diff options
Diffstat (limited to 'distrib/sdl-1.2.15/src/audio/SDL_audiocvt.c')
| -rw-r--r-- | distrib/sdl-1.2.15/src/audio/SDL_audiocvt.c | 1510 |
1 files changed, 1510 insertions, 0 deletions
diff --git a/distrib/sdl-1.2.15/src/audio/SDL_audiocvt.c b/distrib/sdl-1.2.15/src/audio/SDL_audiocvt.c new file mode 100644 index 0000000..9b8fbcd --- /dev/null +++ b/distrib/sdl-1.2.15/src/audio/SDL_audiocvt.c @@ -0,0 +1,1510 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 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_config.h" + +/* Functions for audio drivers to perform runtime conversion of audio format */ + +#include "SDL_audio.h" + + +/* Effectively mix right and left channels into a single channel */ +void SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 ) { + SDL_SetError("No buffer allocated for conversion"); + 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 (SDLCALL *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); +} |