mirror of
https://github.com/mpv-player/mpv
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47ef593617
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@26759 b3059339-0415-0410-9bf9-f77b7e298cf2
344 lines
8.8 KiB
C
344 lines
8.8 KiB
C
/*
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* DTS Coherent Acoustics stream decoder using libdca
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* This file is partially based on dtsdec.c r9036 from FFmpeg and ad_liba52.c
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*
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* Copyright (C) 2007 Roberto Togni
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*
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* This file is part of MPlayer.
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*
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* MPlayer 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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* MPlayer 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 along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <assert.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "ad_internal.h"
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#include <dts.h>
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static ad_info_t info =
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{
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"DTS decoding with libdca",
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"libdca",
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"Roberto Togni",
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"",
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""
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};
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LIBAD_EXTERN(libdca)
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#define DTSBUFFER_SIZE 18726
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#define HEADER_SIZE 14
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#define CONVERT_LEVEL 1
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#define CONVERT_BIAS 0
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static const char ch2flags[6] = {
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DTS_MONO,
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DTS_STEREO,
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DTS_3F,
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DTS_2F2R,
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DTS_3F2R,
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DTS_3F2R | DTS_LFE
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};
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static inline int16_t convert(sample_t s)
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{
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int i = s * 0x7fff;
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return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
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}
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static void convert2s16_multi(sample_t *f, int16_t *s16, int flags, int ch_out)
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{
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int i;
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switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
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case DTS_MONO:
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if (ch_out == 1)
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for(i = 0; i < 256; i++)
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s16[i] = convert(f[i]);
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else
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for(i = 0; i < 256; i++){
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s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0;
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s16[5*i+4] = convert(f[i]);
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}
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break;
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case DTS_CHANNEL:
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case DTS_STEREO:
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case DTS_DOLBY:
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for(i = 0; i < 256; i++){
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s16[2*i] = convert(f[i]);
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s16[2*i+1] = convert(f[i+256]);
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}
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break;
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case DTS_3F:
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for(i = 0; i < 256; i++){
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s16[3*i] = convert(f[i+256]);
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s16[3*i+1] = convert(f[i+512]);
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s16[3*i+2] = convert(f[i]);
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}
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break;
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case DTS_2F2R:
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for(i = 0; i < 256; i++){
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s16[4*i] = convert(f[i]);
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s16[4*i+1] = convert(f[i+256]);
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s16[4*i+2] = convert(f[i+512]);
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s16[4*i+3] = convert(f[i+768]);
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}
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break;
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case DTS_3F2R:
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for(i = 0; i < 256; i++){
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s16[5*i] = convert(f[i+256]);
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s16[5*i+1] = convert(f[i+512]);
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s16[5*i+2] = convert(f[i+768]);
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s16[5*i+3] = convert(f[i+1024]);
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s16[5*i+4] = convert(f[i]);
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}
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break;
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case DTS_MONO | DTS_LFE:
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for(i = 0; i < 256; i++){
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s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0;
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s16[6*i+4] = convert(f[i]);
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s16[6*i+5] = convert(f[i+256]);
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}
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break;
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case DTS_CHANNEL | DTS_LFE:
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case DTS_STEREO | DTS_LFE:
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case DTS_DOLBY | DTS_LFE:
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for(i = 0; i < 256; i++){
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s16[6*i] = convert(f[i]);
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s16[6*i+1] = convert(f[i+256]);
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s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
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s16[6*i+5] = convert(f[i+512]);
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}
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break;
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case DTS_3F | DTS_LFE:
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for(i = 0; i < 256; i++){
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s16[6*i] = convert(f[i+256]);
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s16[6*i+1] = convert(f[i+512]);
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s16[6*i+2] = s16[6*i+3] = 0;
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s16[6*i+4] = convert(f[i]);
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s16[6*i+5] = convert(f[i+768]);
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}
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break;
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case DTS_2F2R | DTS_LFE:
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for(i = 0; i < 256; i++){
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s16[6*i] = convert(f[i]);
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s16[6*i+1] = convert(f[i+256]);
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s16[6*i+2] = convert(f[i+512]);
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s16[6*i+3] = convert(f[i+768]);
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s16[6*i+4] = 0;
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s16[6*i+5] = convert(f[1024]);
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}
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break;
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case DTS_3F2R | DTS_LFE:
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for(i = 0; i < 256; i++){
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s16[6*i] = convert(f[i+256]);
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s16[6*i+1] = convert(f[i+512]);
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s16[6*i+2] = convert(f[i+768]);
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s16[6*i+3] = convert(f[i+1024]);
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s16[6*i+4] = convert(f[i]);
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s16[6*i+5] = convert(f[i+1280]);
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}
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break;
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}
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}
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static void channels_info(int flags)
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{
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int lfe = 0;
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char lfestr[5] = "";
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if (flags & DTS_LFE) {
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lfe = 1;
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strcpy(lfestr, "+lfe");
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}
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mp_msg(MSGT_DECAUDIO, MSGL_V, "DTS: ");
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switch(flags & DTS_CHANNEL_MASK){
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case DTS_MONO:
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mp_msg(MSGT_DECAUDIO, MSGL_V, "1.%d (mono%s)", lfe, lfestr);
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break;
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case DTS_CHANNEL:
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mp_msg(MSGT_DECAUDIO, MSGL_V, "2.%d (channel%s)", lfe, lfestr);
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break;
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case DTS_STEREO:
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mp_msg(MSGT_DECAUDIO, MSGL_V, "2.%d (stereo%s)", lfe, lfestr);
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break;
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case DTS_3F:
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mp_msg(MSGT_DECAUDIO, MSGL_V, "3.%d (3f%s)", lfe, lfestr);
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break;
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case DTS_2F2R:
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mp_msg(MSGT_DECAUDIO, MSGL_V, "4.%d (2f+2r%s)", lfe, lfestr);
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break;
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case DTS_3F2R:
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mp_msg(MSGT_DECAUDIO, MSGL_V, "5.%d (3f+2r%s)", lfe, lfestr);
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break;
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default:
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mp_msg(MSGT_DECAUDIO, MSGL_V, "x.%d (unknown%s)", lfe, lfestr);
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}
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mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
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}
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static int dts_sync(sh_audio_t *sh, int *flags)
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{
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dts_state_t *s = sh->context;
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int length;
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int sample_rate;
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int frame_length;
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int bit_rate;
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sh->a_in_buffer_len=0;
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while(1) {
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while(sh->a_in_buffer_len < HEADER_SIZE) {
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int c = demux_getc(sh->ds);
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if(c < 0)
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return -1;
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sh->a_in_buffer[sh->a_in_buffer_len++] = c;
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}
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length = dts_syncinfo(s, sh->a_in_buffer, flags, &sample_rate,
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&bit_rate, &frame_length);
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if(length >= HEADER_SIZE)
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break;
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// mp_msg(MSGT_DECAUDIO, MSGL_V, "skip\n");
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memmove(sh->a_in_buffer, sh->a_in_buffer+1, HEADER_SIZE-1);
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--sh->a_in_buffer_len;
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}
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demux_read_data(sh->ds, sh->a_in_buffer + HEADER_SIZE, length - HEADER_SIZE);
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sh->samplerate = sample_rate;
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sh->i_bps = bit_rate/8;
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return length;
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}
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static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen, int maxlen)
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{
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dts_state_t *s = sh->context;
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int16_t *out_samples = (int16_t*)buf;
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int flags;
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level_t level;
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sample_t bias;
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int nblocks;
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int i;
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int data_size = 0;
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if(!sh->a_in_buffer_len)
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if(dts_sync(sh, &flags) < 0) return -1; /* EOF */
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sh->a_in_buffer_len=0;
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flags &= ~(DTS_CHANNEL_MASK | DTS_LFE);
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flags |= ch2flags[sh->channels - 1];
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level = CONVERT_LEVEL;
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bias = CONVERT_BIAS;
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flags |= DTS_ADJUST_LEVEL;
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if(dts_frame(s, sh->a_in_buffer, &flags, &level, bias)) {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_frame() failed\n");
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goto end;
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}
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nblocks = dts_blocks_num(s);
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for(i = 0; i < nblocks; i++) {
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if(dts_block(s)) {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_block() failed\n");
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goto end;
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}
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convert2s16_multi(dts_samples(s), out_samples, flags, sh->channels);
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out_samples += 256 * sh->channels;
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data_size += 256 * sizeof(int16_t) * sh->channels;
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}
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end:
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return data_size;
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}
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static int preinit(sh_audio_t *sh)
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{
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/* 256 = samples per block, 16 = max number of blocks */
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sh->audio_out_minsize = audio_output_channels * sizeof(int16_t) * 256 * 16;
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sh->audio_in_minsize = DTSBUFFER_SIZE;
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sh->samplesize=2;
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return 1;
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}
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static int init(sh_audio_t *sh)
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{
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dts_state_t *s;
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int flags;
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int decoded_bytes;
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s = dts_init(0);
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if(s == NULL) {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_init() failed\n");
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return 0;
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}
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sh->context = s;
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if(dts_sync(sh, &flags) < 0) {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts sync failed\n");
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dts_free(s);
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return 0;
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}
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channels_info(flags);
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assert(audio_output_channels >= 1 && audio_output_channels <= 6);
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sh->channels = audio_output_channels;
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decoded_bytes = decode_audio(sh, sh->a_buffer, 1, sh->a_buffer_size);
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if(decoded_bytes > 0)
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sh->a_buffer_len = decoded_bytes;
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else {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts decode failed on first frame (up/downmix problem?)\n");
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dts_free(s);
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return 0;
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}
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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{
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dts_state_t *s = sh->context;
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dts_free(s);
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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{
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int flags;
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switch(cmd){
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case ADCTRL_RESYNC_STREAM:
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dts_sync(sh, &flags);
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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