mirror of https://github.com/mpv-player/mpv
252 lines
8.9 KiB
C
252 lines
8.9 KiB
C
/*
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This is an ao2 plugin to do simple decoding of matrixed surround
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sound. This will provide a (basic) surround-sound effect from
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audio encoded for Dolby Surround, Pro Logic etc.
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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Original author: Steve Davies <steve@daviesfam.org>
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*/
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/* The principle: Make rear channels by extracting anti-phase data
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from the front channels, delay by 20msec and feed to rear in anti-phase
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*/
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// SPLITREAR: Define to decode two distinct rear channels -
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// this doesn't work so well in practice because
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// separation in a passive matrix is not high.
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// C (dialogue) to Ls and Rs 14dB or so -
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// so dialogue leaks to the rear.
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// Still - give it a try and send feedback.
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// comment this define for old behaviour of a single
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// surround sent to rear in anti-phase
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#define SPLITREAR
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <unistd.h>
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#include "audio_out.h"
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#include "audio_plugin.h"
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#include "audio_plugin_internal.h"
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#include "afmt.h"
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#include "remez.h"
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#include "firfilter.c"
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static ao_info_t info =
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{
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"Surround decoder plugin",
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"surround",
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"Steve Davies <steve@daviesfam.org>",
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""
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};
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LIBAO_PLUGIN_EXTERN(surround)
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// local data
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typedef struct pl_surround_s
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{
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int passthrough; // Just be a "NO-OP"
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int msecs; // Rear channel delay in milliseconds
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int16_t* databuf; // Output audio buffer
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int16_t* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
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int16_t* Rs_delaybuf; // circular buffer to be used for delaying Rs audio
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int delaybuf_len; // delaybuf buffer length in samples
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int delaybuf_pos; // offset in buffer where we are reading/writing
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double* filter_coefs_surround; // FIR filter coefficients for surround sound 7kHz lowpass
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int rate; // input data rate
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int format; // input format
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int input_channels; // input channels
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} pl_surround_t;
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static pl_surround_t pl_surround={0,20,NULL,NULL,NULL,0,0,NULL,0,0,0};
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// to set/get/query special features/parameters
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static int control(int cmd,int arg){
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switch(cmd){
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case AOCONTROL_PLUGIN_SET_LEN:
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if (pl_surround.passthrough) return CONTROL_OK;
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//fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg);
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//fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len);
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// Allocate an output buffer
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if (pl_surround.databuf != NULL) {
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free(pl_surround.databuf); pl_surround.databuf = NULL;
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}
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// Allocate output buffer
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pl_surround.databuf = calloc(ao_plugin_data.len, 1);
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// Return back smaller len so we don't get overflowed...
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ao_plugin_data.len /= 2;
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return CONTROL_OK;
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}
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return -1;
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}
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(){
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fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels);
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if (ao_plugin_data.channels != 2) {
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fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n");
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pl_surround.passthrough = 1;
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return 1;
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}
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if (ao_plugin_data.format != AFMT_S16_LE) {
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fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_LE audio format, using passthrough mode\n");
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pl_surround.passthrough = 1;
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return 1;
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}
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pl_surround.passthrough = 0;
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/* Store info on input format to expect */
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pl_surround.rate=ao_plugin_data.rate;
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pl_surround.format=ao_plugin_data.format;
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pl_surround.input_channels=ao_plugin_data.channels;
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// Input 2 channels, output will be 4 - tell ao_plugin
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ao_plugin_data.channels = 4;
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ao_plugin_data.sz_mult /= 2;
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// Figure out buffer space (in int16_ts) needed for the 15msec delay
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// Extra 31 samples allow for lowpass filter delay (taps-1)
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pl_surround.delaybuf_len = (pl_surround.rate * pl_surround.msecs / 1000) + 31;
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// Allocate delay buffers
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pl_surround.Ls_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
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pl_surround.Rs_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
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fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffers are %d bytes each\n",
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pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len*sizeof(int16_t));
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pl_surround.delaybuf_pos = 0;
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// Surround filer coefficients
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pl_surround.filter_coefs_surround = calc_coefficients_7kHz_lowpass(pl_surround.rate);
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//dump_filter_coefficients(pl_surround.filter_coefs_surround);
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//testfilter(pl_surround.filter_coefs_surround, 32, pl_surround.rate);
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return 1;
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}
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// close plugin
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static void uninit(){
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// fprintf(stderr, "pl_surround: uninit called!\n");
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if (pl_surround.passthrough) return;
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if(pl_surround.Ls_delaybuf)
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free(pl_surround.Ls_delaybuf);
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if(pl_surround.Rs_delaybuf)
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free(pl_surround.Rs_delaybuf);
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if(pl_surround.databuf) {
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free(pl_surround.databuf);
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pl_surround.databuf = NULL;
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}
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pl_surround.delaybuf_len=0;
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}
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// empty buffers
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static void reset()
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{
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if (pl_surround.passthrough) return;
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//fprintf(stderr, "pl_surround: reset called\n");
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pl_surround.delaybuf_pos = 0;
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memset(pl_surround.Ls_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
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memset(pl_surround.Rs_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
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}
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// The beginnings of an active matrix...
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static double steering_matrix[][12] = {
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// LL RL LR RR LS RS LLs RLs LRs RRs LC RC
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{.707, .0, .0, .707, .5, -.5, .5878, -.3928, .3928, -.5878, .5, .5},
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};
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// Experimental moving average dominances
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static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
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// processes 'ao_plugin_data.len' bytes of 'data'
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// called for every block of data
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static int play(){
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int16_t *in, *out;
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int i, samples;
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double *matrix = steering_matrix[0]; // later we'll index based on detected dominance
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if (pl_surround.passthrough) return 1;
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// fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
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samples = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
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out = pl_surround.databuf; in = (int16_t *)ao_plugin_data.data;
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// Testing - place a 1kHz tone on Lt and Rt in anti-phase: should decode in S
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//sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
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//sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);
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for (i=0; i<samples; i++) {
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// Dominance:
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//abs(in[0]) abs(in[1]);
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//abs(in[0]+in[1]) abs(in[0]-in[1]);
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//10 * log( abs(in[0]) / (abs(in[1])|1) );
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//10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) );
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// About volume balancing...
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// Surround encoding does the following:
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// Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
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// So S should be extracted as:
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// (Lt-Rt)
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// But we are splitting the S to two output channels, so we
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// must take 3dB off as we split it:
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// Ls=Rs=.707*(Lt-Rt)
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// Trouble is, Lt could be +32767, Rt -32768, so possibility that S will
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// overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by 6dB (/2).
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// this keeps the overall balance, but guarantees no overflow.
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// output front left and right
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out[0] = matrix[0]*in[0] + matrix[1]*in[1];
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out[1] = matrix[2]*in[0] + matrix[3]*in[1];
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// output Ls and Rs - from 20msec ago, lowpass filtered @ 7kHz
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out[2] = firfilter(pl_surround.Ls_delaybuf, pl_surround.delaybuf_pos,
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pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
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#ifdef SPLITREAR
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out[3] = firfilter(pl_surround.Rs_delaybuf, pl_surround.delaybuf_pos,
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pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
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#else
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out[3] = -out[2];
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#endif
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// calculate and save surround for 20msecs time
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#ifdef SPLITREAR
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pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] =
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matrix[6]*in[0] + matrix[7]*in[1];
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pl_surround.Rs_delaybuf[pl_surround.delaybuf_pos++] =
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matrix[8]*in[0] + matrix[9]*in[1];
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#else
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pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos++] =
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matrix[4]*in[0] + matrix[5]*in[1];
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#endif
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pl_surround.delaybuf_pos %= pl_surround.delaybuf_len;
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// next samples...
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in = &in[pl_surround.input_channels]; out = &out[4];
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}
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// Show some state
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//printf("\npl_surround: delaybuf_pos=%d, samples=%d\r\033[A", pl_surround.delaybuf_pos, samples);
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// Set output block/len
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ao_plugin_data.data=pl_surround.databuf;
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ao_plugin_data.len=samples*sizeof(int16_t)*4;
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return 1;
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}
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