mirror of
https://github.com/mpv-player/mpv
synced 2025-03-19 18:05:21 +00:00
Dolby Surround decoding audio plugin
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3314 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
steve
parent
5c4db58d5f
commit
067b8b5805
177
libao2/pl_surround.c
Normal file
177
libao2/pl_surround.c
Normal file
@ -0,0 +1,177 @@
|
||||
/*
|
||||
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 15msec and feed to rear in anti-phase
|
||||
www.dolby.com has the background
|
||||
*/
|
||||
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
#include "audio_out.h"
|
||||
#include "audio_plugin.h"
|
||||
#include "audio_plugin_internal.h"
|
||||
#include "afmt.h"
|
||||
|
||||
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* delaybuf; // circular buffer to be used for delaying audio signal
|
||||
int delaybuf_len; // local buffer length in samples
|
||||
int delaybuf_ptr; // offset in buffer where we are reading/writing
|
||||
int rate; // input data rate
|
||||
int format; // input format
|
||||
int input_channels; // input channels
|
||||
|
||||
} pl_surround_t;
|
||||
|
||||
static pl_surround_t pl_surround={0,15,NULL,NULL,0,0,0,0,0};
|
||||
|
||||
// to set/get/query special features/parameters
|
||||
static int control(int cmd,int 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;
|
||||
}
|
||||
pl_surround.databuf = calloc(ao_plugin_data.len, 1);
|
||||
// Return back smaller len so we don't get overflowed... (??seems the right thing to do?)
|
||||
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_LE) {
|
||||
fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_LE 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 needed for the 15msec delay
|
||||
pl_surround.delaybuf_len = pl_surround.rate * pl_surround.msecs / 1000;
|
||||
// Allocate delay buffer
|
||||
pl_surround.delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
|
||||
fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffer is %d samples\n",
|
||||
pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len);
|
||||
pl_surround.delaybuf_ptr = 0;
|
||||
|
||||
return 1;
|
||||
}
|
||||
|
||||
// close plugin
|
||||
static void uninit(){
|
||||
// fprintf(stderr, "pl_surround: uninit called!\n");
|
||||
if (pl_surround.passthrough) return;
|
||||
if(pl_surround.delaybuf)
|
||||
free(pl_surround.delaybuf);
|
||||
if(pl_surround.databuf)
|
||||
free(pl_surround.databuf);
|
||||
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_ptr = 0;
|
||||
memset(pl_surround.delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
|
||||
}
|
||||
|
||||
|
||||
// processes 'ao_plugin_data.len' bytes of 'data'
|
||||
// called for every block of data
|
||||
static int play(){
|
||||
int16_t *in, *out;
|
||||
int i, samples;
|
||||
int surround;
|
||||
|
||||
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;
|
||||
for (i=0; i<samples; i++) {
|
||||
// front left and right
|
||||
out[0] = in[0];
|
||||
out[1] = in[1];
|
||||
// surround - from 15msec ago
|
||||
out[2] = pl_surround.delaybuf[pl_surround.delaybuf_ptr];
|
||||
out[3] = -out[2];
|
||||
// calculate and save surround for 15msecs time
|
||||
pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = (in[0]/2 - in[1]/2);
|
||||
pl_surround.delaybuf_ptr %= pl_surround.delaybuf_len;
|
||||
// next samples...
|
||||
in = &in[pl_surround.input_channels]; out = &out[4];
|
||||
}
|
||||
|
||||
// Set output block/len
|
||||
ao_plugin_data.data=pl_surround.databuf;
|
||||
ao_plugin_data.len=samples*sizeof(int16_t)*4;
|
||||
return 1;
|
||||
}
|
||||
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user