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-rw-r--r--libao2/pl_surround.c251
1 files changed, 0 insertions, 251 deletions
diff --git a/libao2/pl_surround.c b/libao2/pl_surround.c
deleted file mode 100644
index 6700bd5471..0000000000
--- a/libao2/pl_surround.c
+++ /dev/null
@@ -1,251 +0,0 @@
-/*
- 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 "libaf/af_format.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 != AF_FORMAT_S16_NE) {
- fprintf(stderr, "pl_surround: I'm dumb and can only handle AF_FORMAT_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;
-}
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