
return 1;
}
pl_surround.passthrough = 0;
/* Store info on input format to expect */
pl_surround.rate=ao_plugin_data.rate;
pl_surround.format=ao_plugin_data.format;
pl_surround.input_channels=ao_plugin_data.channels;
// Input 2 channels, output will be 4 - tell ao_plugin
ao_plugin_data.channels = 4;
ao_plugin_data.sz_mult /= 2;
// Figure out buffer space (in int16_ts) needed for the 15msec delay
// Extra 31 samples allow for lowpass filter delay (taps-1)
pl_surround.delaybuf_len = (pl_surround.rate * pl_surround.msecs / 1000) + 31;
// Allocate delay buffers
pl_surround.Ls_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
pl_surround.Rs_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffers are %d bytes each\n",
pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len*sizeof(int16_t));
pl_surround.delaybuf_pos = 0;
// Surround filer coefficients
pl_surround.filter_coefs_surround = calc_coefficients_7kHz_lowpass(pl_surround.rate);
//dump_filter_coefficients(pl_surround.filter_coefs_surround);
//testfilter(pl_surround.filter_coefs_surround, 32, pl_surround.rate);
return 1;
}
// close plugin
static void uninit(){
// fprintf(stderr, "pl_surround: uninit called!\n");
if (pl_surround.passthrough) return;
if(pl_surround.Ls_delaybuf)
free(pl_surround.Ls_delaybuf);
if(pl_surround.Rs_delaybuf)
free(pl_surround.Rs_delaybuf);
if(pl_surround.databuf) {
free(pl_surround.databuf);
pl_surround.databuf = NULL;
}
pl_surround.delaybuf_len=0;
}
// empty buffers
static void reset()
{
if (pl_surround.passthrough) return;
//fprintf(stderr, "pl_surround: reset called\n");
pl_surround.delaybuf_pos = 0;
memset(pl_surround.Ls_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
memset(pl_surround.Rs_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
}
// The beginnings of an active matrix...
static double 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 dominances
//static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
int16_t *in, *out;
int i, samples;
double *matrix = steering_matrix[0]; // later we'll index based on detected dominance
if (pl_surround.passthrough) return 1;
// fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
samples = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
out = pl_surround.databuf; in = (int16_t *)ao_plugin_data.data;
// Testing - place a 1kHz tone on Lt and Rt in anti-phase: should decode in S
//sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
//sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);
for (i=0; i<samples; i++) {
// 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 +32767, Rt -32768, 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] = matrix[0]*in[0] + matrix[1]*in[1];
out[1] = matrix[2]*in[0] + matrix[3]*in[1];
// output Ls and Rs - from 20msec ago, lowpass filtered @ 7kHz
out[2] = firfilter(pl_surround.Ls_delaybuf, pl_surround.delaybuf_pos,
pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
#ifdef SPLITREAR
out[3] = firfilter(pl_surround.Rs_delaybuf, pl_surround.delaybuf_pos,
pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
#else
out[3] = -out[2];
#endif
// calculate and save surround for 20msecs time
#ifdef SPLITREAR
pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] =
matrix[6]*in[0] + matrix[7]*in[1];
pl_surround.Rs_delaybuf[pl_surround.delaybuf_pos++] =
matrix[8]*in[0] + matrix[9]*in[1];
#else
pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos++] =
matrix[4]*in[0] + matrix[5]*in[1];
#endif
pl_surround.delaybuf_pos %= pl_surround.delaybuf_len;
// next samples...
in = &in[pl_surround.input_channels]; out = &out[4];
}
// Show some state
//printf("\npl_surround: delaybuf_pos=%d, samples=%d\r\033[A", pl_surround.delaybuf_pos, samples);
// Set output block/len
ao_plugin_data.data=pl_surround.databuf;
ao_plugin_data.len=samples*sizeof(int16_t)*4;
return 1;
}