mirror of https://github.com/mpv-player/mpv
249 lines
8.6 KiB
C
249 lines
8.6 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 "af.h"
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#include "dsp.h"
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// instance data
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typedef struct af_surround_s
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{
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float msecs; // Rear channel delay in milliseconds
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float* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
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float* 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_rpos; // offset in buffer where we are reading
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int delaybuf_wpos; // offset in buffer where we are writing
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float filter_coefs_surround[32]; // FIR filter coefficients for surround sound 7kHz lowpass
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} af_surround_t;
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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af_surround_t *instance=af->setup;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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float cutoff;
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af->data->rate = ((af_data_t*)arg)->rate;
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af->data->nch = ((af_data_t*)arg)->nch*2;
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af->data->format = ((af_data_t*)arg)->format;
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af->data->bps = ((af_data_t*)arg)->bps;
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af_msg(AF_MSG_DEBUG0, "[surround]: rear delay=%0.2fms.\n", instance->msecs);
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if (af->data->nch != 4){
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af_msg(AF_MSG_ERROR,"Only Stereo input is supported, filter disabled.\n");
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return AF_DETACH;
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}
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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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// Double size to make virtual ringbuffer easier
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instance->delaybuf_len = ((af->data->rate * instance->msecs / 1000)+31)*2;
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// Free old buffers
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if (instance->Ls_delaybuf != NULL)
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free(instance->Ls_delaybuf);
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if (instance->Rs_delaybuf != NULL)
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free(instance->Rs_delaybuf);
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// Allocate new buffers
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instance->Ls_delaybuf=(void*)calloc(instance->delaybuf_len,sizeof(*instance->Ls_delaybuf));
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instance->Rs_delaybuf=(void*)calloc(instance->delaybuf_len,sizeof(*instance->Rs_delaybuf));
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af_msg(AF_MSG_DEBUG1, "Delay buffers are %d samples each.\n", instance->delaybuf_len);
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instance->delaybuf_wpos = 0;
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instance->delaybuf_rpos = 32; // compensate for fir delay
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// Surround filer coefficients
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cutoff = af->data->rate/7000;
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if (-1 == design_fir(32, instance->filter_coefs_surround, &cutoff, LP|KAISER, 10.0)) {
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af_msg(AF_MSG_ERROR,"[surround] Unable to design prototype filter.\n");
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return AF_ERROR;
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}
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return AF_OK;
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}
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case AF_CONTROL_COMMAND_LINE:{
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float d = 0;
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sscanf((char*)arg,"%f",&d);
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if (d<0){
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af_msg(AF_MSG_ERROR,"Error setting rear delay length in af_surround. Delay has to be positive.\n");
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return AF_ERROR;
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}
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instance->msecs=d;
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return AF_OK;
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}
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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af_surround_t *instance=af->setup;
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if(af->data->audio)
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free(af->data->audio);
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if(af->data)
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free(af->data);
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if(instance->Ls_delaybuf)
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free(instance->Ls_delaybuf);
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if(instance->Rs_delaybuf)
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free(instance->Rs_delaybuf);
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free(af->setup);
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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
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// LLs RLs LRs RRs LC RC
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{.707, .0, .0, .707, .5, -.5,
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.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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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data){
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af_surround_t* instance = (af_surround_t*)af->setup;
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int16_t* in = data->audio;
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int16_t* 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 (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
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return NULL;
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out = af->data->audio;
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// fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
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samples = data->len / (data->nch * sizeof(int16_t));
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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] = fir(32, instance->filter_coefs_surround,
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&instance->Ls_delaybuf[instance->delaybuf_rpos +
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instance->delaybuf_len/2]);
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#ifdef SPLITREAR
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out[3] = fir(32, instance->filter_coefs_surround,
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&instance->Rs_delaybuf[instance->delaybuf_rpos +
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instance->delaybuf_len/2]);
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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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instance->Ls_delaybuf[instance->delaybuf_wpos] =
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instance->Ls_delaybuf[instance->delaybuf_wpos + instance->delaybuf_len/2] =
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matrix[6]*in[0] + matrix[7]*in[1];
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instance->Rs_delaybuf[instance->delaybuf_wpos] =
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instance->Rs_delaybuf[instance->delaybuf_wpos++ + instance->delaybuf_len/2] =
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matrix[8]*in[0] + matrix[9]*in[1];
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#else
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instance->Ls_delaybuf[instance->delaybuf_wpos] =
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instance->Ls_delaybuf[instance->delaybuf_wpos++ + instance->delaybuf_len/2] =
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matrix[4]*in[0] + matrix[5]*in[1];
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#endif
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instance->delaybuf_rpos++;
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instance->delaybuf_wpos %= instance->delaybuf_len/2;
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instance->delaybuf_rpos %= instance->delaybuf_len/2;
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// next samples...
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in = &in[data->nch]; out = &out[af->data->nch];
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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 data
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data->audio = af->data->audio;
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data->len = (data->len*af->mul.n)/af->mul.d;
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data->nch = af->data->nch;
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return data;
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}
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static int open(af_instance_t* af){
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af_surround_t *pl_surround;
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af->control=control;
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af->uninit=uninit;
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af->play=play;
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af->mul.n=2;
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af->mul.d=1;
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af->data=calloc(1,sizeof(af_data_t));
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af->setup=pl_surround=calloc(1,sizeof(af_surround_t));
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pl_surround->msecs=20;
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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return AF_OK;
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}
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af_info_t af_info_surround =
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{
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"Surround decoder filter",
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"surround",
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"Steve Davies <steve@daviesfam.org>",
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"",
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AF_FLAGS_REENTRANT,
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open
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};
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