RepoMirrors/mpv
1
0
Fork 0
mirror of https://github.com/mpv-player/mpv synced 2024-12-16 03:45:23 +00:00
Code Issues Releases Wiki Activity
b4b9552c60
mpv/libmpcodecs/ad_libdca.c
ulion 098170a7e2 Prevent from outputing mass of 'skip' log messages in verbose level. 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@25306 b3059339-0415-0410-9bf9-f77b7e298cf2
2007-12-05 18:24:40 +00:00

344 lines
8.8 KiB
C
Raw Blame History

/*
* ad_libdca.c : DTS Coherent Acoustics stream decoder using libdca
* This file is partially based on dtsdec.c r9036 from FFmpeg and ad_liba52.c
* Copyright (C) 2007 Roberto Togni
*
* This file is part of MPlayer.
*
* MPlayer 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.
*
* MPlayer 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 MPlayer; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <assert.h>
#include "config.h"
#include "mp_msg.h"
#include "ad_internal.h"
#include <dts.h>
static ad_info_t info =
{
"DTS decoding with libdca",
"libdca",
"Roberto Togni",
"",
""
};
LIBAD_EXTERN(libdca)
#define DTSBUFFER_SIZE 18726
#define HEADER_SIZE 14
#define CONVERT_LEVEL 1
#define CONVERT_BIAS 0
static const char ch2flags[6] = {
DTS_MONO,
DTS_STEREO,
DTS_3F,
DTS_2F2R,
DTS_3F2R,
DTS_3F2R | DTS_LFE
};
static inline int16_t convert(sample_t s)
{
int i = s * 0x7fff;
return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
}
static void convert2s16_multi(sample_t *f, int16_t *s16, int flags, int ch_out)
{
int i;
switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
case DTS_MONO:
if (ch_out == 1)
for(i = 0; i < 256; i++)
s16[i] = convert(f[i]);
else
for(i = 0; i < 256; i++){
s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0;
s16[5*i+4] = convert(f[i]);
}
break;
case DTS_CHANNEL:
case DTS_STEREO:
case DTS_DOLBY:
for(i = 0; i < 256; i++){
s16[2*i] = convert(f[i]);
s16[2*i+1] = convert(f[i+256]);
}
break;
case DTS_3F:
for(i = 0; i < 256; i++){
s16[3*i] = convert(f[i+256]);
s16[3*i+1] = convert(f[i+512]);
s16[3*i+2] = convert(f[i]);
}
break;
case DTS_2F2R:
for(i = 0; i < 256; i++){
s16[4*i] = convert(f[i]);
s16[4*i+1] = convert(f[i+256]);
s16[4*i+2] = convert(f[i+512]);
s16[4*i+3] = convert(f[i+768]);
}
break;
case DTS_3F2R:
for(i = 0; i < 256; i++){
s16[5*i] = convert(f[i+256]);
s16[5*i+1] = convert(f[i+512]);
s16[5*i+2] = convert(f[i+768]);
s16[5*i+3] = convert(f[i+1024]);
s16[5*i+4] = convert(f[i]);
}
break;
case DTS_MONO | DTS_LFE:
for(i = 0; i < 256; i++){
s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0;
s16[6*i+4] = convert(f[i]);
s16[6*i+5] = convert(f[i+256]);
}
break;
case DTS_CHANNEL | DTS_LFE:
case DTS_STEREO | DTS_LFE:
case DTS_DOLBY | DTS_LFE:
for(i = 0; i < 256; i++){
s16[6*i] = convert(f[i]);
s16[6*i+1] = convert(f[i+256]);
s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
s16[6*i+5] = convert(f[i+512]);
}
break;
case DTS_3F | DTS_LFE:
for(i = 0; i < 256; i++){
s16[6*i] = convert(f[i+256]);
s16[6*i+1] = convert(f[i+512]);
s16[6*i+2] = s16[6*i+3] = 0;
s16[6*i+4] = convert(f[i]);
s16[6*i+5] = convert(f[i+768]);
}
break;
case DTS_2F2R | DTS_LFE:
for(i = 0; i < 256; i++){
s16[6*i] = convert(f[i]);
s16[6*i+1] = convert(f[i+256]);
s16[6*i+2] = convert(f[i+512]);
s16[6*i+3] = convert(f[i+768]);
s16[6*i+4] = 0;
s16[6*i+5] = convert(f[1024]);
}
break;
case DTS_3F2R | DTS_LFE:
for(i = 0; i < 256; i++){
s16[6*i] = convert(f[i+256]);
s16[6*i+1] = convert(f[i+512]);
s16[6*i+2] = convert(f[i+768]);
s16[6*i+3] = convert(f[i+1024]);
s16[6*i+4] = convert(f[i]);
s16[6*i+5] = convert(f[i+1280]);
}
break;
}
}
static void channels_info(int flags)
{
int lfe = 0;
char lfestr[5] = "";
if (flags & DTS_LFE) {
lfe = 1;
strcpy(lfestr, "+lfe");
}
mp_msg(MSGT_DECAUDIO, MSGL_V, "DTS: ");
switch(flags & DTS_CHANNEL_MASK){
case DTS_MONO:
mp_msg(MSGT_DECAUDIO, MSGL_V, "1.%d (mono%s)", lfe, lfestr);
break;
case DTS_CHANNEL:
mp_msg(MSGT_DECAUDIO, MSGL_V, "2.%d (channel%s)", lfe, lfestr);
break;
case DTS_STEREO:
mp_msg(MSGT_DECAUDIO, MSGL_V, "2.%d (stereo%s)", lfe, lfestr);
break;
case DTS_3F:
mp_msg(MSGT_DECAUDIO, MSGL_V, "3.%d (3f%s)", lfe, lfestr);
break;
case DTS_2F2R:
mp_msg(MSGT_DECAUDIO, MSGL_V, "4.%d (2f+2r%s)", lfe, lfestr);
break;
case DTS_3F2R:
mp_msg(MSGT_DECAUDIO, MSGL_V, "5.%d (3f+2r%s)", lfe, lfestr);
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_V, "x.%d (unknown%s)", lfe, lfestr);
}
mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
}
static int dts_sync(sh_audio_t *sh, int *flags)
{
dts_state_t *s = sh->context;
int length;
int sample_rate;
int frame_length;
int bit_rate;
sh->a_in_buffer_len=0;
while(1) {
while(sh->a_in_buffer_len < HEADER_SIZE) {
int c = demux_getc(sh->ds);
if(c < 0)
return -1;
sh->a_in_buffer[sh->a_in_buffer_len++] = c;
}
length = dts_syncinfo(s, sh->a_in_buffer, flags, &sample_rate,
&bit_rate, &frame_length);
if(length >= HEADER_SIZE)
break;
// mp_msg(MSGT_DECAUDIO, MSGL_V, "skip\n");
memmove(sh->a_in_buffer, sh->a_in_buffer+1, HEADER_SIZE-1);
--sh->a_in_buffer_len;
}
demux_read_data(sh->ds, sh->a_in_buffer + HEADER_SIZE, length - HEADER_SIZE);
sh->samplerate = sample_rate;
sh->i_bps = bit_rate/8;
return length;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen, int maxlen)
{
dts_state_t *s = sh->context;
int16_t *out_samples = (int16_t*)buf;
int flags;
level_t level;
sample_t bias;
int nblocks;
int i;
int data_size = 0;
if(!sh->a_in_buffer_len)
if(dts_sync(sh, &flags) < 0) return -1; /* EOF */
sh->a_in_buffer_len=0;
flags &= ~(DTS_CHANNEL_MASK | DTS_LFE);
flags |= ch2flags[sh->channels - 1];
level = CONVERT_LEVEL;
bias = CONVERT_BIAS;
flags |= DTS_ADJUST_LEVEL;
if(dts_frame(s, sh->a_in_buffer, &flags, &level, bias)) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_frame() failed\n");
goto end;
}
nblocks = dts_blocks_num(s);
for(i = 0; i < nblocks; i++) {
if(dts_block(s)) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_block() failed\n");
goto end;
}
convert2s16_multi(dts_samples(s), out_samples, flags, sh->channels);
out_samples += 256 * sh->channels;
data_size += 256 * sizeof(int16_t) * sh->channels;
}
end:
return data_size;
}
static int preinit(sh_audio_t *sh)
{
/* 256 = samples per block, 16 = max number of blocks */
sh->audio_out_minsize = audio_output_channels * sizeof(int16_t) * 256 * 16;
sh->audio_in_minsize = DTSBUFFER_SIZE;
sh->samplesize=2;
return 1;
}
static int init(sh_audio_t *sh)
{
dts_state_t *s;
int flags;
int decoded_bytes;
s = dts_init(0);
if(s == NULL) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_init() failed\n");
return 0;
}
sh->context = s;
if(dts_sync(sh, &flags) < 0) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts sync failed\n");
dts_free(s);
return 0;
}
channels_info(flags);
assert(audio_output_channels >= 1 && audio_output_channels <= 6);
sh->channels = audio_output_channels;
decoded_bytes = decode_audio(sh, sh->a_buffer, 1, sh->a_buffer_size);
if(decoded_bytes > 0)
sh->a_buffer_len = decoded_bytes;
else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts decode failed on first frame (up/downmix problem?)\n");
dts_free(s);
return 0;
}
return 1;
}
static void uninit(sh_audio_t *sh)
{
dts_state_t *s = sh->context;
dts_free(s);
}
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
int flags;
switch(cmd){
case ADCTRL_RESYNC_STREAM:
dts_sync(sh, &flags);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
Reference in New Issue View Git Blame Copy Permalink