# HG changeset patch # User Matti Hamalainen # Date 1375821988 -10800 # Node ID 7d2857f170e210176a2f209fbdab6b8f9b5d2e7f # Parent 9c63b355c82b6b33ceb9854d8f3554ecd4687ff4 Remove the audio converter. diff -r 9c63b355c82b -r 7d2857f170e2 liboggplayer-src/src/SDL_audiocvt.cpp --- a/liboggplayer-src/src/SDL_audiocvt.cpp Tue Aug 06 23:45:05 2013 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,1509 +0,0 @@ -/* - This is a modified version of SDL_audiocvt.c from SDL 1.2.13 - It does not depend on any other files from SDL -*/ -/* - SDL - Simple DirectMedia Layer - Copyright (C) 1997-2006 Sam Lantinga - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU Lesser General Public - License as published by the Free Software Foundation; either - version 2.1 of the License, or (at your option) any later version. - - This library 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 - Lesser General Public License for more details. - - You should have received a copy of the GNU Lesser General Public - License along with this library; if not, write to the Free Software - Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - - Sam Lantinga - slouken@libsdl.org -*/ - -#include "SDL_audiocvt.hpp" - -/* Effectively mix right and left channels into a single channel */ -void SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Sint32 sample; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to mono\n"); -#endif - switch (format&0x8018) { - - case AUDIO_U8: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - for ( i=cvt->len_cvt/2; i; --i ) { - sample = src[0] + src[1]; - *dst = (Uint8)(sample / 2); - src += 2; - dst += 1; - } - } - break; - - case AUDIO_S8: { - Sint8 *src, *dst; - - src = (Sint8 *)cvt->buf; - dst = (Sint8 *)cvt->buf; - for ( i=cvt->len_cvt/2; i; --i ) { - sample = src[0] + src[1]; - *dst = (Sint8)(sample / 2); - src += 2; - dst += 1; - } - } - break; - - case AUDIO_U16: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/4; i; --i ) { - sample = (Uint16)((src[0]<<8)|src[1])+ - (Uint16)((src[2]<<8)|src[3]); - sample /= 2; - dst[1] = (sample&0xFF); - sample >>= 8; - dst[0] = (sample&0xFF); - src += 4; - dst += 2; - } - } else { - for ( i=cvt->len_cvt/4; i; --i ) { - sample = (Uint16)((src[1]<<8)|src[0])+ - (Uint16)((src[3]<<8)|src[2]); - sample /= 2; - dst[0] = (sample&0xFF); - sample >>= 8; - dst[1] = (sample&0xFF); - src += 4; - dst += 2; - } - } - } - break; - - case AUDIO_S16: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/4; i; --i ) { - sample = (Sint16)((src[0]<<8)|src[1])+ - (Sint16)((src[2]<<8)|src[3]); - sample /= 2; - dst[1] = (sample&0xFF); - sample >>= 8; - dst[0] = (sample&0xFF); - src += 4; - dst += 2; - } - } else { - for ( i=cvt->len_cvt/4; i; --i ) { - sample = (Sint16)((src[1]<<8)|src[0])+ - (Sint16)((src[3]<<8)|src[2]); - sample /= 2; - dst[0] = (sample&0xFF); - sample >>= 8; - dst[1] = (sample&0xFF); - src += 4; - dst += 2; - } - } - } - break; - } - cvt->len_cvt /= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Discard top 4 channels */ -void SDL_ConvertStrip(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Sint32 lsample, rsample; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting down to stereo\n"); -#endif - switch (format&0x8018) { - - case AUDIO_U8: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - for ( i=cvt->len_cvt/6; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 6; - dst += 2; - } - } - break; - - case AUDIO_S8: { - Sint8 *src, *dst; - - src = (Sint8 *)cvt->buf; - dst = (Sint8 *)cvt->buf; - for ( i=cvt->len_cvt/6; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 6; - dst += 2; - } - } - break; - - case AUDIO_U16: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/12; i; --i ) { - lsample = (Uint16)((src[0]<<8)|src[1]); - rsample = (Uint16)((src[2]<<8)|src[3]); - dst[1] = (lsample&0xFF); - lsample >>= 8; - dst[0] = (lsample&0xFF); - dst[3] = (rsample&0xFF); - rsample >>= 8; - dst[2] = (rsample&0xFF); - src += 12; - dst += 4; - } - } else { - for ( i=cvt->len_cvt/12; i; --i ) { - lsample = (Uint16)((src[1]<<8)|src[0]); - rsample = (Uint16)((src[3]<<8)|src[2]); - dst[0] = (lsample&0xFF); - lsample >>= 8; - dst[1] = (lsample&0xFF); - dst[2] = (rsample&0xFF); - rsample >>= 8; - dst[3] = (rsample&0xFF); - src += 12; - dst += 4; - } - } - } - break; - - case AUDIO_S16: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/12; i; --i ) { - lsample = (Sint16)((src[0]<<8)|src[1]); - rsample = (Sint16)((src[2]<<8)|src[3]); - dst[1] = (lsample&0xFF); - lsample >>= 8; - dst[0] = (lsample&0xFF); - dst[3] = (rsample&0xFF); - rsample >>= 8; - dst[2] = (rsample&0xFF); - src += 12; - dst += 4; - } - } else { - for ( i=cvt->len_cvt/12; i; --i ) { - lsample = (Sint16)((src[1]<<8)|src[0]); - rsample = (Sint16)((src[3]<<8)|src[2]); - dst[0] = (lsample&0xFF); - lsample >>= 8; - dst[1] = (lsample&0xFF); - dst[2] = (rsample&0xFF); - rsample >>= 8; - dst[3] = (rsample&0xFF); - src += 12; - dst += 4; - } - } - } - break; - } - cvt->len_cvt /= 3; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Discard top 2 channels of 6 */ -void SDL_ConvertStrip_2(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Sint32 lsample, rsample; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting 6 down to quad\n"); -#endif - switch (format&0x8018) { - - case AUDIO_U8: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - for ( i=cvt->len_cvt/4; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } - } - break; - - case AUDIO_S8: { - Sint8 *src, *dst; - - src = (Sint8 *)cvt->buf; - dst = (Sint8 *)cvt->buf; - for ( i=cvt->len_cvt/4; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } - } - break; - - case AUDIO_U16: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/8; i; --i ) { - lsample = (Uint16)((src[0]<<8)|src[1]); - rsample = (Uint16)((src[2]<<8)|src[3]); - dst[1] = (lsample&0xFF); - lsample >>= 8; - dst[0] = (lsample&0xFF); - dst[3] = (rsample&0xFF); - rsample >>= 8; - dst[2] = (rsample&0xFF); - src += 8; - dst += 4; - } - } else { - for ( i=cvt->len_cvt/8; i; --i ) { - lsample = (Uint16)((src[1]<<8)|src[0]); - rsample = (Uint16)((src[3]<<8)|src[2]); - dst[0] = (lsample&0xFF); - lsample >>= 8; - dst[1] = (lsample&0xFF); - dst[2] = (rsample&0xFF); - rsample >>= 8; - dst[3] = (rsample&0xFF); - src += 8; - dst += 4; - } - } - } - break; - - case AUDIO_S16: { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/8; i; --i ) { - lsample = (Sint16)((src[0]<<8)|src[1]); - rsample = (Sint16)((src[2]<<8)|src[3]); - dst[1] = (lsample&0xFF); - lsample >>= 8; - dst[0] = (lsample&0xFF); - dst[3] = (rsample&0xFF); - rsample >>= 8; - dst[2] = (rsample&0xFF); - src += 8; - dst += 4; - } - } else { - for ( i=cvt->len_cvt/8; i; --i ) { - lsample = (Sint16)((src[1]<<8)|src[0]); - rsample = (Sint16)((src[3]<<8)|src[2]); - dst[0] = (lsample&0xFF); - lsample >>= 8; - dst[1] = (lsample&0xFF); - dst[2] = (rsample&0xFF); - rsample >>= 8; - dst[3] = (rsample&0xFF); - src += 8; - dst += 4; - } - } - } - break; - } - cvt->len_cvt /= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Duplicate a mono channel to both stereo channels */ -void SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to stereo\n"); -#endif - if ( (format & 0xFF) == 16 ) { - Uint16 *src, *dst; - - src = (Uint16 *)(cvt->buf+cvt->len_cvt); - dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2); - for ( i=cvt->len_cvt/2; i; --i ) { - dst -= 2; - src -= 1; - dst[0] = src[0]; - dst[1] = src[0]; - } - } else { - Uint8 *src, *dst; - - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - for ( i=cvt->len_cvt; i; --i ) { - dst -= 2; - src -= 1; - dst[0] = src[0]; - dst[1] = src[0]; - } - } - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Duplicate a stereo channel to a pseudo-5.1 stream */ -void SDL_ConvertSurround(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting stereo to surround\n"); -#endif - switch (format&0x8018) { - - case AUDIO_U8: { - Uint8 *src, *dst, lf, rf, ce; - - src = (Uint8 *)(cvt->buf+cvt->len_cvt); - dst = (Uint8 *)(cvt->buf+cvt->len_cvt*3); - for ( i=cvt->len_cvt; i; --i ) { - dst -= 6; - src -= 2; - lf = src[0]; - rf = src[1]; - ce = (lf/2) + (rf/2); - dst[0] = lf; - dst[1] = rf; - dst[2] = lf - ce; - dst[3] = rf - ce; - dst[4] = ce; - dst[5] = ce; - } - } - break; - - case AUDIO_S8: { - Sint8 *src, *dst, lf, rf, ce; - - src = (Sint8 *)cvt->buf+cvt->len_cvt; - dst = (Sint8 *)cvt->buf+cvt->len_cvt*3; - for ( i=cvt->len_cvt; i; --i ) { - dst -= 6; - src -= 2; - lf = src[0]; - rf = src[1]; - ce = (lf/2) + (rf/2); - dst[0] = lf; - dst[1] = rf; - dst[2] = lf - ce; - dst[3] = rf - ce; - dst[4] = ce; - dst[5] = ce; - } - } - break; - - case AUDIO_U16: { - Uint8 *src, *dst; - Uint16 lf, rf, ce, lr, rr; - - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*3; - - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 12; - src -= 4; - lf = (Uint16)((src[0]<<8)|src[1]); - rf = (Uint16)((src[2]<<8)|src[3]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[1] = (lf&0xFF); - dst[0] = ((lf>>8)&0xFF); - dst[3] = (rf&0xFF); - dst[2] = ((rf>>8)&0xFF); - - dst[1+4] = (lr&0xFF); - dst[0+4] = ((lr>>8)&0xFF); - dst[3+4] = (rr&0xFF); - dst[2+4] = ((rr>>8)&0xFF); - - dst[1+8] = (ce&0xFF); - dst[0+8] = ((ce>>8)&0xFF); - dst[3+8] = (ce&0xFF); - dst[2+8] = ((ce>>8)&0xFF); - } - } else { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 12; - src -= 4; - lf = (Uint16)((src[1]<<8)|src[0]); - rf = (Uint16)((src[3]<<8)|src[2]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[0] = (lf&0xFF); - dst[1] = ((lf>>8)&0xFF); - dst[2] = (rf&0xFF); - dst[3] = ((rf>>8)&0xFF); - - dst[0+4] = (lr&0xFF); - dst[1+4] = ((lr>>8)&0xFF); - dst[2+4] = (rr&0xFF); - dst[3+4] = ((rr>>8)&0xFF); - - dst[0+8] = (ce&0xFF); - dst[1+8] = ((ce>>8)&0xFF); - dst[2+8] = (ce&0xFF); - dst[3+8] = ((ce>>8)&0xFF); - } - } - } - break; - - case AUDIO_S16: { - Uint8 *src, *dst; - Sint16 lf, rf, ce, lr, rr; - - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*3; - - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 12; - src -= 4; - lf = (Sint16)((src[0]<<8)|src[1]); - rf = (Sint16)((src[2]<<8)|src[3]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[1] = (lf&0xFF); - dst[0] = ((lf>>8)&0xFF); - dst[3] = (rf&0xFF); - dst[2] = ((rf>>8)&0xFF); - - dst[1+4] = (lr&0xFF); - dst[0+4] = ((lr>>8)&0xFF); - dst[3+4] = (rr&0xFF); - dst[2+4] = ((rr>>8)&0xFF); - - dst[1+8] = (ce&0xFF); - dst[0+8] = ((ce>>8)&0xFF); - dst[3+8] = (ce&0xFF); - dst[2+8] = ((ce>>8)&0xFF); - } - } else { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 12; - src -= 4; - lf = (Sint16)((src[1]<<8)|src[0]); - rf = (Sint16)((src[3]<<8)|src[2]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[0] = (lf&0xFF); - dst[1] = ((lf>>8)&0xFF); - dst[2] = (rf&0xFF); - dst[3] = ((rf>>8)&0xFF); - - dst[0+4] = (lr&0xFF); - dst[1+4] = ((lr>>8)&0xFF); - dst[2+4] = (rr&0xFF); - dst[3+4] = ((rr>>8)&0xFF); - - dst[0+8] = (ce&0xFF); - dst[1+8] = ((ce>>8)&0xFF); - dst[2+8] = (ce&0xFF); - dst[3+8] = ((ce>>8)&0xFF); - } - } - } - break; - } - cvt->len_cvt *= 3; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Duplicate a stereo channel to a pseudo-4.0 stream */ -void SDL_ConvertSurround_4(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting stereo to quad\n"); -#endif - switch (format&0x8018) { - - case AUDIO_U8: { - Uint8 *src, *dst, lf, rf, ce; - - src = (Uint8 *)(cvt->buf+cvt->len_cvt); - dst = (Uint8 *)(cvt->buf+cvt->len_cvt*2); - for ( i=cvt->len_cvt; i; --i ) { - dst -= 4; - src -= 2; - lf = src[0]; - rf = src[1]; - ce = (lf/2) + (rf/2); - dst[0] = lf; - dst[1] = rf; - dst[2] = lf - ce; - dst[3] = rf - ce; - } - } - break; - - case AUDIO_S8: { - Sint8 *src, *dst, lf, rf, ce; - - src = (Sint8 *)cvt->buf+cvt->len_cvt; - dst = (Sint8 *)cvt->buf+cvt->len_cvt*2; - for ( i=cvt->len_cvt; i; --i ) { - dst -= 4; - src -= 2; - lf = src[0]; - rf = src[1]; - ce = (lf/2) + (rf/2); - dst[0] = lf; - dst[1] = rf; - dst[2] = lf - ce; - dst[3] = rf - ce; - } - } - break; - - case AUDIO_U16: { - Uint8 *src, *dst; - Uint16 lf, rf, ce, lr, rr; - - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 8; - src -= 4; - lf = (Uint16)((src[0]<<8)|src[1]); - rf = (Uint16)((src[2]<<8)|src[3]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[1] = (lf&0xFF); - dst[0] = ((lf>>8)&0xFF); - dst[3] = (rf&0xFF); - dst[2] = ((rf>>8)&0xFF); - - dst[1+4] = (lr&0xFF); - dst[0+4] = ((lr>>8)&0xFF); - dst[3+4] = (rr&0xFF); - dst[2+4] = ((rr>>8)&0xFF); - } - } else { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 8; - src -= 4; - lf = (Uint16)((src[1]<<8)|src[0]); - rf = (Uint16)((src[3]<<8)|src[2]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[0] = (lf&0xFF); - dst[1] = ((lf>>8)&0xFF); - dst[2] = (rf&0xFF); - dst[3] = ((rf>>8)&0xFF); - - dst[0+4] = (lr&0xFF); - dst[1+4] = ((lr>>8)&0xFF); - dst[2+4] = (rr&0xFF); - dst[3+4] = ((rr>>8)&0xFF); - } - } - } - break; - - case AUDIO_S16: { - Uint8 *src, *dst; - Sint16 lf, rf, ce, lr, rr; - - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - - if ( (format & 0x1000) == 0x1000 ) { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 8; - src -= 4; - lf = (Sint16)((src[0]<<8)|src[1]); - rf = (Sint16)((src[2]<<8)|src[3]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[1] = (lf&0xFF); - dst[0] = ((lf>>8)&0xFF); - dst[3] = (rf&0xFF); - dst[2] = ((rf>>8)&0xFF); - - dst[1+4] = (lr&0xFF); - dst[0+4] = ((lr>>8)&0xFF); - dst[3+4] = (rr&0xFF); - dst[2+4] = ((rr>>8)&0xFF); - } - } else { - for ( i=cvt->len_cvt/4; i; --i ) { - dst -= 8; - src -= 4; - lf = (Sint16)((src[1]<<8)|src[0]); - rf = (Sint16)((src[3]<<8)|src[2]); - ce = (lf/2) + (rf/2); - rr = lf - ce; - lr = rf - ce; - dst[0] = (lf&0xFF); - dst[1] = ((lf>>8)&0xFF); - dst[2] = (rf&0xFF); - dst[3] = ((rf>>8)&0xFF); - - dst[0+4] = (lr&0xFF); - dst[1+4] = ((lr>>8)&0xFF); - dst[2+4] = (rr&0xFF); - dst[3+4] = ((rr>>8)&0xFF); - } - } - } - break; - } - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Convert 8-bit to 16-bit - LSB */ -void SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to 16-bit LSB\n"); -#endif - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - for ( i=cvt->len_cvt; i; --i ) { - src -= 1; - dst -= 2; - dst[1] = *src; - dst[0] = 0; - } - format = ((format & ~0x0008) | AUDIO_U16LSB); - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} -/* Convert 8-bit to 16-bit - MSB */ -void SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to 16-bit MSB\n"); -#endif - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - for ( i=cvt->len_cvt; i; --i ) { - src -= 1; - dst -= 2; - dst[0] = *src; - dst[1] = 0; - } - format = ((format & ~0x0008) | AUDIO_U16MSB); - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Convert 16-bit to 8-bit */ -void SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to 8-bit\n"); -#endif - src = cvt->buf; - dst = cvt->buf; - if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ - ++src; - } - for ( i=cvt->len_cvt/2; i; --i ) { - *dst = *src; - src += 2; - dst += 1; - } - format = ((format & ~0x9010) | AUDIO_U8); - cvt->len_cvt /= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Toggle signed/unsigned */ -void SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *data; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio signedness\n"); -#endif - data = cvt->buf; - if ( (format & 0xFF) == 16 ) { - if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ - ++data; - } - for ( i=cvt->len_cvt/2; i; --i ) { - *data ^= 0x80; - data += 2; - } - } else { - for ( i=cvt->len_cvt; i; --i ) { - *data++ ^= 0x80; - } - } - format = (format ^ 0x8000); - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Toggle endianness */ -void SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *data, tmp; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio endianness\n"); -#endif - data = cvt->buf; - for ( i=cvt->len_cvt/2; i; --i ) { - tmp = data[0]; - data[0] = data[1]; - data[1] = tmp; - data += 2; - } - format = (format ^ 0x1000); - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Convert rate up by multiple of 2 */ -void SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); -#endif - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt; i; --i ) { - src -= 1; - dst -= 2; - dst[0] = src[0]; - dst[1] = src[0]; - } - break; - case 16: - for ( i=cvt->len_cvt/2; i; --i ) { - src -= 2; - dst -= 4; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[0]; - dst[3] = src[1]; - } - break; - } - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Convert rate up by multiple of 2, for stereo */ -void SDL_RateMUL2_c2(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); -#endif - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt/2; i; --i ) { - src -= 2; - dst -= 4; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[0]; - dst[3] = src[1]; - } - break; - case 16: - for ( i=cvt->len_cvt/4; i; --i ) { - src -= 4; - dst -= 8; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[0]; - dst[5] = src[1]; - dst[6] = src[2]; - dst[7] = src[3]; - } - break; - } - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Convert rate up by multiple of 2, for quad */ -void SDL_RateMUL2_c4(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); -#endif - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt/4; i; --i ) { - src -= 4; - dst -= 8; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[0]; - dst[5] = src[1]; - dst[6] = src[2]; - dst[7] = src[3]; - } - break; - case 16: - for ( i=cvt->len_cvt/8; i; --i ) { - src -= 8; - dst -= 16; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - dst[8] = src[0]; - dst[9] = src[1]; - dst[10] = src[2]; - dst[11] = src[3]; - dst[12] = src[4]; - dst[13] = src[5]; - dst[14] = src[6]; - dst[15] = src[7]; - } - break; - } - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Convert rate up by multiple of 2, for 5.1 */ -void SDL_RateMUL2_c6(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); -#endif - src = cvt->buf+cvt->len_cvt; - dst = cvt->buf+cvt->len_cvt*2; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt/6; i; --i ) { - src -= 6; - dst -= 12; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[0]; - dst[7] = src[1]; - dst[8] = src[2]; - dst[9] = src[3]; - dst[10] = src[4]; - dst[11] = src[5]; - } - break; - case 16: - for ( i=cvt->len_cvt/12; i; --i ) { - src -= 12; - dst -= 24; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - dst[8] = src[8]; - dst[9] = src[9]; - dst[10] = src[10]; - dst[11] = src[11]; - dst[12] = src[0]; - dst[13] = src[1]; - dst[14] = src[2]; - dst[15] = src[3]; - dst[16] = src[4]; - dst[17] = src[5]; - dst[18] = src[6]; - dst[19] = src[7]; - dst[20] = src[8]; - dst[21] = src[9]; - dst[22] = src[10]; - dst[23] = src[11]; - } - break; - } - cvt->len_cvt *= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Convert rate down by multiple of 2 */ -void SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); -#endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt/2; i; --i ) { - dst[0] = src[0]; - src += 2; - dst += 1; - } - break; - case 16: - for ( i=cvt->len_cvt/4; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } - break; - } - cvt->len_cvt /= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Convert rate down by multiple of 2, for stereo */ -void SDL_RateDIV2_c2(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); -#endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt/4; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } - break; - case 16: - for ( i=cvt->len_cvt/8; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - src += 8; - dst += 4; - } - break; - } - cvt->len_cvt /= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - - -/* Convert rate down by multiple of 2, for quad */ -void SDL_RateDIV2_c4(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); -#endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt/8; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - src += 8; - dst += 4; - } - break; - case 16: - for ( i=cvt->len_cvt/16; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - src += 16; - dst += 8; - } - break; - } - cvt->len_cvt /= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Convert rate down by multiple of 2, for 5.1 */ -void SDL_RateDIV2_c6(SDL_AudioCVT *cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); -#endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { - case 8: - for ( i=cvt->len_cvt/12; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - src += 12; - dst += 6; - } - break; - case 16: - for ( i=cvt->len_cvt/24; i; --i ) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - dst[8] = src[8]; - dst[9] = src[9]; - dst[10] = src[10]; - dst[11] = src[11]; - src += 24; - dst += 12; - } - break; - } - cvt->len_cvt /= 2; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* Very slow rate conversion routine */ -void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format) -{ - double ipos; - int i, clen; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr); -#endif - clen = (int)((double)cvt->len_cvt / cvt->rate_incr); - if ( cvt->rate_incr > 1.0 ) { - switch (format & 0xFF) { - case 8: { - Uint8 *output; - - output = cvt->buf; - ipos = 0.0; - for ( i=clen; i; --i ) { - *output = cvt->buf[(int)ipos]; - ipos += cvt->rate_incr; - output += 1; - } - } - break; - - case 16: { - Uint16 *output; - - clen &= ~1; - output = (Uint16 *)cvt->buf; - ipos = 0.0; - for ( i=clen/2; i; --i ) { - *output=((Uint16 *)cvt->buf)[(int)ipos]; - ipos += cvt->rate_incr; - output += 1; - } - } - break; - } - } else { - switch (format & 0xFF) { - case 8: { - Uint8 *output; - - output = cvt->buf+clen; - ipos = (double)cvt->len_cvt; - for ( i=clen; i; --i ) { - ipos -= cvt->rate_incr; - output -= 1; - *output = cvt->buf[(int)ipos]; - } - } - break; - - case 16: { - Uint16 *output; - - clen &= ~1; - output = (Uint16 *)(cvt->buf+clen); - ipos = (double)cvt->len_cvt/2; - for ( i=clen/2; i; --i ) { - ipos -= cvt->rate_incr; - output -= 1; - *output=((Uint16 *)cvt->buf)[(int)ipos]; - } - } - break; - } - } - cvt->len_cvt = clen; - if ( cvt->filters[++cvt->filter_index] ) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -int SDL_ConvertAudio(SDL_AudioCVT *cvt) -{ - /* Make sure there's data to convert */ - if ( cvt->buf == NULL ) { - return(-1); - } - /* Return okay if no conversion is necessary */ - cvt->len_cvt = cvt->len; - if ( cvt->filters[0] == NULL ) { - return(0); - } - - /* Set up the conversion and go! */ - cvt->filter_index = 0; - cvt->filters[0](cvt, cvt->src_format); - return(0); -} - -/* Creates a set of audio filters to convert from one format to another. - Returns -1 if the format conversion is not supported, or 1 if the - audio filter is set up. -*/ - -int SDL_BuildAudioCVT(SDL_AudioCVT *cvt, - Uint16 src_format, Uint8 src_channels, int src_rate, - Uint16 dst_format, Uint8 dst_channels, int dst_rate) -{ -/*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n", - src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/ - /* Start off with no conversion necessary */ - cvt->needed = 0; - cvt->filter_index = 0; - cvt->filters[0] = NULL; - cvt->len_mult = 1; - cvt->len_ratio = 1.0; - - /* First filter: Endian conversion from src to dst */ - if ( (src_format & 0x1000) != (dst_format & 0x1000) - && ((src_format & 0xff) == 16) && ((dst_format & 0xff) == 16)) { - cvt->filters[cvt->filter_index++] = SDL_ConvertEndian; - } - - /* Second filter: Sign conversion -- signed/unsigned */ - if ( (src_format & 0x8000) != (dst_format & 0x8000) ) { - cvt->filters[cvt->filter_index++] = SDL_ConvertSign; - } - - /* Next filter: Convert 16 bit <--> 8 bit PCM */ - if ( (src_format & 0xFF) != (dst_format & 0xFF) ) { - switch (dst_format&0x10FF) { - case AUDIO_U8: - cvt->filters[cvt->filter_index++] = - SDL_Convert8; - cvt->len_ratio /= 2; - break; - case AUDIO_U16LSB: - cvt->filters[cvt->filter_index++] = - SDL_Convert16LSB; - cvt->len_mult *= 2; - cvt->len_ratio *= 2; - break; - case AUDIO_U16MSB: - cvt->filters[cvt->filter_index++] = - SDL_Convert16MSB; - cvt->len_mult *= 2; - cvt->len_ratio *= 2; - break; - } - } - - /* Last filter: Mono/Stereo conversion */ - if ( src_channels != dst_channels ) { - if ( (src_channels == 1) && (dst_channels > 1) ) { - cvt->filters[cvt->filter_index++] = - SDL_ConvertStereo; - cvt->len_mult *= 2; - src_channels = 2; - cvt->len_ratio *= 2; - } - if ( (src_channels == 2) && - (dst_channels == 6) ) { - cvt->filters[cvt->filter_index++] = - SDL_ConvertSurround; - src_channels = 6; - cvt->len_mult *= 3; - cvt->len_ratio *= 3; - } - if ( (src_channels == 2) && - (dst_channels == 4) ) { - cvt->filters[cvt->filter_index++] = - SDL_ConvertSurround_4; - src_channels = 4; - cvt->len_mult *= 2; - cvt->len_ratio *= 2; - } - while ( (src_channels*2) <= dst_channels ) { - cvt->filters[cvt->filter_index++] = - SDL_ConvertStereo; - cvt->len_mult *= 2; - src_channels *= 2; - cvt->len_ratio *= 2; - } - if ( (src_channels == 6) && - (dst_channels <= 2) ) { - cvt->filters[cvt->filter_index++] = - SDL_ConvertStrip; - src_channels = 2; - cvt->len_ratio /= 3; - } - if ( (src_channels == 6) && - (dst_channels == 4) ) { - cvt->filters[cvt->filter_index++] = - SDL_ConvertStrip_2; - src_channels = 4; - cvt->len_ratio /= 2; - } - /* This assumes that 4 channel audio is in the format: - Left {front/back} + Right {front/back} - so converting to L/R stereo works properly. - */ - while ( ((src_channels%2) == 0) && - ((src_channels/2) >= dst_channels) ) { - cvt->filters[cvt->filter_index++] = - SDL_ConvertMono; - src_channels /= 2; - cvt->len_ratio /= 2; - } - if ( src_channels != dst_channels ) { - /* Uh oh.. */; - } - } - - /* Do rate conversion */ - cvt->rate_incr = 0.0; - if ( (src_rate/100) != (dst_rate/100) ) { - Uint32 hi_rate, lo_rate; - int len_mult; - double len_ratio; - void ( *rate_cvt)(SDL_AudioCVT *cvt, Uint16 format); - - if ( src_rate > dst_rate ) { - hi_rate = src_rate; - lo_rate = dst_rate; - switch (src_channels) { - case 1: rate_cvt = SDL_RateDIV2; break; - case 2: rate_cvt = SDL_RateDIV2_c2; break; - case 4: rate_cvt = SDL_RateDIV2_c4; break; - case 6: rate_cvt = SDL_RateDIV2_c6; break; - default: return -1; - } - len_mult = 1; - len_ratio = 0.5; - } else { - hi_rate = dst_rate; - lo_rate = src_rate; - switch (src_channels) { - case 1: rate_cvt = SDL_RateMUL2; break; - case 2: rate_cvt = SDL_RateMUL2_c2; break; - case 4: rate_cvt = SDL_RateMUL2_c4; break; - case 6: rate_cvt = SDL_RateMUL2_c6; break; - default: return -1; - } - len_mult = 2; - len_ratio = 2.0; - } - /* If hi_rate = lo_rate*2^x then conversion is easy */ - while ( ((lo_rate*2)/100) <= (hi_rate/100) ) { - cvt->filters[cvt->filter_index++] = rate_cvt; - cvt->len_mult *= len_mult; - lo_rate *= 2; - cvt->len_ratio *= len_ratio; - } - /* We may need a slow conversion here to finish up */ - if ( (lo_rate/100) != (hi_rate/100) ) { -#if 1 - /* The problem with this is that if the input buffer is - say 1K, and the conversion rate is say 1.1, then the - output buffer is 1.1K, which may not be an acceptable - buffer size for the audio driver (not a power of 2) - */ - /* For now, punt and hope the rate distortion isn't great. - */ -#else - if ( src_rate < dst_rate ) { - cvt->rate_incr = (double)lo_rate/hi_rate; - cvt->len_mult *= 2; - cvt->len_ratio /= cvt->rate_incr; - } else { - cvt->rate_incr = (double)hi_rate/lo_rate; - cvt->len_ratio *= cvt->rate_incr; - } - cvt->filters[cvt->filter_index++] = SDL_RateSLOW; -#endif - } - } - - /* Set up the filter information */ - if ( cvt->filter_index != 0 ) { - cvt->needed = 1; - cvt->src_format = src_format; - cvt->dst_format = dst_format; - cvt->len = 0; - cvt->buf = NULL; - cvt->filters[cvt->filter_index] = NULL; - } - return(cvt->needed); -} diff -r 9c63b355c82b -r 7d2857f170e2 liboggplayer-src/src/SDL_audiocvt.hpp --- a/liboggplayer-src/src/SDL_audiocvt.hpp Tue Aug 06 23:45:05 2013 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,49 +0,0 @@ -#pragma once - -#ifndef _WINDOWS - #include -#else - typedef signed __int8 int8_t; - typedef unsigned __int8 uint8_t; - typedef signed __int16 int16_t; - typedef unsigned __int16 uint16_t; - typedef signed __int32 int32_t; - typedef unsigned __int32 uint32_t; -#endif - -#include -/* Audio format flags (defaults to LSB byte order) */ -#define AUDIO_U8 0x0008 /* Unsigned 8-bit samples */ -#define AUDIO_S8 0x8008 /* Signed 8-bit samples */ -#define AUDIO_U16LSB 0x0010 /* Unsigned 16-bit samples */ -#define AUDIO_S16LSB 0x8010 /* Signed 16-bit samples */ -#define AUDIO_U16MSB 0x1010 /* As above, but big-endian byte order */ -#define AUDIO_S16MSB 0x9010 /* As above, but big-endian byte order */ -#define AUDIO_U16 AUDIO_U16LSB -#define AUDIO_S16 AUDIO_S16LSB - -typedef int8_t Sint8; -typedef uint8_t Uint8; -typedef int16_t Sint16; -typedef uint16_t Uint16; -typedef int32_t Sint32; -typedef uint32_t Uint32; - -/* A structure to hold a set of audio conversion filters and buffers */ -struct SDL_AudioCVT { - int needed; /* Set to 1 if conversion possible */ - Uint16 src_format; /* Source audio format */ - Uint16 dst_format; /* Target audio format */ - double rate_incr; /* Rate conversion increment */ - Uint8 *buf; /* Buffer to hold entire audio data */ - int len; /* Length of original audio buffer */ - int len_cvt; /* Length of converted audio buffer */ - int len_mult; /* buffer must be len*len_mult big */ - double len_ratio; /* Given len, final size is len*len_ratio */ - void (*filters[10])(struct SDL_AudioCVT *cvt, Uint16 format); - int filter_index; /* Current audio conversion function */ -}; -int SDL_BuildAudioCVT(SDL_AudioCVT *cvt, - Uint16 src_format, Uint8 src_channels, int src_rate, - Uint16 dst_format, Uint8 dst_channels, int dst_rate); -int SDL_ConvertAudio(SDL_AudioCVT *cvt);