diff options
Diffstat (limited to 'src')
25 files changed, 2613 insertions, 2318 deletions
diff --git a/src/Makefile.am b/src/Makefile.am index d96543a0a..350099b66 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -1,6 +1,6 @@ SUBDIRS = audio_out video_out dxr3 input libmpeg2 libspudec demuxers \ - libac3 libffmpeg liblpcm libw32dll libmad xine-engine + liba52 libffmpeg liblpcm libw32dll libmad xine-engine debug: @list='$(SUBDIRS)'; for subdir in $$list; do \ diff --git a/src/audio_out/audio_alsa05_out.c b/src/audio_out/audio_alsa05_out.c index 79c547414..d66bc192a 100644 --- a/src/audio_out/audio_alsa05_out.c +++ b/src/audio_out/audio_alsa05_out.c @@ -21,10 +21,10 @@ * Plugin for ALSA Version 0.5.x * * Credits go - * for the SPDIF AC3 sync part + * for the SPDIF A52 sync part * (c) 2000 Andy Lo A Foe <andy@alsaplayer.org> * - * $Id: audio_alsa05_out.c,v 1.8 2001/08/25 08:19:58 guenter Exp $ + * $Id: audio_alsa05_out.c,v 1.9 2001/08/29 00:51:57 guenter Exp $ */ /* required for swab() */ @@ -174,7 +174,7 @@ static int ao_alsa_open(ao_driver_t *this_gen,uint32_t bits, uint32_t rate, int switch (ao_mode) { case AO_CAP_MODE_STEREO: - case AO_CAP_MODE_AC3: + case AO_CAP_MODE_A52: channels = 2; break; @@ -206,7 +206,7 @@ static int ao_alsa_open(ao_driver_t *this_gen,uint32_t bits, uint32_t rate, int this->bytes_in_buffer = 0; this->direction = SND_PCM_CHANNEL_PLAYBACK; - if (ao_mode == AO_CAP_MODE_AC3) { + if (ao_mode == AO_CAP_MODE_A52) { this->pcm_default_device = 2; mode = SND_PCM_MODE_BLOCK; } @@ -295,7 +295,7 @@ static int ao_alsa_open(ao_driver_t *this_gen,uint32_t bits, uint32_t rate, int this->stop_mode = pcm_chan_params.stop_mode; this->ao_mode = ao_mode; - if (ao_mode == AO_CAP_MODE_AC3) { + if (ao_mode == AO_CAP_MODE_A52) { pcm_chan_params.digital.dig_valid = 1; pcm_chan_params.digital.dig_status[0] = SND_PCM_DIG0_NONAUDIO; pcm_chan_params.digital.dig_status[0] |= SND_PCM_DIG0_PROFESSIONAL; @@ -528,8 +528,8 @@ ao_driver_t *init_audio_out_plugin(config_values_t *config) { this->pcm_default_device); this->capabilities = AO_CAP_MODE_STEREO; - if (config->lookup_int (config, "ac3_pass_through", 0)) - this->capabilities |= AO_CAP_MODE_AC3; + if (config->lookup_int (config, "a52_pass_through", 0)) + this->capabilities |= AO_CAP_MODE_A52; this->ao_driver.get_capabilities = ao_alsa_get_capabilities; diff --git a/src/audio_out/audio_oss_out.c b/src/audio_out/audio_oss_out.c index c0534673b..a205be43f 100644 --- a/src/audio_out/audio_oss_out.c +++ b/src/audio_out/audio_oss_out.c @@ -17,7 +17,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA * - * $Id: audio_oss_out.c,v 1.34 2001/08/25 08:11:41 guenter Exp $ + * $Id: audio_oss_out.c,v 1.35 2001/08/29 00:51:57 guenter Exp $ * * 20-8-2001 First implementation of Audio sync and Audio driver separation. * Copyright (C) 2001 James Courtier-Dutton James@superbug.demon.co.uk @@ -79,7 +79,7 @@ #endif #ifndef AFMT_AC3 -# define AFMT_AC3 0x00000400 /* Dolby Digital AC3 */ +# define AFMT_AC3 0x00000400 #endif #define AO_OUT_OSS_IFACE_VERSION 2 @@ -167,9 +167,9 @@ static int ao_oss_open(ao_driver_t *this_gen, /* * configure audio device - * In AC3 mode, skip all other SNDCTL commands + * In A52 mode, skip all other SNDCTL commands */ - if(!(mode & AO_CAP_MODE_AC3)) { + if(!(mode & AO_CAP_MODE_A52)) { tmp = (mode & AO_CAP_MODE_STEREO) ? 1 : 0; ioctl(this->audio_fd,SNDCTL_DSP_STEREO,&tmp); @@ -225,7 +225,7 @@ static int ao_oss_open(ao_driver_t *this_gen, ioctl(this->audio_fd, SNDCTL_DSP_CHANNELS, &tmp); this->num_channels = tmp; break; - case AO_CAP_MODE_AC3: + case AO_CAP_MODE_A52: tmp = AFMT_AC3; if (ioctl(this->audio_fd, SNDCTL_DSP_SETFMT, &tmp) < 0 || tmp != AFMT_AC3) { printf("audio_oss_out: AC3 SNDCTL_DSP_SETFMT failed. %d\n",tmp); @@ -233,7 +233,7 @@ static int ao_oss_open(ao_driver_t *this_gen, } this->num_channels = 2; /* FIXME: is this correct ? */ this->output_sample_rate = this->input_sample_rate; - printf ("audio_oss_out : AO_CAP_MODE_AC3\n"); + printf ("audio_oss_out : AO_CAP_MODE_A52\n"); break; } @@ -548,7 +548,7 @@ ao_driver_t *init_audio_out_plugin (config_values_t *config) { ioctl(audio_fd,SNDCTL_DSP_GETFMTS,&caps); if (caps & AFMT_AC3) { if (config->lookup_int (config, "ac3_pass_through", 0)) { - this->capabilities |= AO_CAP_MODE_AC3; + this->capabilities |= AO_CAP_MODE_A52; printf ("ac3-pass-through "); } else printf ("(ac3-pass-through not enabled in .xinerc)"); diff --git a/src/libac3/Makefile.am b/src/liba52/Makefile.am index 38b5f5b39..78ea2055f 100644 --- a/src/libac3/Makefile.am +++ b/src/liba52/Makefile.am @@ -4,14 +4,14 @@ LIBTOOL = $(SHELL) $(top_builddir)/libtool-nofpic libdir = $(XINE_PLUGINDIR) -lib_LTLIBRARIES = xineplug_decode_ac3.la +lib_LTLIBRARIES = xineplug_decode_a52.la -xineplug_decode_ac3_la_SOURCES = bitstream.c bit_allocate.c \ +xineplug_decode_a52_la_SOURCES = bitstream.c bit_allocate.c \ parse.c downmix.c imdct.c imdct_mlib.c \ xine_decoder.c -xineplug_decode_ac3_la_LDFLAGS = -avoid-version -module +xineplug_decode_a52_la_LDFLAGS = -avoid-version -module -noinst_HEADERS = ac3.h ac3_internal.h \ +noinst_HEADERS = a52.h a52_internal.h \ bitstream.h tables.h ## diff --git a/src/liba52/a52.h b/src/liba52/a52.h new file mode 100644 index 000000000..6825ec2bf --- /dev/null +++ b/src/liba52/a52.h @@ -0,0 +1,115 @@ +/* + * a52.h + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> + * + * This file is part of a52dec, a free ATSC A-52 stream decoder. + * + * a52dec 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. + * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef LIBA52_DOUBLE +typedef float sample_t; +#else +typedef double sample_t; +#endif + +typedef struct a52_ba_s { + uint16_t fsnroffst; /* fine SNR offset */ + uint16_t fgaincod; /* fast gain */ + uint16_t deltbae; /* delta bit allocation exists */ + int8_t deltba[50]; /* per-band delta bit allocation */ +} a52_ba_t; + +typedef struct a52_state_s { + uint8_t fscod; /* sample rate */ + uint8_t halfrate; /* halfrate factor */ + uint8_t acmod; /* coded channels */ + sample_t clev; /* centre channel mix level */ + sample_t slev; /* surround channels mix level */ + uint8_t lfeon; /* coded lfe channel */ + + int output; /* type of output */ + sample_t level; /* output level */ + sample_t bias; /* output bias */ + + int dynrnge; /* apply dynamic range */ + sample_t dynrng; /* dynamic range */ + void * dynrngdata; /* dynamic range callback funtion and data */ + sample_t (* dynrngcall) (sample_t range, void * dynrngdata); + + uint16_t cplinu; /* coupling in use */ + uint16_t chincpl[5]; /* channel coupled */ + uint16_t phsflginu; /* phase flags in use (stereo only) */ + uint16_t cplbndstrc[18]; /* coupling band structure */ + uint16_t cplstrtmant; /* coupling channel start mantissa */ + uint16_t cplendmant; /* coupling channel end mantissa */ + sample_t cplco[5][18]; /* coupling coordinates */ + + /* derived information */ + uint16_t cplstrtbnd; /* coupling start band (for bit allocation) */ + uint16_t ncplbnd; /* number of coupling bands */ + + uint16_t rematflg[4]; /* stereo rematrixing */ + + uint16_t endmant[5]; /* channel end mantissa */ + + uint8_t cpl_exp[256]; /* decoded coupling channel exponents */ + uint8_t fbw_exp[5][256]; /* decoded channel exponents */ + uint8_t lfe_exp[7]; /* decoded lfe channel exponents */ + + uint16_t sdcycod; /* slow decay */ + uint16_t fdcycod; /* fast decay */ + uint16_t sgaincod; /* slow gain */ + uint16_t dbpbcod; /* dB per bit - encodes the dbknee value */ + uint16_t floorcod; /* masking floor */ + + uint16_t csnroffst; /* coarse SNR offset */ + a52_ba_t cplba; /* coupling bit allocation parameters */ + a52_ba_t ba[5]; /* channel bit allocation parameters */ + a52_ba_t lfeba; /* lfe bit allocation parameters */ + + uint16_t cplfleak; /* coupling fast leak init */ + uint16_t cplsleak; /* coupling slow leak init */ + + /* derived bit allocation information */ + int8_t fbw_bap[5][256]; + int8_t cpl_bap[256]; + int8_t lfe_bap[7]; +} a52_state_t; + +#define A52_CHANNEL 0 +#define A52_MONO 1 +#define A52_STEREO 2 +#define A52_3F 3 +#define A52_2F1R 4 +#define A52_3F1R 5 +#define A52_2F2R 6 +#define A52_3F2R 7 +#define A52_CHANNEL1 8 +#define A52_CHANNEL2 9 +#define A52_DOLBY 10 +#define A52_CHANNEL_MASK 15 + +#define A52_LFE 16 +#define A52_ADJUST_LEVEL 32 + +sample_t * a52_init (uint32_t mm_accel); +int a52_syncinfo (uint8_t * buf, int * flags, + int * sample_rate, int * bit_rate); +int a52_frame (a52_state_t * state, uint8_t * buf, int * flags, + sample_t * level, sample_t bias); +void a52_dynrng (a52_state_t * state, + sample_t (* call) (sample_t, void *), void * data); +int a52_block (a52_state_t * state, sample_t * samples); diff --git a/src/liba52/a52_internal.h b/src/liba52/a52_internal.h new file mode 100644 index 000000000..a1b88c229 --- /dev/null +++ b/src/liba52/a52_internal.h @@ -0,0 +1,54 @@ +/* + * a52_internal.h + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> + * + * This file is part of a52dec, a free ATSC A-52 stream decoder. + * + * a52dec 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. + * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#define LEVEL_PLUS6DB 2.0 +#define LEVEL_PLUS3DB 1.4142135623730951 +#define LEVEL_3DB 0.7071067811865476 +#define LEVEL_45DB 0.5946035575013605 +#define LEVEL_6DB 0.5 + +#define EXP_REUSE (0) +#define EXP_D15 (1) +#define EXP_D25 (2) +#define EXP_D45 (3) + +#define DELTA_BIT_REUSE (0) +#define DELTA_BIT_NEW (1) +#define DELTA_BIT_NONE (2) +#define DELTA_BIT_RESERVED (3) + +void bit_allocate (a52_state_t * state, a52_ba_t * ba, int bndstart, + int start, int end, int fastleak, int slowleak, + uint8_t * exp, int8_t * bap); + +int downmix_init (int input, int flags, sample_t * level, + sample_t clev, sample_t slev); +int downmix_coeff (sample_t * coeff, int acmod, int output, sample_t level, + sample_t clev, sample_t slev); +void downmix (sample_t * samples, int acmod, int output, sample_t bias, + sample_t clev, sample_t slev); +void upmix (sample_t * samples, int acmod, int output); + +void imdct_init (uint32_t mm_accel); +extern void (* imdct_256) (sample_t * data, sample_t * delay, sample_t bias); +extern void (* imdct_512) (sample_t * data, sample_t * delay, sample_t bias); +void imdct_do_256_mlib (sample_t * data, sample_t * delay, sample_t bias); +void imdct_do_512_mlib (sample_t * data, sample_t * delay, sample_t bias); diff --git a/src/libac3/bit_allocate.c b/src/liba52/bit_allocate.c index 45bc16cef..a44142ae3 100644 --- a/src/libac3/bit_allocate.c +++ b/src/liba52/bit_allocate.c @@ -1,32 +1,30 @@ -/* - * bit_allocate.c +/* + * bit_allocate.c + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> * - * Copyright (C) Aaron Holtzman - May 1999 + * This file is part of a52dec, a free ATSC A-52 stream decoder. * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. + * a52dec 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. * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include "config.h" #include <inttypes.h> -#include "ac3.h" -#include "ac3_internal.h" +#include "a52.h" +#include "a52_internal.h" static int hthtab[3][50] = { {0x730, 0x730, 0x7c0, 0x800, 0x820, 0x840, 0x850, 0x850, 0x860, 0x860, @@ -52,7 +50,7 @@ static int8_t baptab[305] = { 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, - 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, // 93 padding entries + 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, /* 93 padding elems */ 16, 16, 16, 16, 16, 16, 16, 16, 16, 14, 14, 14, 14, 14, 14, 14, 14, 12, 12, 12, 12, 11, 11, 11, 11, 10, 10, 10, 10, 9, 9, 9, @@ -68,7 +66,7 @@ static int8_t baptab[305] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, 0 // 148 padding entries + 0, 0, 0, 0 /* 148 padding elems */ }; static int bndtab[30] = {21, 22, 23, 24, 25, 26, 27, 28, 31, 34, @@ -121,7 +119,7 @@ do { \ mask -= floor; \ } while (0) -void bit_allocate (ac3_state_t * state, ac3_ba_t * ba, int bndstart, +void bit_allocate (a52_state_t * state, a52_ba_t * ba, int bndstart, int start, int end, int fastleak, int slowleak, uint8_t * exp, int8_t * bap) { @@ -155,7 +153,7 @@ void bit_allocate (ac3_state_t * state, ac3_ba_t * ba, int bndstart, i = bndstart; j = start; - if (start == 0) { // not the coupling channel + if (start == 0) { /* not the coupling channel */ int lowcomp; lowcomp = 0; @@ -170,8 +168,7 @@ void bit_allocate (ac3_state_t * state, ac3_ba_t * ba, int bndstart, psd = 128 * exp[i]; mask = psd + fgain + lowcomp; COMPUTE_MASK (); - bap[i] = (baptab+156)[mask + 4 * exp[i]]; - i++; + bap[i++] = (baptab+156)[mask + 4 * exp[i]]; } while ((i < 3) || ((i < 7) && (exp[i] > exp[i-1]))); fastleak = psd + fgain; slowleak = psd + sgain; @@ -188,11 +185,10 @@ void bit_allocate (ac3_state_t * state, ac3_ba_t * ba, int bndstart, mask = ((fastleak + lowcomp < slowleak) ? fastleak + lowcomp : slowleak); COMPUTE_MASK (); - bap[i] = (baptab+156)[mask + 4 * exp[i]]; - i++; + bap[i++] = (baptab+156)[mask + 4 * exp[i]]; } - if (end == 7) // lfe channel + if (end == 7) /* lfe channel */ return; do { @@ -205,19 +201,17 @@ void bit_allocate (ac3_state_t * state, ac3_ba_t * ba, int bndstart, mask = ((fastleak + lowcomp < slowleak) ? fastleak + lowcomp : slowleak); COMPUTE_MASK (); - bap[i] = (baptab+156)[mask + 4 * exp[i]]; - i++; + bap[i++] = (baptab+156)[mask + 4 * exp[i]]; } while (i < 20); - while (lowcomp > 128) { // two iterations maximum + while (lowcomp > 128) { /* two iterations maximum */ lowcomp -= 128; psd = 128 * exp[i]; UPDATE_LEAK (); mask = ((fastleak + lowcomp < slowleak) ? fastleak + lowcomp : slowleak); COMPUTE_MASK (); - bap[i] = (baptab+156)[mask + 4 * exp[i]]; - i++; + bap[i++] = (baptab+156)[mask + 4 * exp[i]]; } j = i; } @@ -245,17 +239,16 @@ void bit_allocate (ac3_state_t * state, ac3_ba_t * ba, int bndstart, break; } } - // minpsd = -289 + /* minpsd = -289 */ UPDATE_LEAK (); mask = (fastleak < slowleak) ? fastleak : slowleak; COMPUTE_MASK (); i++; j = startband; do { - // max(mask+4*exp)=147=-(minpsd+fgain-deltba-snroffset)>>5+4*exp - // min(mask+4*exp)=-156=-(sgain-deltba-snroffset)>>5 - bap[j] = (baptab+156)[mask + 4 * exp[j]]; - j++; + /* max(mask+4*exp)=147=-(minpsd+fgain-deltba-snroffset)>>5+4*exp */ + /* min(mask+4*exp)=-156=-(sgain-deltba-snroffset)>>5 */ + bap[j++] = (baptab+156)[mask + 4 * exp[j]]; } while (j < endband); } while (j < end); } diff --git a/src/liba52/bitstream.c b/src/liba52/bitstream.c new file mode 100644 index 000000000..30caa5ea3 --- /dev/null +++ b/src/liba52/bitstream.c @@ -0,0 +1,99 @@ +/* + * bitstream.c + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> + * + * This file is part of a52dec, a free ATSC A-52 stream decoder. + * + * a52dec 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. + * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "config.h" + +#include <inttypes.h> + +#include "a52.h" +#include "a52_internal.h" +#include "bitstream.h" + +#define BUFFER_SIZE 4096 + +static uint32_t * buffer_start; + +uint32_t bits_left; +uint32_t current_word; + +void bitstream_set_ptr (uint8_t * buf) +{ + int align; + + align = (int)buf & 3; + buffer_start = (uint32_t *) (buf - align); + bits_left = 0; + bitstream_get (align * 8); +} + +static inline void +bitstream_fill_current() +{ + uint32_t tmp; + + tmp = *(buffer_start++); + current_word = swab32 (tmp); +} + +/* + * The fast paths for _get is in the + * bitstream.h header file so it can be inlined. + * + * The "bottom half" of this routine is suffixed _bh + * + * -ah + */ + +uint32_t +bitstream_get_bh(uint32_t num_bits) +{ + uint32_t result; + + num_bits -= bits_left; + result = (current_word << (32 - bits_left)) >> (32 - bits_left); + + bitstream_fill_current(); + + if(num_bits != 0) + result = (result << num_bits) | (current_word >> (32 - num_bits)); + + bits_left = 32 - num_bits; + + return result; +} + +int32_t +bitstream_get_bh_2(uint32_t num_bits) +{ + int32_t result; + + num_bits -= bits_left; + result = (((int32_t)current_word) << (32 - bits_left)) >> (32 - bits_left); + + bitstream_fill_current(); + + if(num_bits != 0) + result = (result << num_bits) | (current_word >> (32 - num_bits)); + + bits_left = 32 - num_bits; + + return result; +} diff --git a/src/liba52/bitstream.h b/src/liba52/bitstream.h new file mode 100644 index 000000000..1f3c447c0 --- /dev/null +++ b/src/liba52/bitstream.h @@ -0,0 +1,80 @@ +/* + * bitstream.h + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> + * + * This file is part of a52dec, a free ATSC A-52 stream decoder. + * + * a52dec 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. + * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +/* (stolen from the kernel) */ +#ifdef WORDS_BIGENDIAN + +# define swab32(x) (x) + +#else + +# if defined (__i386__) + +# define swab32(x) __i386_swab32(x) + static inline const uint32_t __i386_swab32(uint32_t x) + { + __asm__("bswap %0" : "=r" (x) : "0" (x)); + return x; + } + +# else + +# define swab32(x)\ +((((uint8_t*)&x)[0] << 24) | (((uint8_t*)&x)[1] << 16) | \ + (((uint8_t*)&x)[2] << 8) | (((uint8_t*)&x)[3])) + +# endif +#endif + +extern uint32_t bits_left; +extern uint32_t current_word; + +void bitstream_set_ptr (uint8_t * buf); +uint32_t bitstream_get_bh(uint32_t num_bits); +int32_t bitstream_get_bh_2(uint32_t num_bits); + +static inline uint32_t +bitstream_get(uint32_t num_bits) +{ + uint32_t result; + + if(num_bits < bits_left) { + result = (current_word << (32 - bits_left)) >> (32 - num_bits); + bits_left -= num_bits; + return result; + } + + return bitstream_get_bh(num_bits); +} + +static inline int32_t +bitstream_get_2(uint32_t num_bits) +{ + int32_t result; + + if(num_bits < bits_left) { + result = (((int32_t)current_word) << (32 - bits_left)) >> (32 - num_bits); + bits_left -= num_bits; + return result; + } + + return bitstream_get_bh_2(num_bits); +} diff --git a/src/liba52/downmix.c b/src/liba52/downmix.c new file mode 100644 index 000000000..6feeab433 --- /dev/null +++ b/src/liba52/downmix.c @@ -0,0 +1,653 @@ +/* + * downmix.c + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> + * + * This file is part of a52dec, a free ATSC A-52 stream decoder. + * + * a52dec 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. + * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "config.h" + +#include <inttypes.h> +#include <string.h> + +#include "a52.h" +#include "a52_internal.h" + +#define CONVERT(acmod,output) (((output) << 3) + (acmod)) + +int downmix_init (int input, int flags, sample_t * level, + sample_t clev, sample_t slev) +{ + static uint8_t table[11][8] = { + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_STEREO, + A52_STEREO, A52_STEREO, A52_STEREO, A52_STEREO}, + {A52_MONO, A52_MONO, A52_MONO, A52_MONO, + A52_MONO, A52_MONO, A52_MONO, A52_MONO}, + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_STEREO, + A52_STEREO, A52_STEREO, A52_STEREO, A52_STEREO}, + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_3F, + A52_STEREO, A52_3F, A52_STEREO, A52_3F}, + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_STEREO, + A52_2F1R, A52_2F1R, A52_2F1R, A52_2F1R}, + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_STEREO, + A52_2F1R, A52_3F1R, A52_2F1R, A52_3F1R}, + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_3F, + A52_2F2R, A52_2F2R, A52_2F2R, A52_2F2R}, + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_3F, + A52_2F2R, A52_3F2R, A52_2F2R, A52_3F2R}, + {A52_CHANNEL1, A52_MONO, A52_MONO, A52_MONO, + A52_MONO, A52_MONO, A52_MONO, A52_MONO}, + {A52_CHANNEL2, A52_MONO, A52_MONO, A52_MONO, + A52_MONO, A52_MONO, A52_MONO, A52_MONO}, + {A52_CHANNEL, A52_DOLBY, A52_STEREO, A52_DOLBY, + A52_DOLBY, A52_DOLBY, A52_DOLBY, A52_DOLBY} + }; + int output; + + output = flags & A52_CHANNEL_MASK; + if (output > A52_DOLBY) + return -1; + + output = table[output][input & 7]; + + if ((output == A52_STEREO) && + ((input == A52_DOLBY) || ((input == A52_3F) && (clev == LEVEL_3DB)))) + output = A52_DOLBY; + + if (flags & A52_ADJUST_LEVEL) + switch (CONVERT (input & 7, output)) { + + case CONVERT (A52_3F, A52_MONO): + *level *= LEVEL_3DB / (1 + clev); + break; + + case CONVERT (A52_STEREO, A52_MONO): + case CONVERT (A52_2F2R, A52_2F1R): + case CONVERT (A52_3F2R, A52_3F1R): + level_3db: + *level *= LEVEL_3DB; + break; + + case CONVERT (A52_3F2R, A52_2F1R): + if (clev < LEVEL_PLUS3DB - 1) + goto level_3db; + /* break thru */ + case CONVERT (A52_3F, A52_STEREO): + case CONVERT (A52_3F1R, A52_2F1R): + case CONVERT (A52_3F1R, A52_2F2R): + case CONVERT (A52_3F2R, A52_2F2R): + *level /= 1 + clev; + break; + + case CONVERT (A52_2F1R, A52_MONO): + *level *= LEVEL_PLUS3DB / (2 + slev); + break; + + case CONVERT (A52_2F1R, A52_STEREO): + case CONVERT (A52_3F1R, A52_3F): + *level /= 1 + slev * LEVEL_3DB; + break; + + case CONVERT (A52_3F1R, A52_MONO): + *level *= LEVEL_3DB / (1 + clev + 0.5 * slev); + break; + + case CONVERT (A52_3F1R, A52_STEREO): + *level /= 1 + clev + slev * LEVEL_3DB; + break; + + case CONVERT (A52_2F2R, A52_MONO): + *level *= LEVEL_3DB / (1 + slev); + break; + + case CONVERT (A52_2F2R, A52_STEREO): + case CONVERT (A52_3F2R, A52_3F): + *level /= 1 + slev; + break; + + case CONVERT (A52_3F2R, A52_MONO): + *level *= LEVEL_3DB / (1 + clev + slev); + break; + + case CONVERT (A52_3F2R, A52_STEREO): + *level /= 1 + clev + slev; + break; + + case CONVERT (A52_MONO, A52_DOLBY): + *level *= LEVEL_PLUS3DB; + break; + + case CONVERT (A52_3F, A52_DOLBY): + case CONVERT (A52_2F1R, A52_DOLBY): + *level *= 1 / (1 + LEVEL_3DB); + break; + + case CONVERT (A52_3F1R, A52_DOLBY): + case CONVERT (A52_2F2R, A52_DOLBY): + *level *= 1 / (1 + 2 * LEVEL_3DB); + break; + + case CONVERT (A52_3F2R, A52_DOLBY): + *level *= 1 / (1 + 3 * LEVEL_3DB); + break; + } + + return output; +} + +int downmix_coeff (sample_t * coeff, int acmod, int output, sample_t level, + sample_t clev, sample_t slev) +{ + switch (CONVERT (acmod, output & A52_CHANNEL_MASK)) { + + case CONVERT (A52_CHANNEL, A52_CHANNEL): + case CONVERT (A52_MONO, A52_MONO): + case CONVERT (A52_STEREO, A52_STEREO): + case CONVERT (A52_3F, A52_3F): + case CONVERT (A52_2F1R, A52_2F1R): + case CONVERT (A52_3F1R, A52_3F1R): + case CONVERT (A52_2F2R, A52_2F2R): + case CONVERT (A52_3F2R, A52_3F2R): + case CONVERT (A52_STEREO, A52_DOLBY): + coeff[0] = coeff[1] = coeff[2] = coeff[3] = coeff[4] = level; + return 0; + + case CONVERT (A52_CHANNEL, A52_MONO): + coeff[0] = coeff[1] = level * LEVEL_6DB; + return 3; + + case CONVERT (A52_STEREO, A52_MONO): + coeff[0] = coeff[1] = level * LEVEL_3DB; + return 3; + + case CONVERT (A52_3F, A52_MONO): + coeff[0] = coeff[2] = level * LEVEL_3DB; + coeff[1] = level * clev * LEVEL_PLUS3DB; + return 7; + + case CONVERT (A52_2F1R, A52_MONO): + coeff[0] = coeff[1] = level * LEVEL_3DB; + coeff[2] = level * slev * LEVEL_3DB; + return 7; + + case CONVERT (A52_2F2R, A52_MONO): + coeff[0] = coeff[1] = level * LEVEL_3DB; + coeff[2] = coeff[3] = level * slev * LEVEL_3DB; + return 15; + + case CONVERT (A52_3F1R, A52_MONO): + coeff[0] = coeff[2] = level * LEVEL_3DB; + coeff[1] = level * clev * LEVEL_PLUS3DB; + coeff[3] = level * slev * LEVEL_3DB; + return 15; + + case CONVERT (A52_3F2R, A52_MONO): + coeff[0] = coeff[2] = level * LEVEL_3DB; + coeff[1] = level * clev * LEVEL_PLUS3DB; + coeff[3] = coeff[4] = level * slev * LEVEL_3DB; + return 31; + + case CONVERT (A52_MONO, A52_DOLBY): + coeff[0] = level * LEVEL_3DB; + return 0; + + case CONVERT (A52_3F, A52_DOLBY): + clev = LEVEL_3DB; + case CONVERT (A52_3F, A52_STEREO): + case CONVERT (A52_3F1R, A52_2F1R): + case CONVERT (A52_3F2R, A52_2F2R): + coeff[0] = coeff[2] = coeff[3] = coeff[4] = level; + coeff[1] = level * clev; + return 7; + + case CONVERT (A52_2F1R, A52_DOLBY): + slev = 1; + case CONVERT (A52_2F1R, A52_STEREO): + coeff[0] = coeff[1] = level; + coeff[2] = level * slev * LEVEL_3DB; + return 7; + + case CONVERT (A52_3F1R, A52_DOLBY): + clev = LEVEL_3DB; + slev = 1; + case CONVERT (A52_3F1R, A52_STEREO): + coeff[0] = coeff[2] = level; + coeff[1] = level * clev; + coeff[3] = level * slev * LEVEL_3DB; + return 15; + + case CONVERT (A52_2F2R, A52_DOLBY): + slev = LEVEL_3DB; + case CONVERT (A52_2F2R, A52_STEREO): + coeff[0] = coeff[1] = level; + coeff[2] = coeff[3] = level * slev; + return 15; + + case CONVERT (A52_3F2R, A52_DOLBY): + clev = LEVEL_3DB; + case CONVERT (A52_3F2R, A52_2F1R): + slev = LEVEL_3DB; + case CONVERT (A52_3F2R, A52_STEREO): + coeff[0] = coeff[2] = level; + coeff[1] = level * clev; + coeff[3] = coeff[4] = level * slev; + return 31; + + case CONVERT (A52_3F1R, A52_3F): + coeff[0] = coeff[1] = coeff[2] = level; + coeff[3] = level * slev * LEVEL_3DB; + return 13; + + case CONVERT (A52_3F2R, A52_3F): + coeff[0] = coeff[1] = coeff[2] = level; + coeff[3] = coeff[4] = level * slev; + return 29; + + case CONVERT (A52_2F2R, A52_2F1R): + coeff[0] = coeff[1] = level; + coeff[2] = coeff[3] = level * LEVEL_3DB; + return 12; + + case CONVERT (A52_3F2R, A52_3F1R): + coeff[0] = coeff[1] = coeff[2] = level; + coeff[3] = coeff[4] = level * LEVEL_3DB; + return 24; + + case CONVERT (A52_2F1R, A52_2F2R): + coeff[0] = coeff[1] = level; + coeff[2] = level * LEVEL_3DB; + return 0; + + case CONVERT (A52_3F1R, A52_2F2R): + coeff[0] = coeff[2] = level; + coeff[1] = level * clev; + coeff[3] = level * LEVEL_3DB; + return 7; + + case CONVERT (A52_3F1R, A52_3F2R): + coeff[0] = coeff[1] = coeff[2] = level; + coeff[3] = level * LEVEL_3DB; + return 0; + + case CONVERT (A52_CHANNEL, A52_CHANNEL1): + coeff[0] = level; + coeff[1] = 0; + return 0; + + case CONVERT (A52_CHANNEL, A52_CHANNEL2): + coeff[0] = 0; + coeff[1] = level; + return 0; + } + + return -1; /* NOTREACHED */ +} + +static void mix2to1 (sample_t * dest, sample_t * src, sample_t bias) +{ + int i; + + for (i = 0; i < 256; i++) + dest[i] += src[i] + bias; +} + +static void mix3to1 (sample_t * samples, sample_t bias) +{ + int i; + + for (i = 0; i < 256; i++) + samples[i] += samples[i + 256] + samples[i + 512] + bias; +} + +static void mix4to1 (sample_t * samples, sample_t bias) +{ + int i; + + for (i = 0; i < 256; i++) + samples[i] += (samples[i + 256] + samples[i + 512] + + samples[i + 768] + bias); +} + +static void mix5to1 (sample_t * samples, sample_t bias) +{ + int i; + + for (i = 0; i < 256; i++) + samples[i] += (samples[i + 256] + samples[i + 512] + + samples[i + 768] + samples[i + 1024] + bias); +} + +static void mix3to2 (sample_t * samples, sample_t bias) +{ + int i; + sample_t common; + + for (i = 0; i < 256; i++) { + common = samples[i + 256] + bias; + samples[i] += common; + samples[i + 256] = samples[i + 512] + common; + } +} + +static void mix21to2 (sample_t * left, sample_t * right, sample_t bias) +{ + int i; + sample_t common; + + for (i = 0; i < 256; i++) { + common = right[i + 256] + bias; + left[i] += common; + right[i] += common; + } +} + +static void mix21toS (sample_t * samples, sample_t bias) +{ + int i; + sample_t surround; + + for (i = 0; i < 256; i++) { + surround = samples[i + 512]; + samples[i] += bias - surround; + samples[i + 256] += bias + surround; + } +} + +static void mix31to2 (sample_t * samples, sample_t bias) +{ + int i; + sample_t common; + + for (i = 0; i < 256; i++) { + common = samples[i + 256] + samples[i + 768] + bias; + samples[i] += common; + samples[i + 256] = samples[i + 512] + common; + } +} + +static void mix31toS (sample_t * samples, sample_t bias) +{ + int i; + sample_t common, surround; + + for (i = 0; i < 256; i++) { + common = samples[i + 256] + bias; + surround = samples[i + 768]; + samples[i] += common - surround; + samples[i + 256] = samples[i + 512] + common + surround; + } +} + +static void mix22toS (sample_t * samples, sample_t bias) +{ + int i; + sample_t surround; + + for (i = 0; i < 256; i++) { + surround = samples[i + 512] + samples[i + 768]; + samples[i] += bias - surround; + samples[i + 256] += bias + surround; + } +} + +static void mix32to2 (sample_t * samples, sample_t bias) +{ + int i; + sample_t common; + + for (i = 0; i < 256; i++) { + common = samples[i + 256] + bias; + samples[i] += common + samples[i + 768]; + samples[i + 256] = common + samples[i + 512] + samples[i + 1024]; + } +} + +static void mix32toS (sample_t * samples, sample_t bias) +{ + int i; + sample_t common, surround; + + for (i = 0; i < 256; i++) { + common = samples[i + 256] + bias; + surround = samples[i + 768] + samples[i + 1024]; + samples[i] += common - surround; + samples[i + 256] = samples[i + 512] + common + surround; + } +} + +static void move2to1 (sample_t * src, sample_t * dest, sample_t bias) +{ + int i; + + for (i = 0; i < 256; i++) + dest[i] = src[i] + src[i + 256] + bias; +} + +static void zero (sample_t * samples) +{ + int i; + + for (i = 0; i < 256; i++) + samples[i] = 0; +} + +void downmix (sample_t * samples, int acmod, int output, sample_t bias, + sample_t clev, sample_t slev) +{ + switch (CONVERT (acmod, output & A52_CHANNEL_MASK)) { + + case CONVERT (A52_CHANNEL, A52_CHANNEL2): + memcpy (samples, samples + 256, 256 * sizeof (sample_t)); + break; + + case CONVERT (A52_CHANNEL, A52_MONO): + case CONVERT (A52_STEREO, A52_MONO): + mix_2to1: + mix2to1 (samples, samples + 256, bias); + break; + + case CONVERT (A52_2F1R, A52_MONO): + if (slev == 0) + goto mix_2to1; + case CONVERT (A52_3F, A52_MONO): + mix_3to1: + mix3to1 (samples, bias); + break; + + case CONVERT (A52_3F1R, A52_MONO): + if (slev == 0) + goto mix_3to1; + case CONVERT (A52_2F2R, A52_MONO): + if (slev == 0) + goto mix_2to1; + mix4to1 (samples, bias); + break; + + case CONVERT (A52_3F2R, A52_MONO): + if (slev == 0) + goto mix_3to1; + mix5to1 (samples, bias); + break; + + case CONVERT (A52_MONO, A52_DOLBY): + memcpy (samples + 256, samples, 256 * sizeof (sample_t)); + break; + + case CONVERT (A52_3F, A52_STEREO): + case CONVERT (A52_3F, A52_DOLBY): + mix_3to2: + mix3to2 (samples, bias); + break; + + case CONVERT (A52_2F1R, A52_STEREO): + if (slev == 0) + break; + mix21to2 (samples, samples + 256, bias); + break; + + case CONVERT (A52_2F1R, A52_DOLBY): + mix21toS (samples, bias); + break; + + case CONVERT (A52_3F1R, A52_STEREO): + if (slev == 0) + goto mix_3to2; + mix31to2 (samples, bias); + break; + + case CONVERT (A52_3F1R, A52_DOLBY): + mix31toS (samples, bias); + break; + + case CONVERT (A52_2F2R, A52_STEREO): + if (slev == 0) + break; + mix2to1 (samples, samples + 512, bias); + mix2to1 (samples + 256, samples + 768, bias); + break; + + case CONVERT (A52_2F2R, A52_DOLBY): + mix22toS (samples, bias); + break; + + case CONVERT (A52_3F2R, A52_STEREO): + if (slev == 0) + goto mix_3to2; + mix32to2 (samples, bias); + break; + + case CONVERT (A52_3F2R, A52_DOLBY): + mix32toS (samples, bias); + break; + + case CONVERT (A52_3F1R, A52_3F): + if (slev == 0) + break; + mix21to2 (samples, samples + 512, bias); + break; + + case CONVERT (A52_3F2R, A52_3F): + if (slev == 0) + break; + mix2to1 (samples, samples + 768, bias); + mix2to1 (samples + 512, samples + 1024, bias); + break; + + case CONVERT (A52_3F1R, A52_2F1R): + mix3to2 (samples, bias); + memcpy (samples + 512, samples + 768, 256 * sizeof (sample_t)); + break; + + case CONVERT (A52_2F2R, A52_2F1R): + mix2to1 (samples + 512, samples + 768, bias); + break; + + case CONVERT (A52_3F2R, A52_2F1R): + mix3to2 (samples, bias); + move2to1 (samples + 768, samples + 512, bias); + break; + + case CONVERT (A52_3F2R, A52_3F1R): + mix2to1 (samples + 768, samples + 1024, bias); + break; + + case CONVERT (A52_2F1R, A52_2F2R): + memcpy (samples + 768, samples + 512, 256 * sizeof (sample_t)); + break; + + case CONVERT (A52_3F1R, A52_2F2R): + mix3to2 (samples, bias); + memcpy (samples + 512, samples + 768, 256 * sizeof (sample_t)); + break; + + case CONVERT (A52_3F2R, A52_2F2R): + mix3to2 (samples, bias); + memcpy (samples + 512, samples + 768, 256 * sizeof (sample_t)); + memcpy (samples + 768, samples + 1024, 256 * sizeof (sample_t)); + break; + + case CONVERT (A52_3F1R, A52_3F2R): + memcpy (samples + 1027, samples + 768, 256 * sizeof (sample_t)); + break; + } +} + +void upmix (sample_t * samples, int acmod, int output) +{ + switch (CONVERT (acmod, output & A52_CHANNEL_MASK)) { + + case CONVERT (A52_CHANNEL, A52_CHANNEL2): + memcpy (samples + 256, samples, 256 * sizeof (sample_t)); + break; + + case CONVERT (A52_3F2R, A52_MONO): + zero (samples + 1024); + case CONVERT (A52_3F1R, A52_MONO): + case CONVERT (A52_2F2R, A52_MONO): + zero (samples + 768); + case CONVERT (A52_3F, A52_MONO): + case CONVERT (A52_2F1R, A52_MONO): + zero (samples + 512); + case CONVERT (A52_CHANNEL, A52_MONO): + case CONVERT (A52_STEREO, A52_MONO): + zero (samples + 256); + break; + + case CONVERT (A52_3F2R, A52_STEREO): + case CONVERT (A52_3F2R, A52_DOLBY): + zero (samples + 1024); + case CONVERT (A52_3F1R, A52_STEREO): + case CONVERT (A52_3F1R, A52_DOLBY): + zero (samples + 768); + case CONVERT (A52_3F, A52_STEREO): + case CONVERT (A52_3F, A52_DOLBY): + mix_3to2: + memcpy (samples + 512, samples + 256, 256 * sizeof (sample_t)); + zero (samples + 256); + break; + + case CONVERT (A52_2F2R, A52_STEREO): + case CONVERT (A52_2F2R, A52_DOLBY): + zero (samples + 768); + case CONVERT (A52_2F1R, A52_STEREO): + case CONVERT (A52_2F1R, A52_DOLBY): + zero (samples + 512); + break; + + case CONVERT (A52_3F2R, A52_3F): + zero (samples + 1024); + case CONVERT (A52_3F1R, A52_3F): + case CONVERT (A52_2F2R, A52_2F1R): + zero (samples + 768); + break; + + case CONVERT (A52_3F2R, A52_3F1R): + zero (samples + 1024); + break; + + case CONVERT (A52_3F2R, A52_2F1R): + zero (samples + 1024); + case CONVERT (A52_3F1R, A52_2F1R): + mix_31to21: + memcpy (samples + 768, samples + 512, 256 * sizeof (sample_t)); + goto mix_3to2; + + case CONVERT (A52_3F2R, A52_2F2R): + memcpy (samples + 1024, samples + 768, 256 * sizeof (sample_t)); + goto mix_31to21; + } +} diff --git a/src/libac3/imdct.c b/src/liba52/imdct.c index 7cc221c89..726cda973 100644 --- a/src/libac3/imdct.c +++ b/src/liba52/imdct.c @@ -1,25 +1,22 @@ -/* - * imdct.c +/* + * imdct.c + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> * - * Copyright (C) Aaron Holtzman - May 1999 + * This file is part of a52dec, a free ATSC A-52 stream decoder. * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. + * a52dec 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. * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include "config.h" @@ -27,11 +24,11 @@ #include <inttypes.h> #include <math.h> -#include "ac3.h" -#include "ac3_internal.h" +#include "a52.h" +#include "a52_internal.h" -void (* imdct_256) (sample_t data[], sample_t delay[]); -void (* imdct_512) (sample_t data[], sample_t delay[]); +void (* imdct_256) (sample_t data[], sample_t delay[], sample_t bias); +void (* imdct_512) (sample_t data[], sample_t delay[], sample_t bias); typedef struct complex_s { sample_t real; @@ -144,7 +141,7 @@ static inline complex_t cmplx_mult(complex_t a, complex_t b) } void -imdct_do_512(sample_t data[],sample_t delay[]) +imdct_do_512(sample_t data[],sample_t delay[], sample_t bias) { int i,k; int p,q; @@ -161,18 +158,16 @@ imdct_do_512(sample_t data[],sample_t delay[]) sample_t *delay_ptr; sample_t *window_ptr; - // - // 512 IMDCT with source and dest data in 'data' - // + /* 512 IMDCT with source and dest data in 'data' */ - // Pre IFFT complex multiply plus IFFT cmplx conjugate + /* Pre IFFT complex multiply plus IFFT cmplx conjugate */ for( i=0; i < 128; i++) { /* z[i] = (X[256-2*i-1] + j * X[2*i]) * (xcos1[i] + j * xsin1[i]) ; */ buf[i].real = (data[256-2*i-1] * xcos1[i]) - (data[2*i] * xsin1[i]); buf[i].imag = -1.0 * ((data[2*i] * xcos1[i]) + (data[256-2*i-1] * xsin1[i])); } - //Bit reversed shuffling + /* Bit reversed shuffling */ for(i=0; i<128; i++) { k = bit_reverse_512[i]; if (k < i) @@ -219,13 +214,13 @@ imdct_do_512(sample_t data[],sample_t delay[]) /* Window and convert to real valued signal */ for(i=0; i< 64; i++) { - *data_ptr++ = 2.0f * (-buf[64+i].imag * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf[64-i-1].real * *window_ptr++ + *delay_ptr++); + *data_ptr++ = -buf[64+i].imag * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf[64-i-1].real * *window_ptr++ + *delay_ptr++ + bias; } for(i=0; i< 64; i++) { - *data_ptr++ = 2.0f * (-buf[i].real * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf[128-i-1].imag * *window_ptr++ + *delay_ptr++); + *data_ptr++ = -buf[i].real * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf[128-i-1].imag * *window_ptr++ + *delay_ptr++ + bias; } /* The trailing edge of the window goes into the delay line */ @@ -243,7 +238,7 @@ imdct_do_512(sample_t data[],sample_t delay[]) } void -imdct_do_256(sample_t data[],sample_t delay[]) +imdct_do_256(sample_t data[],sample_t delay[],sample_t bias) { int i,k; int p,q; @@ -281,7 +276,7 @@ imdct_do_256(sample_t data[],sample_t delay[]) buf_2[k].imag = -1.0f * ( data[q + 1] * xcos2[k] + data[p + 1] * xsin2[k]); } - //IFFT Bit reversed shuffling + /* IFFT Bit reversed shuffling */ for(i=0; i<64; i++) { k = bit_reverse_256[i]; if (k < i) { @@ -295,7 +290,7 @@ imdct_do_256(sample_t data[],sample_t delay[]) two_m = (1 << m); two_m_plus_one = (1 << (m+1)); - //FIXME + /* FIXME */ if(m) two_m = (1 << m); else @@ -305,7 +300,7 @@ imdct_do_256(sample_t data[],sample_t delay[]) for(i = 0; i < 64; i += two_m_plus_one) { p = k + i; q = p + two_m; - //Do block 1 + /* Do block 1 */ tmp_a_r = buf_1[p].real; tmp_a_i = buf_1[p].imag; tmp_b_r = buf_1[q].real * w[m][k].real - buf_1[q].imag * w[m][k].imag; @@ -315,7 +310,7 @@ imdct_do_256(sample_t data[],sample_t delay[]) buf_1[q].real = tmp_a_r - tmp_b_r; buf_1[q].imag = tmp_a_i - tmp_b_i; - //Do block 2 + /* Do block 2 */ tmp_a_r = buf_2[p].real; tmp_a_i = buf_2[p].imag; tmp_b_r = buf_2[q].real * w[m][k].real - buf_2[q].imag * w[m][k].imag; @@ -348,13 +343,13 @@ imdct_do_256(sample_t data[],sample_t delay[]) /* Window and convert to real valued signal */ for(i=0; i< 64; i++) { - *data_ptr++ = 2.0f * (-buf_1[i].imag * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf_1[64-i-1].real * *window_ptr++ + *delay_ptr++); + *data_ptr++ = -buf_1[i].imag * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf_1[64-i-1].real * *window_ptr++ + *delay_ptr++ + bias; } for(i=0; i< 64; i++) { - *data_ptr++ = 2.0f * (-buf_1[i].real * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf_1[64-i-1].imag * *window_ptr++ + *delay_ptr++); + *data_ptr++ = -buf_1[i].real * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf_1[64-i-1].imag * *window_ptr++ + *delay_ptr++ + bias; } delay_ptr = delay; @@ -370,34 +365,37 @@ imdct_do_256(sample_t data[],sample_t delay[]) } } -void imdct_init (void) +void imdct_init (uint32_t mm_accel) { -#ifdef LIBAC3_MLIB - imdct_512 = imdct_do_512_mlib; - imdct_256 = imdct_do_256_mlib; -#else - int i, j, k; - - /* Twiddle factors to turn IFFT into IMDCT */ - for (i = 0; i < 128; i++) { - xcos1[i] = -cos ((M_PI / 2048) * (8 * i + 1)); - xsin1[i] = -sin ((M_PI / 2048) * (8 * i + 1)); - } +#ifdef LIBA52_MLIB + if (mm_accel & MM_ACCEL_MLIB) { + imdct_512 = imdct_do_512_mlib; + imdct_256 = imdct_do_256_mlib; + } else +#endif + { + int i, j, k; - /* More twiddle factors to turn IFFT into IMDCT */ - for (i = 0; i < 64; i++) { - xcos2[i] = -cos ((M_PI / 1024) * (8 * i + 1)); - xsin2[i] = -sin ((M_PI / 1024) * (8 * i + 1)); - } + /* Twiddle factors to turn IFFT into IMDCT */ + for (i = 0; i < 128; i++) { + xcos1[i] = -cos ((M_PI / 2048) * (8 * i + 1)); + xsin1[i] = -sin ((M_PI / 2048) * (8 * i + 1)); + } - for (i = 0; i < 7; i++) { - j = 1 << i; - for (k = 0; k < j; k++) { - w[i][k].real = cos (-M_PI * k / j); - w[i][k].imag = sin (-M_PI * k / j); + /* More twiddle factors to turn IFFT into IMDCT */ + for (i = 0; i < 64; i++) { + xcos2[i] = -cos ((M_PI / 1024) * (8 * i + 1)); + xsin2[i] = -sin ((M_PI / 1024) * (8 * i + 1)); } + + for (i = 0; i < 7; i++) { + j = 1 << i; + for (k = 0; k < j; k++) { + w[i][k].real = cos (-M_PI * k / j); + w[i][k].imag = sin (-M_PI * k / j); + } + } + imdct_512 = imdct_do_512; + imdct_256 = imdct_do_256; } - imdct_512 = imdct_do_512; - imdct_256 = imdct_do_256; -#endif } diff --git a/src/liba52/imdct_mlib.c b/src/liba52/imdct_mlib.c new file mode 100644 index 000000000..dba4b0af3 --- /dev/null +++ b/src/liba52/imdct_mlib.c @@ -0,0 +1,135 @@ +/* + * imdct_mlib.c + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> + * + * This file is part of a52dec, a free ATSC A-52 stream decoder. + * + * a52dec 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. + * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "config.h" + +#ifdef LIBA52_MLIB + +#include <inttypes.h> +#include <mlib_types.h> +#include <mlib_status.h> +#include <mlib_signal.h> + +#include "a52.h" +#include "a52_internal.h" + +extern sample_t imdct_window[]; + +void +imdct_do_512_mlib(sample_t data[], sample_t delay[], sample_t bias) +{ + sample_t *buf_real; + sample_t *buf_imag; + sample_t *data_ptr; + sample_t *delay_ptr; + sample_t *window_ptr; + sample_t tmp[256] __attribute__ ((__aligned__ (16))); + int i; + + memcpy(tmp, data, 256 * sizeof(sample_t)); + mlib_SignalIMDCT_F32(tmp); + + buf_real = tmp; + buf_imag = tmp + 128; + data_ptr = data; + delay_ptr = delay; + window_ptr = imdct_window; + + /* Window and convert to real valued signal */ + for(i=0; i< 64; i++) + { + *data_ptr++ = -buf_imag[64+i] * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf_real[64-i-1] * *window_ptr++ + *delay_ptr++ + bias; + } + + for(i=0; i< 64; i++) + { + *data_ptr++ = -buf_real[i] * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf_imag[128-i-1] * *window_ptr++ + *delay_ptr++ + bias; + } + + /* The trailing edge of the window goes into the delay line */ + delay_ptr = delay; + + for(i=0; i< 64; i++) + { + *delay_ptr++ = -buf_real[64+i] * *--window_ptr; + *delay_ptr++ = buf_imag[64-i-1] * *--window_ptr; + } + + for(i=0; i<64; i++) + { + *delay_ptr++ = buf_imag[i] * *--window_ptr; + *delay_ptr++ = -buf_real[128-i-1] * *--window_ptr; + } +} + +void +imdct_do_256_mlib(sample_t data[], sample_t delay[], sample_t bias) +{ + sample_t *buf1_real, *buf1_imag; + sample_t *buf2_real, *buf2_imag; + sample_t *data_ptr; + sample_t *delay_ptr; + sample_t *window_ptr; + sample_t tmp[256] __attribute__ ((__aligned__ (16))); + int i; + + memcpy(tmp, data, 256 * sizeof(sample_t)); + mlib_SignalIMDCT_F32(tmp); + + buf1_real = tmp; + buf1_imag = tmp + 64; + buf2_real = tmp + 128; + buf2_imag = tmp + 128 + 64; + data_ptr = data; + delay_ptr = delay; + window_ptr = imdct_window; + + /* Window and convert to real valued signal */ + for(i=0; i< 64; i++) + { + *data_ptr++ = -buf1_imag[i] * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf1_real[64-i-1] * *window_ptr++ + *delay_ptr++ + bias; + } + + for(i=0; i< 64; i++) + { + *data_ptr++ = -buf1_real[i] * *window_ptr++ + *delay_ptr++ + bias; + *data_ptr++ = buf1_imag[64-i-1] * *window_ptr++ + *delay_ptr++ + bias; + } + + delay_ptr = delay; + + for(i=0; i< 64; i++) + { + *delay_ptr++ = -buf2_real[i] * *--window_ptr; + *delay_ptr++ = buf2_imag[64-i-1] * *--window_ptr; + } + + for(i=0; i< 64; i++) + { + *delay_ptr++ = buf2_imag[i] * *--window_ptr; + *delay_ptr++ = -buf2_real[64-i-1] * *--window_ptr; + } +} + +#endif diff --git a/src/liba52/parse.c b/src/liba52/parse.c new file mode 100644 index 000000000..b6f47247e --- /dev/null +++ b/src/liba52/parse.c @@ -0,0 +1,861 @@ +/* + * parse.c + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> + * + * This file is part of a52dec, a free ATSC A-52 stream decoder. + * + * a52dec 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. + * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "config.h" + +#include <inttypes.h> +#include <string.h> + +#include "a52.h" +#include "a52_internal.h" +#include "bitstream.h" +#include "tables.h" + +#ifdef HAVE_MEMALIGN +/* some systems have memalign() but no declaration for it */ +void * memalign (size_t align, size_t size); +#else +/* assume malloc alignment is sufficient */ +#define memalign(align,size) malloc (size) +#endif + +sample_t * a52_init (uint32_t mm_accel) +{ + sample_t * samples; + int i; + + imdct_init (mm_accel); + + samples = memalign (16, 256 * 12 * sizeof (sample_t)); + if (samples == NULL) + return NULL; + + for (i = 0; i < 256 * 12; i++) + samples[i] = 0; + + return samples; +} + +static uint8_t halfrate[12] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 2, 3}; + +int a52_syncinfo (uint8_t * buf, int * flags, + int * sample_rate, int * bit_rate) +{ + static int rate[] = { 32, 40, 48, 56, 64, 80, 96, 112, + 128, 160, 192, 224, 256, 320, 384, 448, + 512, 576, 640}; + static uint8_t lfeon[8] = {0x10, 0x10, 0x04, 0x04, 0x04, 0x01, 0x04, 0x01}; + int frmsizecod; + int bitrate; + int half; + int acmod; + + if ((buf[0] != 0x0b) || (buf[1] != 0x77)) /* syncword */ + return 0; + + if (buf[5] >= 0x60) /* bsid >= 12 */ + return 0; + half = halfrate[buf[5] >> 3]; + + /* acmod, dsurmod and lfeon */ + acmod = buf[6] >> 5; + *flags = ((((buf[6] & 0xf8) == 0x50) ? A52_DOLBY : acmod) | + ((buf[6] & lfeon[acmod]) ? A52_LFE : 0)); + + frmsizecod = buf[4] & 63; + if (frmsizecod >= 38) + return 0; + bitrate = rate [frmsizecod >> 1]; + *bit_rate = (bitrate * 1000) >> half; + + switch (buf[4] & 0xc0) { + case 0: /* 48 KHz */ + *sample_rate = 48000 >> half; + return 4 * bitrate; + case 0x40: + *sample_rate = 44100 >> half; + return 2 * (320 * bitrate / 147 + (frmsizecod & 1)); + case 0x80: + *sample_rate = 32000 >> half; + return 6 * bitrate; + default: + return 0; + } +} + +int a52_frame (a52_state_t * state, uint8_t * buf, int * flags, + sample_t * level, sample_t bias) +{ + static sample_t clev[4] = {LEVEL_3DB, LEVEL_45DB, LEVEL_6DB, LEVEL_45DB}; + static sample_t slev[4] = {LEVEL_3DB, LEVEL_6DB, 0, LEVEL_6DB}; + int chaninfo; + int acmod; + + state->fscod = buf[4] >> 6; + state->halfrate = halfrate[buf[5] >> 3]; + state->acmod = acmod = buf[6] >> 5; + + bitstream_set_ptr (buf + 6); + bitstream_get (3); /* skip acmod we already parsed */ + + if ((acmod == 2) && (bitstream_get (2) == 2)) /* dsurmod */ + acmod = A52_DOLBY; + + if ((acmod & 1) && (acmod != 1)) + state->clev = clev[bitstream_get (2)]; /* cmixlev */ + + if (acmod & 4) + state->slev = slev[bitstream_get (2)]; /* surmixlev */ + + state->lfeon = bitstream_get (1); + + state->output = downmix_init (acmod, *flags, level, + state->clev, state->slev); + if (state->output < 0) + return 1; + if (state->lfeon && (*flags & A52_LFE)) + state->output |= A52_LFE; + *flags = state->output; + // the 2* compensates for differences in imdct + state->dynrng = state->level = 2 * *level; + state->bias = bias; + state->dynrnge = 1; + state->dynrngcall = NULL; + + chaninfo = !acmod; + do { + bitstream_get (5); /* dialnorm */ + if (bitstream_get (1)) /* compre */ + bitstream_get (8); /* compr */ + if (bitstream_get (1)) /* langcode */ + bitstream_get (8); /* langcod */ + if (bitstream_get (1)) /* audprodie */ + bitstream_get (7); /* mixlevel + roomtyp */ + } while (chaninfo--); + + bitstream_get (2); /* copyrightb + origbs */ + + if (bitstream_get (1)) /* timecod1e */ + bitstream_get (14); /* timecod1 */ + if (bitstream_get (1)) /* timecod2e */ + bitstream_get (14); /* timecod2 */ + + if (bitstream_get (1)) { /* addbsie */ + int addbsil; + + addbsil = bitstream_get (6); + do { + bitstream_get (8); /* addbsi */ + } while (addbsil--); + } + + return 0; +} + +void a52_dynrng (a52_state_t * state, + sample_t (* call) (sample_t, void *), void * data) +{ + state->dynrnge = 0; + if (call) { + state->dynrnge = 1; + state->dynrngcall = call; + state->dynrngdata = data; + } +} + +static int parse_exponents (int expstr, int ngrps, uint8_t exponent, + uint8_t * dest) +{ + int exps; + + while (ngrps--) { + exps = bitstream_get (7); + + exponent += exp_1[exps]; + if (exponent > 24) + return 1; + + switch (expstr) { + case EXP_D45: + *(dest++) = exponent; + *(dest++) = exponent; + case EXP_D25: + *(dest++) = exponent; + case EXP_D15: + *(dest++) = exponent; + } + + exponent += exp_2[exps]; + if (exponent > 24) + return 1; + + switch (expstr) { + case EXP_D45: + *(dest++) = exponent; + *(dest++) = exponent; + case EXP_D25: + *(dest++) = exponent; + case EXP_D15: + *(dest++) = exponent; + } + + exponent += exp_3[exps]; + if (exponent > 24) + return 1; + + switch (expstr) { + case EXP_D45: + *(dest++) = exponent; + *(dest++) = exponent; + case EXP_D25: + *(dest++) = exponent; + case EXP_D15: + *(dest++) = exponent; + } + } + + return 0; +} + +static int parse_deltba (int8_t * deltba) +{ + int deltnseg, deltlen, delta, j; + + memset (deltba, 0, 50); + + deltnseg = bitstream_get (3); + j = 0; + do { + j += bitstream_get (5); + deltlen = bitstream_get (4); + delta = bitstream_get (3); + delta -= (delta >= 4) ? 3 : 4; + if (!deltlen) + continue; + if (j + deltlen >= 50) + return 1; + while (deltlen--) + deltba[j++] = delta; + } while (deltnseg--); + + return 0; +} + +static inline int zero_snr_offsets (int nfchans, a52_state_t * state) +{ + int i; + + if ((state->csnroffst) || (state->cplinu && state->cplba.fsnroffst) || + (state->lfeon && state->lfeba.fsnroffst)) + return 0; + for (i = 0; i < nfchans; i++) + if (state->ba[i].fsnroffst) + return 0; + return 1; +} + +static sample_t q_1[2]; +static sample_t q_2[2]; +static sample_t q_4; +static int q_1_pointer; +static int q_2_pointer; +static int q_4_pointer; + +static uint16_t lfsr_state = 1; + +static inline int16_t dither_gen (void) +{ + int16_t state; + + state = dither_lut[lfsr_state >> 8] ^ (lfsr_state << 8); + + lfsr_state = (uint16_t) state; + + return state; +} + +static void coeff_get (sample_t * coeff, uint8_t * exp, int8_t * bap, + sample_t level, int dither, int end) +{ + int i; + sample_t factor[25]; + + for (i = 0; i <= 24; i++) + factor[i] = scale_factor[i] * level; + + i = 0; + while (i < end) { + int bapi; + + bapi = bap[i]; + switch (bapi) { + case 0: + if (dither) { + coeff[i++] = dither_gen() * LEVEL_3DB * factor[exp[i]]; + continue; + } else { + coeff[i++] = 0; + continue; + } + + case -1: + if (q_1_pointer >= 0) { + coeff[i++] = q_1[q_1_pointer--] * factor[exp[i]]; + continue; + } else { + int code; + + code = bitstream_get (5); + + q_1_pointer = 1; + q_1[0] = q_1_2[code]; + q_1[1] = q_1_1[code]; + coeff[i++] = q_1_0[code] * factor[exp[i]]; + continue; + } + + case -2: + if (q_2_pointer >= 0) { + coeff[i++] = q_2[q_2_pointer--] * factor[exp[i]]; + continue; + } else { + int code; + + code = bitstream_get (7); + + q_2_pointer = 1; + q_2[0] = q_2_2[code]; + q_2[1] = q_2_1[code]; + coeff[i++] = q_2_0[code] * factor[exp[i]]; + continue; + } + + case 3: + coeff[i++] = q_3[bitstream_get (3)] * factor[exp[i]]; + continue; + + case -3: + if (q_4_pointer == 0) { + q_4_pointer = -1; + coeff[i++] = q_4 * factor[exp[i]]; + continue; + } else { + int code; + + code = bitstream_get (7); + + q_4_pointer = 0; + q_4 = q_4_1[code]; + coeff[i++] = q_4_0[code] * factor[exp[i]]; + continue; + } + + case 4: + coeff[i++] = q_5[bitstream_get (4)] * factor[exp[i]]; + continue; + + default: + coeff[i++] = ((bitstream_get_2 (bapi) << (16 - bapi)) * + factor[exp[i]]); + } + } +} + +static void coeff_get_coupling (a52_state_t * state, int nfchans, + sample_t * coeff, + sample_t (* samples)[256], uint8_t dithflag[5]) +{ + int sub_bnd, bnd, i, i_end, ch; + int8_t * bap; + uint8_t * exp; + sample_t cplco[5]; + + bap = state->cpl_bap; + exp = state->cpl_exp; + sub_bnd = bnd = 0; + i = state->cplstrtmant; + while (i < state->cplendmant) { + i_end = i + 12; + while (state->cplbndstrc[sub_bnd++]) + i_end += 12; + for (ch = 0; ch < nfchans; ch++) + cplco[ch] = state->cplco[ch][bnd] * coeff[ch]; + bnd++; + + while (i < i_end) { + sample_t cplcoeff; + int bapi; + + bapi = bap[i]; + switch (bapi) { + case 0: + cplcoeff = LEVEL_3DB * scale_factor[exp[i]]; + for (ch = 0; ch < nfchans; ch++) + if (state->chincpl[ch]) { + if (dithflag[ch]) + samples[ch][i] = (cplcoeff * cplco[ch] * + dither_gen ()); + else + samples[ch][i] = 0; + } + i++; + continue; + + case -1: + if (q_1_pointer >= 0) { + cplcoeff = q_1[q_1_pointer--]; + break; + } else { + int code; + + code = bitstream_get (5); + + q_1_pointer = 1; + q_1[0] = q_1_2[code]; + q_1[1] = q_1_1[code]; + cplcoeff = q_1_0[code]; + break; + } + + case -2: + if (q_2_pointer >= 0) { + cplcoeff = q_2[q_2_pointer--]; + break; + } else { + int code; + + code = bitstream_get (7); + + q_2_pointer = 1; + q_2[0] = q_2_2[code]; + q_2[1] = q_2_1[code]; + cplcoeff = q_2_0[code]; + break; + } + + case 3: + cplcoeff = q_3[bitstream_get (3)]; + break; + + case -3: + if (q_4_pointer == 0) { + q_4_pointer = -1; + cplcoeff = q_4; + break; + } else { + int code; + + code = bitstream_get (7); + + q_4_pointer = 0; + q_4 = q_4_1[code]; + cplcoeff = q_4_0[code]; + break; + } + + case 4: + cplcoeff = q_5[bitstream_get (4)]; + break; + + default: + cplcoeff = bitstream_get_2 (bapi) << (16 - bapi); + } + + cplcoeff *= scale_factor[exp[i]]; + for (ch = 0; ch < nfchans; ch++) + if (state->chincpl[ch]) + samples[ch][i] = cplcoeff * cplco[ch]; + i++; + } + } +} + +int a52_block (a52_state_t * state, sample_t * samples) +{ + static const uint8_t nfchans_tbl[] = {2, 1, 2, 3, 3, 4, 4, 5, 1, 1, 2}; + static int rematrix_band[4] = {25, 37, 61, 253}; + int i, nfchans, chaninfo; + uint8_t cplexpstr, chexpstr[5], lfeexpstr, do_bit_alloc, done_cpl; + uint8_t blksw[5], dithflag[5]; + sample_t coeff[5]; + int chanbias; + + nfchans = nfchans_tbl[state->acmod]; + + for (i = 0; i < nfchans; i++) + blksw[i] = bitstream_get (1); + + for (i = 0; i < nfchans; i++) + dithflag[i] = bitstream_get (1); + + chaninfo = !(state->acmod); + do { + if (bitstream_get (1)) { /* dynrnge */ + int dynrng; + + dynrng = bitstream_get_2 (8); + if (state->dynrnge) { + sample_t range; + + range = ((((dynrng & 0x1f) | 0x20) << 13) * + scale_factor[3 - (dynrng >> 5)]); + if (state->dynrngcall) + range = state->dynrngcall (range, state->dynrngdata); + state->dynrng = state->level * range; + } + } + } while (chaninfo--); + + if (bitstream_get (1)) { /* cplstre */ + state->cplinu = bitstream_get (1); + if (state->cplinu) { + static int bndtab[16] = {31, 35, 37, 39, 41, 42, 43, 44, + 45, 45, 46, 46, 47, 47, 48, 48}; + int cplbegf; + int cplendf; + int ncplsubnd; + + for (i = 0; i < nfchans; i++) + state->chincpl[i] = bitstream_get (1); + switch (state->acmod) { + case 0: case 1: + return 1; + case 2: + state->phsflginu = bitstream_get (1); + } + cplbegf = bitstream_get (4); + cplendf = bitstream_get (4); + + if (cplendf + 3 - cplbegf < 0) + return 1; + state->ncplbnd = ncplsubnd = cplendf + 3 - cplbegf; + state->cplstrtbnd = bndtab[cplbegf]; + state->cplstrtmant = cplbegf * 12 + 37; + state->cplendmant = cplendf * 12 + 73; + + for (i = 0; i < ncplsubnd - 1; i++) { + state->cplbndstrc[i] = bitstream_get (1); + state->ncplbnd -= state->cplbndstrc[i]; + } + state->cplbndstrc[i] = 0; /* last value is a sentinel */ + } + } + + if (state->cplinu) { + int j, cplcoe; + + cplcoe = 0; + for (i = 0; i < nfchans; i++) + if (state->chincpl[i]) + if (bitstream_get (1)) { /* cplcoe */ + int mstrcplco, cplcoexp, cplcomant; + + cplcoe = 1; + mstrcplco = 3 * bitstream_get (2); + for (j = 0; j < state->ncplbnd; j++) { + cplcoexp = bitstream_get (4); + cplcomant = bitstream_get (4); + if (cplcoexp == 15) + cplcomant <<= 14; + else + cplcomant = (cplcomant | 0x10) << 13; + state->cplco[i][j] = + cplcomant * scale_factor[cplcoexp + mstrcplco]; + } + } + if ((state->acmod == 2) && state->phsflginu && cplcoe) + for (j = 0; j < state->ncplbnd; j++) + if (bitstream_get (1)) /* phsflg */ + state->cplco[1][j] = -state->cplco[1][j]; + } + + if ((state->acmod == 2) && (bitstream_get (1))) { /* rematstr */ + int end; + + end = (state->cplinu) ? state->cplstrtmant : 253; + i = 0; + do + state->rematflg[i] = bitstream_get (1); + while (rematrix_band[i++] < end); + } + + cplexpstr = EXP_REUSE; + lfeexpstr = EXP_REUSE; + if (state->cplinu) + cplexpstr = bitstream_get (2); + for (i = 0; i < nfchans; i++) + chexpstr[i] = bitstream_get (2); + if (state->lfeon) + lfeexpstr = bitstream_get (1); + + for (i = 0; i < nfchans; i++) + if (chexpstr[i] != EXP_REUSE) { + if (state->cplinu && state->chincpl[i]) + state->endmant[i] = state->cplstrtmant; + else { + int chbwcod; + + chbwcod = bitstream_get (6); + if (chbwcod > 60) + return 1; + state->endmant[i] = chbwcod * 3 + 73; + } + } + + do_bit_alloc = 0; + + if (cplexpstr != EXP_REUSE) { + int cplabsexp, ncplgrps; + + do_bit_alloc = 64; + ncplgrps = ((state->cplendmant - state->cplstrtmant) / + (3 << (cplexpstr - 1))); + cplabsexp = bitstream_get (4) << 1; + if (parse_exponents (cplexpstr, ncplgrps, cplabsexp, + state->cpl_exp + state->cplstrtmant)) + return 1; + } + for (i = 0; i < nfchans; i++) + if (chexpstr[i] != EXP_REUSE) { + int grp_size, nchgrps; + + do_bit_alloc |= 1 << i; + grp_size = 3 << (chexpstr[i] - 1); + nchgrps = (state->endmant[i] + grp_size - 4) / grp_size; + state->fbw_exp[i][0] = bitstream_get (4); + if (parse_exponents (chexpstr[i], nchgrps, state->fbw_exp[i][0], + state->fbw_exp[i] + 1)) + return 1; + bitstream_get (2); /* gainrng */ + } + if (lfeexpstr != EXP_REUSE) { + do_bit_alloc |= 32; + state->lfe_exp[0] = bitstream_get (4); + if (parse_exponents (lfeexpstr, 2, state->lfe_exp[0], + state->lfe_exp + 1)) + return 1; + } + + if (bitstream_get (1)) { /* baie */ + do_bit_alloc = -1; + state->sdcycod = bitstream_get (2); + state->fdcycod = bitstream_get (2); + state->sgaincod = bitstream_get (2); + state->dbpbcod = bitstream_get (2); + state->floorcod = bitstream_get (3); + } + if (bitstream_get (1)) { /* snroffste */ + do_bit_alloc = -1; + state->csnroffst = bitstream_get (6); + if (state->cplinu) { + state->cplba.fsnroffst = bitstream_get (4); + state->cplba.fgaincod = bitstream_get (3); + } + for (i = 0; i < nfchans; i++) { + state->ba[i].fsnroffst = bitstream_get (4); + state->ba[i].fgaincod = bitstream_get (3); + } + if (state->lfeon) { + state->lfeba.fsnroffst = bitstream_get (4); + state->lfeba.fgaincod = bitstream_get (3); + } + } + if ((state->cplinu) && (bitstream_get (1))) { /* cplleake */ + do_bit_alloc |= 64; + state->cplfleak = 2304 - (bitstream_get (3) << 8); + state->cplsleak = 2304 - (bitstream_get (3) << 8); + } + + if (bitstream_get (1)) { /* deltbaie */ + do_bit_alloc = -1; + if (state->cplinu) + state->cplba.deltbae = bitstream_get (2); + for (i = 0; i < nfchans; i++) + state->ba[i].deltbae = bitstream_get (2); + if (state->cplinu && (state->cplba.deltbae == DELTA_BIT_NEW) && + parse_deltba (state->cplba.deltba)) + return 1; + for (i = 0; i < nfchans; i++) + if ((state->ba[i].deltbae == DELTA_BIT_NEW) && + parse_deltba (state->ba[i].deltba)) + return 1; + } + + if (do_bit_alloc) { + if (zero_snr_offsets (nfchans, state)) { + memset (state->cpl_bap, 0, sizeof (state->cpl_bap)); + memset (state->fbw_bap, 0, sizeof (state->fbw_bap)); + memset (state->lfe_bap, 0, sizeof (state->lfe_bap)); + } else { + if (state->cplinu && (do_bit_alloc & 64)) + bit_allocate (state, &state->cplba, state->cplstrtbnd, + state->cplstrtmant, state->cplendmant, + state->cplfleak, state->cplsleak, + state->cpl_exp, state->cpl_bap); + for (i = 0; i < nfchans; i++) + if (do_bit_alloc & (1 << i)) + bit_allocate (state, state->ba + i, 0, 0, + state->endmant[i], 0, 0, state->fbw_exp[i], + state->fbw_bap[i]); + if (state->lfeon && (do_bit_alloc & 32)) { + state->lfeba.deltbae = DELTA_BIT_NONE; + bit_allocate (state, &state->lfeba, 0, 0, 7, 0, 0, + state->lfe_exp, state->lfe_bap); + } + } + } + + if (bitstream_get (1)) { /* skiple */ + i = bitstream_get (9); /* skipl */ + while (i--) + bitstream_get (8); + } + + if (state->output & A52_LFE) + samples += 256; /* shift for LFE channel */ + + chanbias = downmix_coeff (coeff, state->acmod, state->output, + state->dynrng, state->clev, state->slev); + + q_1_pointer = q_2_pointer = q_4_pointer = -1; + done_cpl = 0; + + for (i = 0; i < nfchans; i++) { + int j; + + coeff_get (samples + 256 * i, state->fbw_exp[i], state->fbw_bap[i], + coeff[i], dithflag[i], state->endmant[i]); + + if (state->cplinu && state->chincpl[i]) { + if (!done_cpl) { + done_cpl = 1; + coeff_get_coupling (state, nfchans, coeff, + (sample_t (*)[256])samples, dithflag); + } + j = state->cplendmant; + } else + j = state->endmant[i]; + do + (samples + 256 * i)[j] = 0; + while (++j < 256); + } + + if (state->acmod == 2) { + int j, end, band; + + end = ((state->endmant[0] < state->endmant[1]) ? + state->endmant[0] : state->endmant[1]); + + i = 0; + j = 13; + do { + if (!state->rematflg[i]) { + j = rematrix_band[i++]; + continue; + } + band = rematrix_band[i++]; + if (band > end) + band = end; + do { + sample_t tmp0, tmp1; + + tmp0 = samples[j]; + tmp1 = (samples+256)[j]; + samples[j] = tmp0 + tmp1; + (samples+256)[j] = tmp0 - tmp1; + } while (++j < band); + } while (j < end); + } + + if (state->lfeon) { + if (state->output & A52_LFE) { + coeff_get (samples - 256, state->lfe_exp, state->lfe_bap, + state->dynrng, 0, 7); + for (i = 7; i < 256; i++) + (samples-256)[i] = 0; + imdct_512 (samples - 256, samples + 1536 - 256, state->bias); + } else { + /* just skip the LFE coefficients */ + coeff_get (samples + 1280, state->lfe_exp, state->lfe_bap, + 0, 0, 7); + } + } + + i = 0; + if (nfchans_tbl[state->output & A52_CHANNEL_MASK] < nfchans) + for (i = 1; i < nfchans; i++) + if (blksw[i] != blksw[0]) + break; + + if (i < nfchans) { + if (samples[2 * 1536 - 1] == (sample_t)0x776b6e21) { + samples[2 * 1536 - 1] = 0; + upmix (samples + 1536, state->acmod, state->output); + } + + for (i = 0; i < nfchans; i++) { + sample_t bias; + + bias = 0; + if (!(chanbias & (1 << i))) + bias = state->bias; + + if (coeff[i]) { + if (blksw[i]) + imdct_256 (samples + 256 * i, samples + 1536 + 256 * i, + bias); + else + imdct_512 (samples + 256 * i, samples + 1536 + 256 * i, + bias); + } else { + int j; + + for (j = 0; j < 256; j++) + (samples + 256 * i)[j] = bias; + } + } + + downmix (samples, state->acmod, state->output, state->bias, + state->clev, state->slev); + } else { + nfchans = nfchans_tbl[state->output & A52_CHANNEL_MASK]; + + downmix (samples, state->acmod, state->output, 0, + state->clev, state->slev); + + if (samples[2 * 1536 - 1] != (sample_t)0x776b6e21) { + downmix (samples + 1536, state->acmod, state->output, 0, + state->clev, state->slev); + samples[2 * 1536 - 1] = (sample_t)0x776b6e21; + } + + if (blksw[0]) + for (i = 0; i < nfchans; i++) + imdct_256 (samples + 256 * i, samples + 1536 + 256 * i, + state->bias); + else + for (i = 0; i < nfchans; i++) + imdct_512 (samples + 256 * i, samples + 1536 + 256 * i, + state->bias); + } + + return 0; +} diff --git a/src/libac3/tables.h b/src/liba52/tables.h index 28f030895..ddac9e5cb 100644 --- a/src/libac3/tables.h +++ b/src/liba52/tables.h @@ -1,25 +1,22 @@ -/* - * parse.c +/* + * tables.h + * Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> * - * Copyright (C) Aaron Holtzman - May 1999 + * This file is part of a52dec, a free ATSC A-52 stream decoder. * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. + * a52dec 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. * + * a52dec is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ static int8_t exp_1[128] = { diff --git a/src/liba52/xine_decoder.c b/src/liba52/xine_decoder.c new file mode 100644 index 000000000..6d4b89078 --- /dev/null +++ b/src/liba52/xine_decoder.c @@ -0,0 +1,477 @@ +/* + * Copyright (C) 2000-2001 the xine project + * + * This file is part of xine, a unix video player. + * + * xine 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. + * + * xine is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA + * + * $Id: xine_decoder.c,v 1.1 2001/08/29 00:51:57 guenter Exp $ + * + * stuff needed to turn liba52 into a xine decoder plugin + */ + + +#include <stdlib.h> +#include <unistd.h> +#include <string.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> + +#include "audio_out.h" +#include "a52.h" +#include "a52_internal.h" +#include "buffer.h" +#include "xine_internal.h" +#include "cpu_accel.h" + +#undef DEBUG_A52 +#ifdef DEBUG_A52 +int a52file; +#endif + +typedef struct a52dec_decoder_s { + audio_decoder_t audio_decoder; + + uint32_t pts; + uint32_t last_pts; + + uint8_t frame_buffer[3840]; + uint8_t *frame_ptr; + int sync_todo; + int frame_length, frame_todo; + uint16_t syncword; + + a52_state_t a52_state; + int a52_flags; + int a52_bit_rate; + int a52_sample_rate; + float a52_level; + int have_lfe; + + int a52_flags_map[8]; + int ao_flags_map[8]; + + int16_t int_samples [6 * 256 * 6]; + sample_t *samples; + + ao_instance_t *audio_out; + int audio_caps; + int bypass_mode; + int output_sampling_rate; + int output_open; + int output_mode; + + int disable_dynrng ; +} a52dec_decoder_t; + +int a52dec_can_handle (audio_decoder_t *this_gen, int buf_type) { + return ((buf_type & 0xFFFF0000) == BUF_AUDIO_A52) ; +} + + +void a52dec_init (audio_decoder_t *this_gen, ao_instance_t *audio_out) { + + a52dec_decoder_t *this = (a52dec_decoder_t *) this_gen; + /* int i; */ + + this->audio_out = audio_out; + this->audio_caps = audio_out->get_capabilities(audio_out); + this->syncword = 0; + this->sync_todo = 7; + this->output_open = 0; + this->pts = 0; + this->last_pts = 0; + + this->samples = a52_init (mm_accel()); + + /* + * find out if this driver supports a52 output + * or, if not, how many channels we've got + */ + + if (this->audio_caps & AO_CAP_MODE_A52) + this->bypass_mode = 1; + else { + this->bypass_mode = 0; + + this->a52_flags_map[A52_MONO] = A52_MONO; + this->a52_flags_map[A52_STEREO] = A52_STEREO; + this->a52_flags_map[A52_3F] = A52_STEREO; + this->a52_flags_map[A52_2F1R] = A52_STEREO; + this->a52_flags_map[A52_3F1R] = A52_STEREO; + this->a52_flags_map[A52_2F2R] = A52_STEREO; + this->a52_flags_map[A52_3F2R] = A52_STEREO; + + this->ao_flags_map[A52_MONO] = AO_CAP_MODE_MONO; + this->ao_flags_map[A52_STEREO] = AO_CAP_MODE_STEREO; + this->ao_flags_map[A52_3F] = AO_CAP_MODE_STEREO; + this->ao_flags_map[A52_2F1R] = AO_CAP_MODE_STEREO; + this->ao_flags_map[A52_3F1R] = AO_CAP_MODE_STEREO; + this->ao_flags_map[A52_2F2R] = AO_CAP_MODE_STEREO; + this->ao_flags_map[A52_3F2R] = AO_CAP_MODE_STEREO; + + /* find best mode */ + if (this->audio_caps & AO_CAP_MODE_5_1CHANNEL) { + + this->a52_flags_map[A52_2F2R] = A52_2F2R; + this->a52_flags_map[A52_3F2R] = A52_3F2R | A52_LFE; + this->ao_flags_map[A52_2F2R] = AO_CAP_MODE_4CHANNEL; + this->ao_flags_map[A52_3F2R] = AO_CAP_MODE_5CHANNEL; + + } else if (this->audio_caps & AO_CAP_MODE_5CHANNEL) { + + this->a52_flags_map[A52_2F2R] = A52_2F2R; + this->a52_flags_map[A52_3F2R] = A52_3F2R; + this->ao_flags_map[A52_2F2R] = AO_CAP_MODE_4CHANNEL; + this->ao_flags_map[A52_3F2R] = AO_CAP_MODE_5CHANNEL; + + } else if (this->audio_caps & AO_CAP_MODE_4CHANNEL) { + + this->a52_flags_map[A52_2F2R] = A52_2F2R; + this->a52_flags_map[A52_3F2R] = A52_2F2R; + + this->ao_flags_map[A52_2F2R] = AO_CAP_MODE_4CHANNEL; + this->ao_flags_map[A52_3F2R] = AO_CAP_MODE_4CHANNEL; + + /* else if (this->audio_caps & AO_CAP_MODE_STEREO) + defaults are ok */ + } else if (!(this->audio_caps & AO_CAP_MODE_STEREO)) { + printf ("HELP! a mono-only audio driver?!\n"); + + this->a52_flags_map[A52_MONO] = A52_MONO; + this->a52_flags_map[A52_STEREO] = A52_MONO; + this->a52_flags_map[A52_3F] = A52_MONO; + this->a52_flags_map[A52_2F1R] = A52_MONO; + this->a52_flags_map[A52_3F1R] = A52_MONO; + this->a52_flags_map[A52_2F2R] = A52_MONO; + this->a52_flags_map[A52_3F2R] = A52_MONO; + + this->ao_flags_map[A52_MONO] = AO_CAP_MODE_MONO; + this->ao_flags_map[A52_STEREO] = AO_CAP_MODE_MONO; + this->ao_flags_map[A52_3F] = AO_CAP_MODE_MONO; + this->ao_flags_map[A52_2F1R] = AO_CAP_MODE_MONO; + this->ao_flags_map[A52_3F1R] = AO_CAP_MODE_MONO; + this->ao_flags_map[A52_2F2R] = AO_CAP_MODE_MONO; + this->ao_flags_map[A52_3F2R] = AO_CAP_MODE_MONO; + } + } + + /* + for (i = 0; i<8; i++) + this->a52_flags_map[i] |= A52_ADJUST_LEVEL; +*/ +#ifdef DEBUG_A52 + a52file = open ("test.a52", O_CREAT | O_WRONLY | O_TRUNC, 0644); +#endif +} + +static inline int16_t blah (int32_t i) +{ + if (i > 0x43c07fff) + return 32767; + else if (i < 0x43bf8000) + return -32768; + else + return i - 0x43c00000; +} + +static inline void float_to_int (float * _f, int16_t * s16, int num_channels) { + int i; + int32_t * f = (int32_t *) _f; /* XXX assumes IEEE float format */ + + for (i = 0; i < 256; i++) { + s16[num_channels*i] = blah (f[i]); + } +} + +static void a52dec_decode_frame (a52dec_decoder_t *this, uint32_t pts) { + + int output_mode = AO_CAP_MODE_STEREO; + + /* + * do we want to decode this frame in software? + */ + + if (!this->bypass_mode) { + + int a52_output_flags, i; + sample_t level = this->a52_level; + + /* + * oki, decode this frame in software + */ + + /* determine output mode */ + + a52_output_flags = this->a52_flags_map[this->a52_flags & A52_CHANNEL_MASK]; + + if (a52_frame (&this->a52_state, + this->frame_buffer, + &a52_output_flags, + &level, 384)) { + printf ("liba52: a52_frame error\n"); + return; + } + + if (this->disable_dynrng) + a52_dynrng (&this->a52_state, NULL, NULL); + + this->have_lfe = a52_output_flags & A52_LFE; + if (this->have_lfe) + output_mode = AO_CAP_MODE_5_1CHANNEL; + else + output_mode = this->ao_flags_map[a52_output_flags]; + + /* + * (re-)open output device + */ + + if (!this->output_open + || (this->a52_sample_rate != this->output_sampling_rate) + || (output_mode != this->output_mode)) { + + if (this->output_open) + this->audio_out->close (this->audio_out); + + + this->output_open = (this->audio_out->open (this->audio_out, 16, + this->a52_sample_rate, + output_mode) == 1); + this->output_sampling_rate = this->a52_sample_rate; + this->output_mode = output_mode; + } + + + if (!this->output_open) + return; + + + /* + * decode a52 and convert/interleave samples + */ + + for (i = 0; i < 6; i++) { + if (a52_block (&this->a52_state, this->samples)) { + printf ("liba52: a52_block error\n"); + return; + } + + switch (output_mode) { + case AO_CAP_MODE_MONO: + float_to_int (&this->samples[0], this->int_samples+(i*256), 1); + break; + case AO_CAP_MODE_STEREO: + float_to_int (&this->samples[0*256], this->int_samples+(i*256*2), 2); + float_to_int (&this->samples[1*256], this->int_samples+(i*256*2)+1, 2); + break; + case AO_CAP_MODE_4CHANNEL: + float_to_int (&this->samples[0*256], this->int_samples+(i*256*4), 4); /* L */ + float_to_int (&this->samples[1*256], this->int_samples+(i*256*4)+1, 4); /* R */ + float_to_int (&this->samples[2*256], this->int_samples+(i*256*4)+2, 4); /* RL */ + float_to_int (&this->samples[3*256], this->int_samples+(i*256*4)+3, 4); /* RR */ + break; + case AO_CAP_MODE_5CHANNEL: + float_to_int (&this->samples[0*256], this->int_samples+(i*256*5)+0, 5); /* L */ + float_to_int (&this->samples[1*256], this->int_samples+(i*256*5)+4, 5); /* C */ + float_to_int (&this->samples[2*256], this->int_samples+(i*256*5)+1, 5); /* R */ + float_to_int (&this->samples[3*256], this->int_samples+(i*256*5)+2, 5); /* RL */ + float_to_int (&this->samples[4*256], this->int_samples+(i*256*5)+3, 5); /* RR */ + break; + case AO_CAP_MODE_5_1CHANNEL: + float_to_int (&this->samples[0*256], this->int_samples+(i*256*6)+5, 6); /* lfe */ + float_to_int (&this->samples[1*256], this->int_samples+(i*256*6)+0, 6); /* L */ + float_to_int (&this->samples[2*256], this->int_samples+(i*256*6)+4, 6); /* C */ + float_to_int (&this->samples[3*256], this->int_samples+(i*256*6)+1, 6); /* R */ + float_to_int (&this->samples[4*256], this->int_samples+(i*256*6)+2, 6); /* RL */ + float_to_int (&this->samples[5*256], this->int_samples+(i*256*6)+3, 6); /* RR */ + break; + default: + printf ("liba52: help - unsupported mode %08x\n", output_mode); + } + } + + /* output decoded samples */ + + this->audio_out->write (this->audio_out, + this->int_samples, + 256*6, + pts); + pts = 0; + + } else { + + /* + * loop through a52 data + */ + + if (!this->output_open) { + + int sample_rate, bit_rate, flags; + + a52_syncinfo (this->frame_buffer, &flags, &sample_rate, &bit_rate); + + this->output_open = (this->audio_out->open (this->audio_out, 16, + sample_rate, + AO_CAP_MODE_A52) == 1); + this->output_mode = AO_CAP_MODE_A52; + } + + if (this->output_open) { + this->audio_out->write (this->audio_out, + (int16_t*)this->frame_buffer, + this->frame_length, + pts); + } + } +} + +void a52dec_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { + + a52dec_decoder_t *this = (a52dec_decoder_t *) this_gen; + uint8_t *current = buf->content; + uint8_t *end = buf->content + buf->size; + uint8_t byte; + + if (buf->decoder_info[0] == 0) + return; + + /* + printf ("liba52: got buffer, pts =%d, pts - last_pts=%d\n", + buf->PTS, buf->PTS - this->last_pts); + + this->last_pts = buf->PTS; + */ + + if (buf->PTS) + this->pts = buf->PTS; + + + while (current != end) { + + if ( (this->sync_todo == 0) && (this->frame_todo == 0) ) { + a52dec_decode_frame (this, this->pts); +#ifdef DEBUG_A52 + write (a52file, this->frame_buffer, this->frame_length); +#endif + this->pts = 0; + this->sync_todo = 7; + this->syncword = 0; + } + + while (1) { + byte = *current++; + + if (this->sync_todo>0) { + + /* search and collect syncinfo */ + + if (this->syncword != 0x0b77) { + this->syncword = (this->syncword << 8) | byte; + + if (this->syncword == 0x0b77) { + + this->frame_buffer[0] = 0x0b; + this->frame_buffer[1] = 0x77; + + this->sync_todo = 5; + this->frame_ptr = this->frame_buffer+2; + } + } else { + *this->frame_ptr++ = byte; + this->sync_todo--; + + if (this->sync_todo==0) { + + this->frame_length = a52_syncinfo (this->frame_buffer, + &this->a52_flags, + &this->a52_sample_rate, + &this->a52_bit_rate); + if (this->frame_length) { + this->frame_todo = this->frame_length - 7; + } else { + this->sync_todo = 7; + this->syncword = 0; + printf ("liba52: skip frame of zero length\n"); + } + + } + + } + } else { + + *this->frame_ptr++ = byte; + this->frame_todo--; + + if (this->frame_todo == 0) { + if (current == end) + return ; + break; + } + } + + if (current == end) + return ; + } + } +} + +void a52dec_close (audio_decoder_t *this_gen) { + + a52dec_decoder_t *this = (a52dec_decoder_t *) this_gen; + + if (this->output_open) + this->audio_out->close (this->audio_out); + + this->output_open = 0; + +#ifdef DEBUG_A52 + close (a52file); +#endif +} + +static char *a52dec_get_id(void) { + return "a/52dec"; +} + +audio_decoder_t *init_audio_decoder_plugin (int iface_version, config_values_t *cfg) { + + a52dec_decoder_t *this ; + + if (iface_version != 2) { + printf( "liba52: plugin doesn't support plugin API version %d.\n" + "liba52: this means there's a version mismatch between xine and this " + "liba52: decoder plugin.\nInstalling current plugins should help.\n", + iface_version); + return NULL; + } + + this = (a52dec_decoder_t *) malloc (sizeof (a52dec_decoder_t)); + + this->audio_decoder.interface_version = 2; + this->audio_decoder.can_handle = a52dec_can_handle; + this->audio_decoder.init = a52dec_init; + this->audio_decoder.decode_data = a52dec_decode_data; + this->audio_decoder.close = a52dec_close; + this->audio_decoder.get_identifier = a52dec_get_id; + this->audio_decoder.priority = 1; + + + this->a52_level = (float) cfg->lookup_int (cfg, "a52_level", 100) / 100.0; + this->disable_dynrng = !cfg->lookup_int (cfg, "a52_dynrng", 0); + + return (audio_decoder_t *) this; +} + diff --git a/src/libac3/ac3.h b/src/libac3/ac3.h deleted file mode 100644 index 03b849357..000000000 --- a/src/libac3/ac3.h +++ /dev/null @@ -1,109 +0,0 @@ -/* - * ac3.h - * - * Copyright (C) Aaron Holtzman - May 1999 - * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. - * - * - */ - -typedef float sample_t; - -typedef struct ac3_ba_s { - uint16_t fsnroffst; // fine SNR offset - uint16_t fgaincod; // fast gain - uint16_t deltbae; // delta bit allocation exists - int8_t deltba[50]; // per-band delta bit allocation -} ac3_ba_t; - -typedef struct ac3_state_s { - sample_t * delay; // delay samples for imdct - - uint8_t fscod; // sample rate - uint8_t halfrate; // halfrate factor - uint8_t acmod; // coded channels - sample_t clev; // centre channel mix level - sample_t slev; // surround channels mix level - uint8_t lfeon; // coded lfe channel - - int output; // type of output - sample_t level; // output level - sample_t bias; // output bias - - uint16_t cplinu; // coupling in use - uint16_t chincpl[5]; // channel coupled - uint16_t phsflginu; // phase flags in use (stereo only) - uint16_t cplbndstrc[18]; // coupling band structure - uint16_t cplstrtmant; // coupling channel start mantissa - uint16_t cplendmant; // coupling channel end mantissa - sample_t cplco[5][18]; // coupling coordinates - - // derived information - uint16_t cplstrtbnd; // coupling start band (for bit allocation) - uint16_t ncplbnd; // number of coupling bands - - uint16_t rematflg[4]; // stereo rematrixing - - uint16_t endmant[5]; // channel end mantissa - - uint8_t cpl_exp[256]; // decoded coupling channel exponents - uint8_t fbw_exp[5][256]; // decoded channel exponents - uint8_t lfe_exp[7]; // decoded lfe channel exponents - - uint16_t sdcycod; // slow decay - uint16_t fdcycod; // fast decay - uint16_t sgaincod; // slow gain - uint16_t dbpbcod; // dB per bit - encodes the dbknee value - uint16_t floorcod; // masking floor - - uint16_t csnroffst; // coarse SNR offset - ac3_ba_t cplba; // coupling bit allocation parameters - ac3_ba_t ba[5]; // channel bit allocation parameters - ac3_ba_t lfeba; // lfe bit allocation parameters - - uint16_t cplfleak; // coupling fast leak init - uint16_t cplsleak; // coupling slow leak init - - // derived bit allocation information - int8_t fbw_bap[5][256]; - int8_t cpl_bap[256]; - int8_t lfe_bap[7]; -} ac3_state_t; - -#define AC3_CHANNEL 0 -#define AC3_MONO 1 -#define AC3_STEREO 2 -#define AC3_3F 3 -#define AC3_2F1R 4 -#define AC3_3F1R 5 -#define AC3_2F2R 6 -#define AC3_3F2R 7 -#define AC3_CHANNEL1 8 -#define AC3_CHANNEL2 9 -#define AC3_DOLBY 10 -#define AC3_CHANNEL_MASK 15 - -#define AC3_LFE 16 -#define AC3_ADJUST_LEVEL 32 - -void ac3_init (void); -int ac3_syncinfo (uint8_t * buf, int * flags, - int * sample_rate, int * bit_rate); -int ac3_frame (ac3_state_t * state, uint8_t * buf, int * flags, - sample_t * level, sample_t bias, sample_t * delay); -int ac3_block (ac3_state_t * state, sample_t samples[][256]); diff --git a/src/libac3/ac3_internal.h b/src/libac3/ac3_internal.h deleted file mode 100644 index 065ccbb36..000000000 --- a/src/libac3/ac3_internal.h +++ /dev/null @@ -1,52 +0,0 @@ -/* - * ac3_internal.h - * - * Copyright (C) Aaron Holtzman - May 1999 - * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#define LEVEL_PLUS3DB 1.4142135623730951 -#define LEVEL_3DB 0.7071067811865476 -#define LEVEL_45DB 0.5946035575013605 -#define LEVEL_6DB 0.5 - -#define EXP_REUSE (0) -#define EXP_D15 (1) -#define EXP_D25 (2) -#define EXP_D45 (3) - -#define DELTA_BIT_REUSE (0) -#define DELTA_BIT_NEW (1) -#define DELTA_BIT_NONE (2) -#define DELTA_BIT_RESERVED (3) - -void bit_allocate (ac3_state_t * state, ac3_ba_t * ba, int bndstart, - int start, int end, int fastleak, int slowleak, - uint8_t * exp, int8_t * bap); - -int downmix_init (int input, int flags, sample_t * level, - sample_t clev, sample_t slev); -void downmix (sample_t * samples, int acmod, int output, - sample_t level, sample_t bias, sample_t clev, sample_t slev); - -void imdct_init (void); -extern void (* imdct_256) (sample_t * data, sample_t * delay); -extern void (* imdct_512) (sample_t * data, sample_t * delay); -void imdct_do_256_mlib (sample_t * data, sample_t * delay); -void imdct_do_512_mlib (sample_t * data, sample_t * delay); diff --git a/src/libac3/bitstream.c b/src/libac3/bitstream.c deleted file mode 100644 index 6e2e36497..000000000 --- a/src/libac3/bitstream.c +++ /dev/null @@ -1,87 +0,0 @@ -/* - * bitstream.c - * - * Copyright (C) Aaron Holtzman - Dec 1999 - * - * This file is part of ac3dec, a free AC-3 audio decoder - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#include <inttypes.h> -#include <stddef.h> -#include <stdlib.h> -#include <stdio.h> - -#include "ac3.h" -#include "ac3_internal.h" -#include "bitstream.h" - -#define BUFFER_SIZE 4096 - -struct uint32_alignment { - char a; - uint32_t b; -}; -#define UINT32_ALIGNMENT offsetof(struct uint32_alignment, b) - - -static uint32_t *buffer_start; - -uint32_t bits_left; -uint32_t current_word; - -void bitstream_set_ptr (uint8_t * buf) -{ - int align = (long)buf & (UINT32_ALIGNMENT-1); - buffer_start = (uint32_t *) (buf - align); - bits_left = 0; - if (align > 0) bitstream_get(align * 8); -} - -static inline void -bitstream_fill_current() -{ - uint32_t tmp_word = *buffer_start++; - current_word = swab32(tmp_word); -} - -// -// The fast paths for _get is in the -// bitstream.h header file so it can be inlined. -// -// The "bottom half" of this routine is suffixed _bh -// -// -ah -// - -uint32_t -bitstream_get_bh(uint32_t num_bits) -{ - uint32_t result; - - num_bits -= bits_left; - result = (current_word << (32 - bits_left)) >> (32 - bits_left); - - bitstream_fill_current(); - - if(num_bits != 0) - result = (result << num_bits) | (current_word >> (32 - num_bits)); - - bits_left = 32 - num_bits; - - return result; -} diff --git a/src/libac3/bitstream.h b/src/libac3/bitstream.h deleted file mode 100644 index 84f3287c8..000000000 --- a/src/libac3/bitstream.h +++ /dev/null @@ -1,68 +0,0 @@ -/* - * bitstream.h - * - * Copyright (C) Aaron Holtzman - Dec 1999 - * - * This file is part of ac3dec, a free AC-3 audio decoder - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -//My new and improved vego-matic endian swapping routine -//(stolen from the kernel) -#ifdef WORDS_BIGENDIAN - -# define swab32(x) (x) - -#else - -# if defined (__i386__) - -# define swab32(x) __i386_swab32(x) - static inline const uint32_t __i386_swab32(uint32_t x) - { - __asm__("bswap %0" : "=r" (x) : "0" (x)); - return x; - } - -# else - -# define swab32(x)\ -((((uint8_t*)&x)[0] << 24) | (((uint8_t*)&x)[1] << 16) | \ - (((uint8_t*)&x)[2] << 8) | (((uint8_t*)&x)[3])) - -# endif -#endif - -extern uint32_t bits_left; -extern uint32_t current_word; - -void bitstream_set_ptr (uint8_t * buf); -uint32_t bitstream_get_bh(uint32_t num_bits); - -static inline uint32_t -bitstream_get(uint32_t num_bits) -{ - uint32_t result; - - if(num_bits < bits_left) { - result = (current_word << (32 - bits_left)) >> (32 - num_bits); - bits_left -= num_bits; - return result; - } - - return bitstream_get_bh(num_bits); -} diff --git a/src/libac3/downmix.c b/src/libac3/downmix.c deleted file mode 100644 index 498471798..000000000 --- a/src/libac3/downmix.c +++ /dev/null @@ -1,546 +0,0 @@ -/* - * - * downmix.c - * - * Copyright (C) Aaron Holtzman - Sept 1999 - * - * Originally based on code by Yuqing Deng. - * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. - * - * - */ - -#include "config.h" - -#include <inttypes.h> - -#include "ac3.h" -#include "ac3_internal.h" - - -#define CONVERT(acmod,output) (((output) << 3) + (acmod)) - -int downmix_init (int input, int flags, sample_t * level, - sample_t clev, sample_t slev) -{ - static uint8_t table[11][8] = { - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_STEREO, - AC3_STEREO, AC3_STEREO, AC3_STEREO, AC3_STEREO}, - {AC3_MONO, AC3_MONO, AC3_MONO, AC3_MONO, - AC3_MONO, AC3_MONO, AC3_MONO, AC3_MONO}, - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_STEREO, - AC3_STEREO, AC3_STEREO, AC3_STEREO, AC3_STEREO}, - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_3F, - AC3_STEREO, AC3_3F, AC3_STEREO, AC3_3F}, - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_STEREO, - AC3_2F1R, AC3_2F1R, AC3_2F1R, AC3_2F1R}, - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_STEREO, - AC3_2F1R, AC3_3F1R, AC3_2F1R, AC3_3F1R}, - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_3F, - AC3_2F2R, AC3_2F2R, AC3_2F2R, AC3_2F2R}, - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_3F, - AC3_2F2R, AC3_3F2R, AC3_2F2R, AC3_3F2R}, - {AC3_CHANNEL1, AC3_MONO, AC3_MONO, AC3_MONO, - AC3_MONO, AC3_MONO, AC3_MONO, AC3_MONO}, - {AC3_CHANNEL2, AC3_MONO, AC3_MONO, AC3_MONO, - AC3_MONO, AC3_MONO, AC3_MONO, AC3_MONO}, - {AC3_CHANNEL, AC3_DOLBY, AC3_STEREO, AC3_DOLBY, - AC3_DOLBY, AC3_DOLBY, AC3_DOLBY, AC3_DOLBY} - }; - int output; - - output = flags & AC3_CHANNEL_MASK; - if (output > AC3_DOLBY) - return -1; - - output = table[output][input & 7]; - - if ((output == AC3_STEREO) && - ((input == AC3_DOLBY) || ((input == AC3_3F) && (clev == LEVEL_3DB)))) - output = AC3_DOLBY; - - if (flags & AC3_ADJUST_LEVEL) - switch (CONVERT (input & 7, output)) { - - case CONVERT (AC3_3F, AC3_MONO): - *level *= LEVEL_3DB / (1 + clev); - break; - - case CONVERT (AC3_STEREO, AC3_MONO): - case CONVERT (AC3_2F2R, AC3_2F1R): - case CONVERT (AC3_3F2R, AC3_3F1R): - level_3db: - *level *= LEVEL_3DB; - break; - - case CONVERT (AC3_3F2R, AC3_2F1R): - if (clev < LEVEL_PLUS3DB - 1) - goto level_3db; - // break thru - case CONVERT (AC3_3F, AC3_STEREO): - case CONVERT (AC3_3F1R, AC3_2F1R): - case CONVERT (AC3_3F1R, AC3_2F2R): - case CONVERT (AC3_3F2R, AC3_2F2R): - *level /= 1 + clev; - break; - - case CONVERT (AC3_2F1R, AC3_MONO): - *level *= LEVEL_PLUS3DB / (2 + slev); - break; - - case CONVERT (AC3_2F1R, AC3_STEREO): - case CONVERT (AC3_3F1R, AC3_3F): - *level /= 1 + slev * LEVEL_3DB; - break; - - case CONVERT (AC3_3F1R, AC3_MONO): - *level *= LEVEL_3DB / (1 + clev + 0.5 * slev); - break; - - case CONVERT (AC3_3F1R, AC3_STEREO): - *level /= 1 + clev + slev * LEVEL_3DB; - break; - - case CONVERT (AC3_2F2R, AC3_MONO): - *level *= LEVEL_3DB / (1 + slev); - break; - - case CONVERT (AC3_2F2R, AC3_STEREO): - case CONVERT (AC3_3F2R, AC3_3F): - *level /= (1 + slev); - break; - - case CONVERT (AC3_3F2R, AC3_MONO): - *level *= LEVEL_3DB / (1 + clev + slev); - break; - - case CONVERT (AC3_3F2R, AC3_STEREO): - *level /= 1 + clev + slev; - break; - - case CONVERT (AC3_MONO, AC3_DOLBY): - *level *= LEVEL_PLUS3DB; - break; - - case CONVERT (AC3_3F, AC3_DOLBY): - case CONVERT (AC3_2F1R, AC3_DOLBY): - *level *= 1 / (1 + LEVEL_3DB); - break; - - case CONVERT (AC3_3F1R, AC3_DOLBY): - case CONVERT (AC3_2F2R, AC3_DOLBY): - *level *= 1 / (1 + 2 * LEVEL_3DB); - break; - - case CONVERT (AC3_3F2R, AC3_DOLBY): - *level *= 1 / (1 + 3 * LEVEL_3DB); - break; - } - - return output; -} - -static void mix1to1 (sample_t * samples, sample_t level, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - samples[i] = samples[i] * level + bias; -} - -static void move1to1 (sample_t * src, sample_t * dest, - sample_t level, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - dest[i] = src[i] * level + bias; -} - -static void mix2to1 (sample_t * samples, sample_t level, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - samples[i] = (samples[i] + samples[i + 256]) * level + bias; -} - -static void move2to1 (sample_t * src, sample_t * dest, - sample_t level, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - dest[i] = (src[i] + src[i + 256]) * level + bias; -} - -static void mix3to1 (sample_t * samples, sample_t level, sample_t clev, - sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - samples[i] = ((samples[i] + samples[i + 512]) * level + - samples[i + 256] * clev + bias); -} - -static void mix21to1 (sample_t * samples, sample_t level, sample_t slev, - sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - samples[i] = ((samples[i] + samples[i + 256]) * level + - samples[i + 512] * slev + bias); -} - -static void mix31to1 (sample_t * samples, sample_t level, - sample_t clev, sample_t slev, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - samples[i] = ((samples[i] + samples[i + 512]) * level + - samples[i + 256] * clev + samples[i + 768] * slev + - bias); -} - -static void mix22to1 (sample_t * samples, sample_t level, sample_t slev, - sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - samples[i] = ((samples[i] + samples[i + 256]) * level + - (samples[i + 512] + samples[i + 768]) * slev + bias); -} - -static void mix32to1 (sample_t * samples, sample_t level, - sample_t clev, sample_t slev, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - samples[i] = ((samples[i] + samples[i + 512]) * level + - samples[i + 256] * clev + - (samples[i + 768] + samples[i + 1024]) * slev + bias); -} - -static void mix1to2 (sample_t * src, sample_t * dest, - sample_t level, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - dest[i] = src[i] = src[i] * level + bias; -} - -static void mix3to2 (sample_t * samples, sample_t level, sample_t clev, - sample_t bias) -{ - int i; - sample_t common; - - for (i = 0; i < 256; i++) { - common = samples[i + 256] * clev + bias; - samples[i] = samples[i] * level + common; - samples[i + 256] = samples[i + 512] * level + common; - } -} - -static void mix21to2 (sample_t * left, sample_t * right, - sample_t level, sample_t slev, sample_t bias) -{ - int i; - sample_t common; - - for (i = 0; i < 256; i++) { - common = right[i + 256] * slev + bias; - left[i] = left[i] * level + common; - right[i] = right[i] * level + common; - } -} - -static void mix11to1 (sample_t * front, sample_t * rear, - sample_t level, sample_t slev, sample_t bias) -{ - int i; - - for (i = 0; i < 256; i++) - front[i] = front[i] * level + rear[i] * slev + bias; -} - -static void mix31to2 (sample_t * samples, sample_t level, - sample_t clev, sample_t slev, sample_t bias) -{ - int i; - sample_t common; - - for (i = 0; i < 256; i++) { - common = samples[i + 256] * clev + samples[i + 768] * slev + bias; - samples[i] = samples[i] * level + common; - samples[i + 256] = samples[i + 512] * level + common; - } -} - -static void mix32to2 (sample_t * samples, sample_t level, - sample_t clev, sample_t slev, sample_t bias) -{ - int i; - sample_t common; - - for (i = 0; i < 256; i++) { - common = samples[i + 256] * clev + bias; - samples[i] = samples[i] * level + common + samples[i + 768] * slev; - samples[i + 256] = (samples[i + 512] * level + common + - samples[i + 1024] * slev); - } -} - -static void mix21toS (sample_t * samples, - sample_t level, sample_t level3db, sample_t bias) -{ - int i; - sample_t surround; - - for (i = 0; i < 256; i++) { - surround = samples[i + 512] * level3db; - samples[i] = samples[i] * level - surround + bias; - samples[i + 256] = samples[i + 256] * level + surround + bias; - } -} - -static void mix22toS (sample_t * samples, - sample_t level, sample_t level3db, sample_t bias) -{ - int i; - sample_t surround; - - for (i = 0; i < 256; i++) { - surround = (samples[i + 512] + samples[i + 768]) * level3db; - samples[i] = samples[i] * level - surround + bias; - samples[i + 256] = samples[i + 256] * level + surround + bias; - } -} - -static void mix31toS (sample_t * samples, - sample_t level, sample_t level3db, sample_t bias) -{ - int i; - sample_t common, surround; - - for (i = 0; i < 256; i++) { - common = samples[i + 256] * level3db + bias; - surround = samples[i + 768] * level3db; - samples[i] = samples[i] * level + common - surround; - samples[i + 256] = samples[i + 512] * level + common + surround; - } -} - -static void mix32toS (sample_t * samples, - sample_t level, sample_t level3db, sample_t bias) -{ - int i; - sample_t common, surround; - - for (i = 0; i < 256; i++) { - common = samples[i + 256] * level3db + bias; - surround = (samples[i + 768] + samples[i + 1024]) * level3db; - samples[i] = samples[i] * level + common - surround; - samples[i + 256] = samples[i + 512] * level + common + surround; - } -} - -void downmix (sample_t * samples, int acmod, int output, - sample_t level, sample_t bias, sample_t clev, sample_t slev) -{ - switch (CONVERT (acmod, output & AC3_CHANNEL_MASK)) { - - case CONVERT (AC3_3F2R, AC3_3F2R): - mix1to1 (samples + 1024, level, bias); - case CONVERT (AC3_3F1R, AC3_3F1R): - case CONVERT (AC3_2F2R, AC3_2F2R): - mix1to1 (samples + 768, level, bias); - case CONVERT (AC3_3F, AC3_3F): - case CONVERT (AC3_2F1R, AC3_2F1R): - mix_3to3: - mix1to1 (samples + 512, level, bias); - case CONVERT (AC3_CHANNEL, AC3_CHANNEL): - case CONVERT (AC3_STEREO, AC3_STEREO): - case CONVERT (AC3_STEREO, AC3_DOLBY): - mix_2to2: - mix1to1 (samples + 256, level, bias); - case CONVERT (AC3_CHANNEL, AC3_CHANNEL1): - case CONVERT (AC3_MONO, AC3_MONO): - mix1to1 (samples, level, bias); - break; - - case CONVERT (AC3_CHANNEL, AC3_CHANNEL2): - move1to1 (samples + 256, samples, level, bias); - break; - - case CONVERT (AC3_STEREO, AC3_MONO): - mix_2to1: - mix2to1 (samples, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_2F1R, AC3_MONO): - if (slev == 0) - goto mix_2to1; - mix21to1 (samples, level * LEVEL_3DB, level * slev * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_2F2R, AC3_MONO): - if (slev == 0) - goto mix_2to1; - mix22to1 (samples, level * LEVEL_3DB, level * slev * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_3F, AC3_MONO): - mix_3to1: - mix3to1 (samples, level * LEVEL_3DB, level * clev * LEVEL_PLUS3DB, - bias); - break; - - case CONVERT (AC3_3F1R, AC3_MONO): - if (slev == 0) - goto mix_3to1; - mix31to1 (samples, level * LEVEL_3DB, level * clev * LEVEL_PLUS3DB, - level * slev * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_3F2R, AC3_MONO): - if (slev == 0) - goto mix_3to1; - mix32to1 (samples, level * LEVEL_3DB, level * clev * LEVEL_PLUS3DB, - level * slev * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_CHANNEL, AC3_MONO): - mix2to1 (samples, level * LEVEL_6DB, bias); - break; - - case CONVERT (AC3_MONO, AC3_DOLBY): - mix1to2 (samples, samples + 256, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_3F, AC3_DOLBY): - clev = LEVEL_3DB; - case CONVERT (AC3_3F, AC3_STEREO): - mix_3to2: - mix3to2 (samples, level, level * clev, bias); - break; - - case CONVERT (AC3_2F1R, AC3_DOLBY): - mix21toS (samples, level, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_3F1R, AC3_DOLBY): - mix31toS (samples, level, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_2F2R, AC3_DOLBY): - mix22toS (samples, level, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_3F2R, AC3_DOLBY): - mix32toS (samples, level, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_2F1R, AC3_STEREO): - if (slev == 0) - goto mix_2to2; - mix21to2 (samples, samples + 256, level, level * slev * LEVEL_3DB, - bias); - break; - - case CONVERT (AC3_3F1R, AC3_STEREO): - if (slev == 0) - goto mix_3to2; - mix31to2 (samples, level, level * clev, level * slev * LEVEL_3DB, - bias); - break; - - case CONVERT (AC3_2F2R, AC3_STEREO): - if (slev == 0) - goto mix_2to2; - mix11to1 (samples, samples + 512, level, level * slev, bias); - mix11to1 (samples + 256, samples + 768, level, level * slev, bias); - break; - - case CONVERT (AC3_3F2R, AC3_STEREO): - if (slev == 0) - goto mix_3to2; - mix32to2 (samples, level, level * clev, level * slev, bias); - break; - - case CONVERT (AC3_3F1R, AC3_3F): - if (slev == 0) - goto mix_3to3; - mix21to2 (samples, samples + 512, level, level * slev * LEVEL_3DB, - bias); - - case CONVERT (AC3_3F2R, AC3_3F): - if (slev == 0) - goto mix_3to3; - mix11to1 (samples, samples + 768, level, level * slev, bias); - mix11to1 (samples + 512, samples + 1024, level, level * slev, bias); - mix1to1 (samples + 256, level, bias); - break; - - case CONVERT (AC3_2F1R, AC3_2F2R): - mix1to2 (samples + 512, samples + 768, level * LEVEL_3DB, bias); - goto mix_2to2; - - case CONVERT (AC3_3F1R, AC3_3F2R): - mix1to2 (samples + 768, samples + 1024, level * LEVEL_3DB, bias); - goto mix_3to3; - - case CONVERT (AC3_2F2R, AC3_2F1R): - mix2to1 (samples + 512, level * LEVEL_3DB, bias); - goto mix_2to2; - - case CONVERT (AC3_3F2R, AC3_3F1R): - mix2to1 (samples + 768, level * LEVEL_3DB, bias); - goto mix_3to3; - - case CONVERT (AC3_3F1R, AC3_2F2R): - mix3to2 (samples, level, level * clev, bias); - mix1to2 (samples + 768, samples + 512, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_3F1R, AC3_2F1R): - mix3to2 (samples, level, level * clev, bias); - move1to1 (samples + 768, samples + 512, level, bias); - break; - - case CONVERT (AC3_3F2R, AC3_2F1R): - mix3to2 (samples, level, level * clev, bias); - move2to1 (samples + 768, samples + 512, level * LEVEL_3DB, bias); - break; - - case CONVERT (AC3_3F2R, AC3_2F2R): - mix3to2 (samples, level, level * clev, bias); - move1to1 (samples + 768, samples + 512, level, bias); - move1to1 (samples + 1024, samples + 768, level, bias); - break; - } - - if (output & AC3_LFE) - mix1to1 (samples - 256, level, bias); -} diff --git a/src/libac3/imdct_mlib.c b/src/libac3/imdct_mlib.c deleted file mode 100644 index 1ba783359..000000000 --- a/src/libac3/imdct_mlib.c +++ /dev/null @@ -1,142 +0,0 @@ -/* - * imdct_mlib.c - * Copyright (C) 2001 Håkan Hjort <d95hjort@dtek.chalmers.se> - * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. - */ - - -#include "config.h" - -#ifdef LIBAC3_MLIB - -#include <inttypes.h> -#include <mlib_types.h> -#include <mlib_status.h> -#include <mlib_signal.h> - -#include "ac3.h" -#include "ac3_internal.h" - -extern sample_t imdct_window[]; - -void -imdct_do_512_mlib(sample_t data[], sample_t delay[]) -{ - sample_t *buf_real; - sample_t *buf_imag; - sample_t *data_ptr; - sample_t *delay_ptr; - sample_t *window_ptr; - sample_t tmp[256] __attribute__ ((__aligned__ (16))); - int i; - - memcpy(tmp, data, 256 * sizeof(sample_t)); - if(mlib_SignalIMDCT_F32(tmp) != MLIB_SUCCESS) { - fprintf(stderr, "mediaLib failure\n"); - exit(-1); - } - - buf_real = tmp; - buf_imag = tmp + 128; - data_ptr = data; - delay_ptr = delay; - window_ptr = imdct_window; - - /* Window and convert to real valued signal */ - for(i=0; i< 64; i++) - { - *data_ptr++ = 2.0f * (-buf_imag[64+i] * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf_real[64-i-1] * *window_ptr++ + *delay_ptr++); - } - - for(i=0; i< 64; i++) - { - *data_ptr++ = 2.0f * (-buf_real[i] * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf_imag[128-i-1] * *window_ptr++ + *delay_ptr++); - } - - /* The trailing edge of the window goes into the delay line */ - delay_ptr = delay; - - for(i=0; i< 64; i++) - { - *delay_ptr++ = -buf_real[64+i] * *--window_ptr; - *delay_ptr++ = buf_imag[64-i-1] * *--window_ptr; - } - - for(i=0; i<64; i++) - { - *delay_ptr++ = buf_imag[i] * *--window_ptr; - *delay_ptr++ = -buf_real[128-i-1] * *--window_ptr; - } -} - -void -imdct_do_256_mlib(sample_t data[], sample_t delay[]) -{ - sample_t *buf1_real, *buf1_imag; - sample_t *buf2_real, *buf2_imag; - sample_t *data_ptr; - sample_t *delay_ptr; - sample_t *window_ptr; - sample_t tmp[256] __attribute__ ((__aligned__ (16))); - int i; - - memcpy(tmp, data, 256 * sizeof(sample_t)); - if(mlib_SignalIMDCT_F32(tmp) != MLIB_SUCCESS) { - fprintf(stderr, "mediaLib failure\n"); - exit(-1); - } - - buf1_real = tmp; - buf1_imag = tmp + 64; - buf2_real = tmp + 128; - buf2_imag = tmp + 128 + 64; - data_ptr = data; - delay_ptr = delay; - window_ptr = imdct_window; - - /* Window and convert to real valued signal */ - for(i=0; i< 64; i++) - { - *data_ptr++ = 2.0f * (-buf1_imag[i] * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf1_real[64-i-1] * *window_ptr++ + *delay_ptr++); - } - - for(i=0; i< 64; i++) - { - *data_ptr++ = 2.0f * (-buf1_real[i] * *window_ptr++ + *delay_ptr++); - *data_ptr++ = 2.0f * ( buf1_imag[64-i-1] * *window_ptr++ + *delay_ptr++); - } - - delay_ptr = delay; - - for(i=0; i< 64; i++) - { - *delay_ptr++ = -buf2_real[i] * *--window_ptr; - *delay_ptr++ = buf2_imag[64-i-1] * *--window_ptr; - } - - for(i=0; i< 64; i++) - { - *delay_ptr++ = buf2_imag[i] * *--window_ptr; - *delay_ptr++ = -buf2_real[64-i-1] * *--window_ptr; - } -} - -#endif diff --git a/src/libac3/parse.c b/src/libac3/parse.c deleted file mode 100644 index 7d7d7b3d0..000000000 --- a/src/libac3/parse.c +++ /dev/null @@ -1,689 +0,0 @@ -/* - * parse.c - * - * Copyright (C) Aaron Holtzman - May 1999 - * - * This file is part of ac3dec, a free Dolby AC-3 stream decoder. - * - * ac3dec 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, or (at your option) - * any later version. - * - * ac3dec 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 GNU Make; see the file COPYING. If not, write to - * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. - * - * - */ - -#include "config.h" - -#include <inttypes.h> -#include <string.h> - -#include "ac3.h" -#include "ac3_internal.h" -#include "bitstream.h" -#include "tables.h" - -void ac3_init (void) -{ - imdct_init (); -} - -static uint8_t halfrate[12] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 2, 3}; - -int ac3_syncinfo (uint8_t * buf, int * flags, - int * sample_rate, int * bit_rate) -{ - static int rate[] = { 32, 40, 48, 56, 64, 80, 96, 112, - 128, 160, 192, 224, 256, 320, 384, 448, - 512, 576, 640}; - static uint8_t lfeon[8] = {0x10, 0x10, 0x04, 0x04, 0x04, 0x01, 0x04, 0x01}; - int frmsizecod; - int bitrate; - int half; - int acmod; - - if ((buf[0] != 0x0b) || (buf[1] != 0x77)) // syncword - return 0; - - if (buf[5] >= 0x60) // bsid >= 12 - return 0; - half = halfrate[buf[5] >> 3]; - - // acmod, dsurmod and lfeon - acmod = buf[6] >> 5; - *flags = ((((buf[6] & 0xf8) == 0x50) ? AC3_DOLBY : acmod) | - ((buf[6] & lfeon[acmod]) ? AC3_LFE : 0)); - - frmsizecod = buf[4] & 63; - if (frmsizecod >= 38) - return 0; - bitrate = rate [frmsizecod >> 1]; - *bit_rate = (bitrate * 1000) >> half; - - switch (buf[4] & 0xc0) { - case 0: // 48 KHz - *sample_rate = 48000 >> half; - return 4 * bitrate; - case 0x40: - *sample_rate = 44100 >> half; - return 2 * (320 * bitrate / 147 + (frmsizecod & 1)); - case 0x80: - *sample_rate = 32000 >> half; - return 6 * bitrate; - default: - return 0; - } -} - -int ac3_frame (ac3_state_t * state, uint8_t * buf, int * flags, - sample_t * level, sample_t bias, sample_t * delay) -{ - static sample_t clev[4] = {LEVEL_3DB, LEVEL_45DB, LEVEL_6DB, LEVEL_45DB}; - static sample_t slev[4] = {LEVEL_3DB, LEVEL_6DB, 0, LEVEL_6DB}; - int chaninfo; - int acmod; - - state->fscod = buf[4] >> 6; - state->halfrate = halfrate[buf[5] >> 3]; - state->acmod = acmod = buf[6] >> 5; - - bitstream_set_ptr (buf + 6); - bitstream_get (3); // skip acmod we already parsed - - if ((acmod == 2) && (bitstream_get (2) == 2)) // dsurmod - acmod = AC3_DOLBY; - - if ((acmod & 1) && (acmod != 1)) - state->clev = clev[bitstream_get (2)]; // cmixlev - - if (acmod & 4) - state->slev = slev[bitstream_get (2)]; // surmixlev - - state->lfeon = bitstream_get (1); - - state->output = downmix_init (acmod, *flags, level, - state->clev, state->slev); - if (state->output < 0) - return 1; - if (state->lfeon && (*flags & AC3_LFE)) { - state->output |= AC3_LFE; - delay += 256; - } - *flags = state->output; - state->level = *level; - state->bias = bias; - state->delay = delay; - - chaninfo = !acmod; - do { - bitstream_get (5); // dialnorm - if (bitstream_get (1)) // compre - bitstream_get (8); // compr - if (bitstream_get (1)) // langcode - bitstream_get (8); // langcod - if (bitstream_get (1)) // audprodie - bitstream_get (7); // mixlevel + roomtyp - } while (chaninfo--); - - bitstream_get (2); // copyrightb + origbs - - if (bitstream_get (1)) // timecod1e - bitstream_get (14); // timecod1 - if (bitstream_get (1)) // timecod2e - bitstream_get (14); // timecod2 - - if (bitstream_get (1)) { // addbsie - int addbsil; - - addbsil = bitstream_get (6); - do { - bitstream_get (8); // addbsi - } while (addbsil--); - } - - return 0; -} - -static int parse_exponents (int expstr, int ngrps, uint8_t exponent, - uint8_t * dest) -{ - int exps; - - while (ngrps--) { - exps = bitstream_get (7); - - exponent += exp_1[exps]; - if (exponent > 24) - return 1; - - switch (expstr) { - case EXP_D45: - *(dest++) = exponent; - *(dest++) = exponent; - case EXP_D25: - *(dest++) = exponent; - case EXP_D15: - *(dest++) = exponent; - } - - exponent += exp_2[exps]; - if (exponent > 24) - return 1; - - switch (expstr) { - case EXP_D45: - *(dest++) = exponent; - *(dest++) = exponent; - case EXP_D25: - *(dest++) = exponent; - case EXP_D15: - *(dest++) = exponent; - } - - exponent += exp_3[exps]; - if (exponent > 24) - return 1; - - switch (expstr) { - case EXP_D45: - *(dest++) = exponent; - *(dest++) = exponent; - case EXP_D25: - *(dest++) = exponent; - case EXP_D15: - *(dest++) = exponent; - } - } - - return 0; -} - -static int parse_deltba (int8_t * deltba) -{ - int deltnseg, deltlen, delta, j; - - memset (deltba, 0, 50); - - deltnseg = bitstream_get (3); - j = 0; - do { - j += bitstream_get (5); - deltlen = bitstream_get (4); - delta = bitstream_get (3); - delta -= (delta >= 4) ? 3 : 4; - if (!deltlen) - continue; - if (j + deltlen >= 50) - return 1; - while (deltlen--) - deltba[j++] = delta; - } while (deltnseg--); - - return 0; -} - -static inline int zero_snr_offsets (int nfchans, ac3_state_t * state) -{ - int i; - - if ((state->csnroffst) || (state->cplinu && state->cplba.fsnroffst) || - (state->lfeon && state->lfeba.fsnroffst)) - return 0; - for (i = 0; i < nfchans; i++) - if (state->ba[i].fsnroffst) - return 0; - return 1; -} - -static sample_t q_1[2]; -static sample_t q_2[2]; -static sample_t q_4; -static int q_1_pointer; -static int q_2_pointer; -static int q_4_pointer; - -#define GET_COEFF(COEFF,DITHER) \ - switch (bap[i]) { \ - case 0: \ - DITHER (scale_factor[exp[i]]); \ - \ - case -1: \ - if (q_1_pointer >= 0) { \ - COEFF (q_1[q_1_pointer--] * scale_factor[exp[i]]); \ - } else { \ - int code; \ - \ - code = bitstream_get (5); \ - \ - q_1_pointer = 1; \ - q_1[0] = q_1_2[code]; \ - q_1[1] = q_1_1[code]; \ - COEFF (q_1_0[code] * scale_factor[exp[i]]); \ - } \ - \ - case -2: \ - if (q_2_pointer >= 0) { \ - COEFF (q_2[q_2_pointer--] * scale_factor[exp[i]]); \ - } else { \ - int code; \ - \ - code = bitstream_get (7); \ - \ - q_2_pointer = 1; \ - q_2[0] = q_2_2[code]; \ - q_2[1] = q_2_1[code]; \ - COEFF (q_2_0[code] * scale_factor[exp[i]]); \ - } \ - \ - case 3: \ - COEFF (q_3[bitstream_get (3)] * scale_factor[exp[i]]); \ - \ - case -3: \ - if (q_4_pointer == 0) { \ - q_4_pointer = -1; \ - COEFF (q_4 * scale_factor[exp[i]]); \ - } else { \ - int code; \ - \ - code = bitstream_get (7); \ - \ - q_4_pointer = 0; \ - q_4 = q_4_1[code]; \ - COEFF (q_4_0[code] * scale_factor[exp[i]]); \ - } \ - \ - case 4: \ - COEFF (q_5[bitstream_get (4)] * scale_factor[exp[i]]); \ - \ - default: \ - COEFF (((int16_t)(bitstream_get(bap[i]) << (16 - bap[i]))) * \ - scale_factor[exp[i]]); \ - } - -#define CHANNEL_COEFF(val) \ - coeff[i] = val; \ - i++; \ - continue; - -#define CHANNEL_DITHER(val) \ - if (dither) { \ - coeff[i] = dither_gen () * val; \ - i++; \ - continue; \ - } else { \ - coeff[i] = 0; \ - i++; \ - continue; \ - } - -static uint16_t lfsr_state = 1; - -static inline int16_t dither_gen(void) -{ - int16_t state; - - state = dither_lut[lfsr_state >> 8] ^ (lfsr_state << 8); - - lfsr_state = (uint16_t) state; - - return ((state * (int) (LEVEL_3DB * 256)) >> 8); -} - -static void coeff_get (sample_t * coeff, uint8_t * exp, int8_t * bap, - int dither, int end) -{ - int i; - - i = 0; - while (i < end) - GET_COEFF (CHANNEL_COEFF, CHANNEL_DITHER); -} - -#define COUPLING_COEFF(val) \ - cplcoeff = val; \ - break; - -#define COUPLING_DITHER(val) \ - cplcoeff = val; \ - for (ch = 0; ch < nfchans; ch++) \ - if (state->chincpl[ch]) { \ - if (dithflag[ch]) \ - samples[ch][i] = \ - state->cplco[ch][bnd] * dither_gen () * cplcoeff; \ - else \ - samples[ch][i] = 0; \ - } \ - i++; \ - continue; - -int ac3_block (ac3_state_t * state, sample_t samples[][256]) -{ - static const uint8_t nfchans_tbl[8] = {2, 1, 2, 3, 3, 4, 4, 5}; - static int rematrix_band[4] = {25, 37, 61, 253}; - int i, nfchans, chaninfo; - uint8_t cplexpstr, chexpstr[5], lfeexpstr, do_bit_alloc, done_cpl; - uint8_t blksw[5], dithflag[5]; - - nfchans = nfchans_tbl[state->acmod]; - - for (i = 0; i < nfchans; i++) - blksw[i] = bitstream_get (1); - - for (i = 0; i < nfchans; i++) - dithflag[i] = bitstream_get (1); - - chaninfo = !(state->acmod); - do { - if (bitstream_get (1)) // dynrnge - bitstream_get (8); // dynrng - } while (chaninfo--); - - if (bitstream_get (1)) { // cplstre - state->cplinu = bitstream_get (1); - if (state->cplinu) { - static int bndtab[16] = {31, 35, 37, 39, 41, 42, 43, 44, - 45, 45, 46, 46, 47, 47, 48, 48}; - int cplbegf; - int cplendf; - int ncplsubnd; - - for (i = 0; i < nfchans; i++) - state->chincpl[i] = bitstream_get (1); - switch (state->acmod) { - case 0: case 1: - return 1; - case 2: - state->phsflginu = bitstream_get (1); - } - cplbegf = bitstream_get (4); - cplendf = bitstream_get (4); - - if (cplendf + 3 - cplbegf < 0) - return 1; - state->ncplbnd = ncplsubnd = cplendf + 3 - cplbegf; - state->cplstrtbnd = bndtab[cplbegf]; - state->cplstrtmant = cplbegf * 12 + 37; - state->cplendmant = cplendf * 12 + 73; - - for (i = 0; i < ncplsubnd - 1; i++) { - state->cplbndstrc[i] = bitstream_get (1); - state->ncplbnd -= state->cplbndstrc[i]; - } - state->cplbndstrc[i] = 0; // last value is a sentinel - } - } - - if (state->cplinu) { - int j, cplcoe; - - cplcoe = 0; - for (i = 0; i < nfchans; i++) - if (state->chincpl[i]) - if (bitstream_get (1)) { // cplcoe - int mstrcplco, cplcoexp, cplcomant; - - cplcoe = 1; - mstrcplco = 3 * bitstream_get (2); - for (j = 0; j < state->ncplbnd; j++) { - cplcoexp = bitstream_get (4); - cplcomant = bitstream_get (4); - if (cplcoexp == 15) - cplcomant <<= 14; - else - cplcomant = (cplcomant | 0x10) << 13; - state->cplco[i][j] = - cplcomant * scale_factor[cplcoexp + mstrcplco]; - } - } - if ((state->acmod == 2) && state->phsflginu && cplcoe) - for (j = 0; j < state->ncplbnd; j++) - if (bitstream_get (1)) // phsflg - state->cplco[1][j] = -state->cplco[1][j]; - } - - if ((state->acmod == 2) && (bitstream_get (1))) { // rematstr - int end; - - end = (state->cplinu) ? state->cplstrtmant : 253; - i = 0; - do - state->rematflg[i] = bitstream_get (1); - while (rematrix_band[i++] < end); - } - - cplexpstr = EXP_REUSE; - lfeexpstr = EXP_REUSE; - if (state->cplinu) - cplexpstr = bitstream_get (2); - for (i = 0; i < nfchans; i++) - chexpstr[i] = bitstream_get (2); - if (state->lfeon) - lfeexpstr = bitstream_get (1); - - for (i = 0; i < nfchans; i++) - if (chexpstr[i] != EXP_REUSE) { - if (state->cplinu && state->chincpl[i]) - state->endmant[i] = state->cplstrtmant; - else { - int chbwcod; - - chbwcod = bitstream_get (6); - if (chbwcod > 60) - return 1; - state->endmant[i] = chbwcod * 3 + 73; - } - } - - do_bit_alloc = 0; - - if (cplexpstr != EXP_REUSE) { - int cplabsexp, ncplgrps; - - do_bit_alloc = 1; - ncplgrps = ((state->cplendmant - state->cplstrtmant) / - (3 << (cplexpstr - 1))); - cplabsexp = bitstream_get (4) << 1; - if (parse_exponents (cplexpstr, ncplgrps, cplabsexp, - state->cpl_exp + state->cplstrtmant)) - return 1; - } - for (i = 0; i < nfchans; i++) - if (chexpstr[i] != EXP_REUSE) { - int grp_size, nchgrps; - - do_bit_alloc = 1; - grp_size = 3 << (chexpstr[i] - 1); - nchgrps = (state->endmant[i] + grp_size - 4) / grp_size; - state->fbw_exp[i][0] = bitstream_get (4); - if (parse_exponents (chexpstr[i], nchgrps, state->fbw_exp[i][0], - state->fbw_exp[i] + 1)) - return 1; - bitstream_get (2); // gainrng - } - if (lfeexpstr != EXP_REUSE) { - do_bit_alloc = 1; - state->lfe_exp[0] = bitstream_get (4); - if (parse_exponents (lfeexpstr, 2, state->lfe_exp[0], - state->lfe_exp + 1)) - return 1; - } - - if (bitstream_get (1)) { // baie - do_bit_alloc = 1; - state->sdcycod = bitstream_get (2); - state->fdcycod = bitstream_get (2); - state->sgaincod = bitstream_get (2); - state->dbpbcod = bitstream_get (2); - state->floorcod = bitstream_get (3); - } - if (bitstream_get (1)) { //snroffste - do_bit_alloc = 1; - state->csnroffst = bitstream_get (6); - if (state->cplinu) { - state->cplba.fsnroffst = bitstream_get (4); - state->cplba.fgaincod = bitstream_get (3); - } - for (i = 0; i < nfchans; i++) { - state->ba[i].fsnroffst = bitstream_get (4); - state->ba[i].fgaincod = bitstream_get (3); - } - if (state->lfeon) { - state->lfeba.fsnroffst = bitstream_get (4); - state->lfeba.fgaincod = bitstream_get (3); - } - } - if ((state->cplinu) && (bitstream_get (1))) { // cplleake - do_bit_alloc = 1; - state->cplfleak = 2304 - (bitstream_get (3) << 8); - state->cplsleak = 2304 - (bitstream_get (3) << 8); - } - - if (bitstream_get (1)) { // deltbaie - do_bit_alloc = 1; - if (state->cplinu) - state->cplba.deltbae = bitstream_get (2); - for (i = 0; i < nfchans; i++) - state->ba[i].deltbae = bitstream_get (2); - if (state->cplinu && (state->cplba.deltbae == DELTA_BIT_NEW) && - parse_deltba (state->cplba.deltba)) - return 1; - for (i = 0; i < nfchans; i++) - if ((state->ba[i].deltbae == DELTA_BIT_NEW) && - parse_deltba (state->ba[i].deltba)) - return 1; - } - - if (do_bit_alloc) { - if (zero_snr_offsets (nfchans, state)) { - memset (state->cpl_bap, 0, sizeof (state->cpl_bap)); - memset (state->fbw_bap, 0, sizeof (state->fbw_bap)); - memset (state->lfe_bap, 0, sizeof (state->lfe_bap)); - } else { - if (state->cplinu) - bit_allocate (state, &state->cplba, state->cplstrtbnd, - state->cplstrtmant, state->cplendmant, - state->cplfleak, state->cplsleak, - state->cpl_exp, state->cpl_bap); - for (i = 0; i < nfchans; i++) - bit_allocate (state, state->ba + i, 0, 0, state->endmant[i], - 0, 0, state->fbw_exp[i], state->fbw_bap[i]); - if (state->lfeon) { - state->lfeba.deltbae = DELTA_BIT_NONE; - bit_allocate (state, &state->lfeba, 0, 0, 7, 0, 0, - state->lfe_exp, state->lfe_bap); - } - } - } - - if (bitstream_get (1)) { // skiple - i = bitstream_get (9); // skipl - while (i--) - bitstream_get (8); - } - - if (state->output & AC3_LFE) - samples++; // shift for LFE channel - - q_1_pointer = q_2_pointer = q_4_pointer = -1; - done_cpl = 0; - - for (i = 0; i < nfchans; i++) { - int j; - - coeff_get (samples[i], state->fbw_exp[i], state->fbw_bap[i], - dithflag[i], state->endmant[i]); - - if (state->cplinu && state->chincpl[i]) { - if (!done_cpl) { - int i, i_end, bnd, sub_bnd, ch; - sample_t cplcoeff; - - done_cpl = 1; - -#define bap state->cpl_bap -#define exp state->cpl_exp - - sub_bnd = bnd = 0; - i = state->cplstrtmant; - while (i < state->cplendmant) { - i_end = i + 12; - while (state->cplbndstrc[sub_bnd++]) - i_end += 12; - - while (i < i_end) { - GET_COEFF (COUPLING_COEFF, COUPLING_DITHER); - for (ch = 0; ch < nfchans; ch++) - if (state->chincpl[ch]) - samples[ch][i] = - state->cplco[ch][bnd] * cplcoeff; - i++; - } - bnd++; - } - -#undef bap -#undef exp - } - j = state->cplendmant; - } else - j = state->endmant[i]; - for (; j < 256; j++) - samples[i][j] = 0; - } - - if (state->acmod == 2) { - int j, end, band; - - end = ((state->endmant[0] < state->endmant[1]) ? - state->endmant[0] : state->endmant[1]); - - i = 0; - j = 13; - do { - if (!state->rematflg[i]) { - j = rematrix_band[i++]; - continue; - } - band = rematrix_band[i++]; - if (band > end) - band = end; - do { - sample_t tmp0, tmp1; - - tmp0 = samples[0][j]; - tmp1 = samples[1][j]; - samples[0][j] = tmp0 + tmp1; - samples[1][j] = tmp0 - tmp1; - } while (++j < band); - } while (j < end); - } - - if (state->lfeon) { - coeff_get (samples[-1], state->lfe_exp, state->lfe_bap, 0, 7); - if (state->output & AC3_LFE) { - for (i = 7; i < 256; i++) - samples[-1][i] = 0; - imdct_512 (samples[-1], state->delay - 256); - } - } - - for (i = 0; i < nfchans; i++) - if (blksw[i]) - imdct_256 (samples[i], state->delay + i * 256); - else - imdct_512 (samples[i], state->delay + i * 256); - - downmix (*samples, state->acmod, state->output, state->level, state->bias, - state->clev, state->slev); - - return 0; -} diff --git a/src/libac3/xine_decoder.c b/src/libac3/xine_decoder.c deleted file mode 100644 index 7b7ee8794..000000000 --- a/src/libac3/xine_decoder.c +++ /dev/null @@ -1,474 +0,0 @@ -/* - * Copyright (C) 2000-2001 the xine project - * - * This file is part of xine, a unix video player. - * - * xine 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. - * - * xine is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA - * - * $Id: xine_decoder.c,v 1.24 2001/08/28 19:16:19 guenter Exp $ - * - * stuff needed to turn libac3 into a xine decoder plugin - */ - - -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <sys/types.h> -#include <sys/stat.h> -#include <fcntl.h> - -#include "audio_out.h" -#include "ac3.h" -#include "ac3_internal.h" -#include "buffer.h" -#include "xine_internal.h" - -#define FRAME_SIZE 4096+512 - -#undef DEBUG_AC3 -#ifdef DEBUG_AC3 -int ac3file; -#endif - -typedef struct ac3dec_decoder_s { - audio_decoder_t audio_decoder; - - uint32_t pts; - uint32_t last_pts; - - uint8_t frame_buffer[FRAME_SIZE]; - uint8_t *frame_ptr; - int sync_todo; - int frame_length, frame_todo; - uint16_t syncword; - - ac3_state_t ac3_state; - int ac3_flags; - int ac3_bit_rate; - int ac3_sample_rate; - float ac3_level; - int have_lfe; - - int ac3_flags_map[8]; - int ao_flags_map[8]; - - int16_t int_samples [6 * 256 * 6]; - sample_t delay[6*256]; - sample_t samples[6][256]; - - ao_instance_t *audio_out; - int audio_caps; - int bypass_mode; - int output_sampling_rate; - int output_open; - int output_mode; - -} ac3dec_decoder_t; - -int ac3dec_can_handle (audio_decoder_t *this_gen, int buf_type) { - return ((buf_type & 0xFFFF0000) == BUF_AUDIO_A52) ; -} - - -void ac3dec_init (audio_decoder_t *this_gen, ao_instance_t *audio_out) { - - ac3dec_decoder_t *this = (ac3dec_decoder_t *) this_gen; - /* int i; */ - - this->audio_out = audio_out; - this->audio_caps = audio_out->get_capabilities(audio_out); - this->syncword = 0; - this->sync_todo = 7; - this->output_open = 0; - this->pts = 0; - this->last_pts = 0; - - ac3_init (); - - /* - * find out if this driver supports ac3 output - * or, if not, how many channels we've got - */ - - if (this->audio_caps & AO_CAP_MODE_AC3) - this->bypass_mode = 1; - else { - this->bypass_mode = 0; - - this->ac3_flags_map[AC3_MONO] = AC3_MONO; - this->ac3_flags_map[AC3_STEREO] = AC3_STEREO; - this->ac3_flags_map[AC3_3F] = AC3_STEREO; - this->ac3_flags_map[AC3_2F1R] = AC3_STEREO; - this->ac3_flags_map[AC3_3F1R] = AC3_STEREO; - this->ac3_flags_map[AC3_2F2R] = AC3_STEREO; - this->ac3_flags_map[AC3_3F2R] = AC3_STEREO; - - this->ao_flags_map[AC3_MONO] = AO_CAP_MODE_MONO; - this->ao_flags_map[AC3_STEREO] = AO_CAP_MODE_STEREO; - this->ao_flags_map[AC3_3F] = AO_CAP_MODE_STEREO; - this->ao_flags_map[AC3_2F1R] = AO_CAP_MODE_STEREO; - this->ao_flags_map[AC3_3F1R] = AO_CAP_MODE_STEREO; - this->ao_flags_map[AC3_2F2R] = AO_CAP_MODE_STEREO; - this->ao_flags_map[AC3_3F2R] = AO_CAP_MODE_STEREO; - - /* find best mode */ - if (this->audio_caps & AO_CAP_MODE_5_1CHANNEL) { - - this->ac3_flags_map[AC3_2F2R] = AC3_2F2R; - this->ac3_flags_map[AC3_3F2R] = AC3_3F2R | AC3_LFE; - this->ao_flags_map[AC3_2F2R] = AO_CAP_MODE_4CHANNEL; - this->ao_flags_map[AC3_3F2R] = AO_CAP_MODE_5CHANNEL; - - } else if (this->audio_caps & AO_CAP_MODE_5CHANNEL) { - - this->ac3_flags_map[AC3_2F2R] = AC3_2F2R; - this->ac3_flags_map[AC3_3F2R] = AC3_3F2R; - this->ao_flags_map[AC3_2F2R] = AO_CAP_MODE_4CHANNEL; - this->ao_flags_map[AC3_3F2R] = AO_CAP_MODE_5CHANNEL; - - } else if (this->audio_caps & AO_CAP_MODE_4CHANNEL) { - - this->ac3_flags_map[AC3_2F2R] = AC3_2F2R; - this->ac3_flags_map[AC3_3F2R] = AC3_2F2R; - - this->ao_flags_map[AC3_2F2R] = AO_CAP_MODE_4CHANNEL; - this->ao_flags_map[AC3_3F2R] = AO_CAP_MODE_4CHANNEL; - - /* else if (this->audio_caps & AO_CAP_MODE_STEREO) - defaults are ok */ - } else if (!(this->audio_caps & AO_CAP_MODE_STEREO)) { - printf ("HELP! a mono-only audio driver?!\n"); - - this->ac3_flags_map[AC3_MONO] = AC3_MONO; - this->ac3_flags_map[AC3_STEREO] = AC3_MONO; - this->ac3_flags_map[AC3_3F] = AC3_MONO; - this->ac3_flags_map[AC3_2F1R] = AC3_MONO; - this->ac3_flags_map[AC3_3F1R] = AC3_MONO; - this->ac3_flags_map[AC3_2F2R] = AC3_MONO; - this->ac3_flags_map[AC3_3F2R] = AC3_MONO; - - this->ao_flags_map[AC3_MONO] = AO_CAP_MODE_MONO; - this->ao_flags_map[AC3_STEREO] = AO_CAP_MODE_MONO; - this->ao_flags_map[AC3_3F] = AO_CAP_MODE_MONO; - this->ao_flags_map[AC3_2F1R] = AO_CAP_MODE_MONO; - this->ao_flags_map[AC3_3F1R] = AO_CAP_MODE_MONO; - this->ao_flags_map[AC3_2F2R] = AO_CAP_MODE_MONO; - this->ao_flags_map[AC3_3F2R] = AO_CAP_MODE_MONO; - } - } - - /* - for (i = 0; i<8; i++) - this->ac3_flags_map[i] |= AC3_ADJUST_LEVEL; -*/ -#ifdef DEBUG_AC3 - ac3file = open ("test.ac3", O_CREAT | O_WRONLY | O_TRUNC, 0644); -#endif -} - -static inline int16_t blah (int32_t i) -{ - if (i > 0x43c07fff) - return 32767; - else if (i < 0x43bf8000) - return -32768; - else - return i - 0x43c00000; -} - -static inline void float_to_int (float * _f, int16_t * s16, int num_channels) { - int i; - int32_t * f = (int32_t *) _f; /* XXX assumes IEEE float format */ - - for (i = 0; i < 256; i++) { - s16[num_channels*i] = blah (f[i]); - } -} - -static void ac3dec_decode_frame (ac3dec_decoder_t *this, uint32_t pts) { - - int output_mode = AO_CAP_MODE_STEREO; - - /* - * do we want to decode this frame in software? - */ - - if (!this->bypass_mode) { - - int ac3_output_flags, i; - float level = this->ac3_level; - - /* - * oki, decode this frame in software - */ - - /* determine output mode */ - - ac3_output_flags = this->ac3_flags_map[this->ac3_flags & AC3_CHANNEL_MASK]; - - if (ac3_frame (&this->ac3_state, - this->frame_buffer, - &ac3_output_flags, - &level, 384, this->delay)) { - printf ("libac3: ac3_frame error\n"); - return; - } - - this->have_lfe = ac3_output_flags & AC3_LFE; - if (this->have_lfe) - output_mode = AO_CAP_MODE_5_1CHANNEL; - else - output_mode = this->ao_flags_map[ac3_output_flags]; - - /* - * (re-)open output device - */ - - if (!this->output_open - || (this->ac3_sample_rate != this->output_sampling_rate) - || (output_mode != this->output_mode)) { - - if (this->output_open) - this->audio_out->close (this->audio_out); - - - this->output_open = (this->audio_out->open (this->audio_out, 16, - this->ac3_sample_rate, - output_mode) == 1); - this->output_sampling_rate = this->ac3_sample_rate; - this->output_mode = output_mode; - } - - - if (!this->output_open) - return; - - - /* - * decode ac3 and convert/interleave samples - */ - - for (i = 0; i < 6; i++) { - if (ac3_block (&this->ac3_state, this->samples)) { - printf ("libac3: ac3_block error\n"); - return; - } - - switch (output_mode) { - case AO_CAP_MODE_MONO: - float_to_int (this->samples[0], this->int_samples+(i*256), 1); - break; - case AO_CAP_MODE_STEREO: - float_to_int (this->samples[0], this->int_samples+(i*256*2), 2); - float_to_int (this->samples[1], this->int_samples+(i*256*2)+1, 2); - break; - case AO_CAP_MODE_4CHANNEL: - float_to_int (this->samples[0], this->int_samples+(i*256*4), 4); /* L */ - float_to_int (this->samples[1], this->int_samples+(i*256*4)+1, 4); /* R */ - float_to_int (this->samples[2], this->int_samples+(i*256*4)+2, 4); /* RL */ - float_to_int (this->samples[3], this->int_samples+(i*256*4)+3, 4); /* RR */ - break; - case AO_CAP_MODE_5CHANNEL: - float_to_int (this->samples[0], this->int_samples+(i*256*5)+0, 5); /* L */ - float_to_int (this->samples[1], this->int_samples+(i*256*5)+4, 5); /* C */ - float_to_int (this->samples[2], this->int_samples+(i*256*5)+1, 5); /* R */ - float_to_int (this->samples[3], this->int_samples+(i*256*5)+2, 5); /* RL */ - float_to_int (this->samples[4], this->int_samples+(i*256*5)+3, 5); /* RR */ - break; - case AO_CAP_MODE_5_1CHANNEL: - float_to_int (this->samples[0], this->int_samples+(i*256*6)+5, 6); /* lfe */ - float_to_int (this->samples[1], this->int_samples+(i*256*6)+0, 6); /* L */ - float_to_int (this->samples[2], this->int_samples+(i*256*6)+4, 6); /* C */ - float_to_int (this->samples[3], this->int_samples+(i*256*6)+1, 6); /* R */ - float_to_int (this->samples[4], this->int_samples+(i*256*6)+2, 6); /* RL */ - float_to_int (this->samples[5], this->int_samples+(i*256*6)+3, 6); /* RR */ - break; - default: - printf ("libac3: help - unsupported mode %08x\n", output_mode); - } - } - - /* output decoded samples */ - - this->audio_out->write (this->audio_out, - this->int_samples, - 256*6, - pts); - pts = 0; - - } else { - - /* - * loop through ac3 data - */ - - if (!this->output_open) { - - int sample_rate, bit_rate, flags; - - ac3_syncinfo (this->frame_buffer, &flags, &sample_rate, &bit_rate); - - this->output_open = (this->audio_out->open (this->audio_out, 16, - sample_rate, - AO_CAP_MODE_AC3) == 1); - this->output_mode = AO_CAP_MODE_AC3; - } - - if (this->output_open) { - this->audio_out->write (this->audio_out, - (int16_t*)this->frame_buffer, - this->frame_length, - pts); - } - } -} - -void ac3dec_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { - - ac3dec_decoder_t *this = (ac3dec_decoder_t *) this_gen; - uint8_t *current = buf->content; - uint8_t *end = buf->content + buf->size; - uint8_t byte; - - if (buf->decoder_info[0] == 0) - return; - - /* - printf ("libac3: got buffer, pts =%d, pts - last_pts=%d\n", - buf->PTS, buf->PTS - this->last_pts); - - this->last_pts = buf->PTS; - */ - - if (buf->PTS) - this->pts = buf->PTS; - - - while (current != end) { - - if ( (this->sync_todo == 0) && (this->frame_todo == 0) ) { - ac3dec_decode_frame (this, this->pts); -#ifdef DEBUG_AC3 - write (ac3file, this->frame_buffer, this->frame_length); -#endif - this->pts = 0; - this->sync_todo = 7; - this->syncword = 0; - } - - while (1) { - byte = *current++; - - if (this->sync_todo>0) { - - /* search and collect syncinfo */ - - if (this->syncword != 0x0b77) { - this->syncword = (this->syncword << 8) | byte; - - if (this->syncword == 0x0b77) { - - this->frame_buffer[0] = 0x0b; - this->frame_buffer[1] = 0x77; - - this->sync_todo = 5; - this->frame_ptr = this->frame_buffer+2; - } - } else { - *this->frame_ptr++ = byte; - this->sync_todo--; - - if (this->sync_todo==0) { - - this->frame_length = ac3_syncinfo (this->frame_buffer, - &this->ac3_flags, - &this->ac3_sample_rate, - &this->ac3_bit_rate); - if (this->frame_length) { - this->frame_todo = this->frame_length - 7; - } else { - this->sync_todo = 7; - this->syncword = 0; - printf ("libac3: skip frame of zero length\n"); - } - - } - - } - } else { - - *this->frame_ptr++ = byte; - this->frame_todo--; - - if (this->frame_todo == 0) { - if (current == end) - return ; - break; - } - } - - if (current == end) - return ; - } - } -} - -void ac3dec_close (audio_decoder_t *this_gen) { - - ac3dec_decoder_t *this = (ac3dec_decoder_t *) this_gen; - - if (this->output_open) - this->audio_out->close (this->audio_out); - - this->output_open = 0; - -#ifdef DEBUG_AC3 - close (ac3file); -#endif -} - -static char *ac3dec_get_id(void) { - return "ac3dec"; -} - -audio_decoder_t *init_audio_decoder_plugin (int iface_version, config_values_t *cfg) { - - ac3dec_decoder_t *this ; - - if (iface_version != 2) { - printf( "libac3: plugin doesn't support plugin API version %d.\n" - "libac3: this means there's a version mismatch between xine and this " - "libac3: decoder plugin.\nInstalling current plugins should help.\n", - iface_version); - return NULL; - } - - this = (ac3dec_decoder_t *) malloc (sizeof (ac3dec_decoder_t)); - - this->audio_decoder.interface_version = 2; - this->audio_decoder.can_handle = ac3dec_can_handle; - this->audio_decoder.init = ac3dec_init; - this->audio_decoder.decode_data = ac3dec_decode_data; - this->audio_decoder.close = ac3dec_close; - this->audio_decoder.get_identifier = ac3dec_get_id; - this->audio_decoder.priority = 1; - - - this->ac3_level = (float) cfg->lookup_int (cfg, "ac3_level", 100) / 100.0; - - return (audio_decoder_t *) this; -} - diff --git a/src/xine-engine/audio_out.c b/src/xine-engine/audio_out.c index c080866cc..ab6695aa7 100644 --- a/src/xine-engine/audio_out.c +++ b/src/xine-engine/audio_out.c @@ -17,7 +17,7 @@ * along with self program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA * - * $Id: audio_out.c,v 1.8 2001/08/24 01:05:31 guenter Exp $ + * $Id: audio_out.c,v 1.9 2001/08/29 00:51:58 guenter Exp $ * * 22-8-2001 James imported some useful AC3 sections from the previous alsa driver. * (c) 2001 Andy Lo A Foe <andy@alsaplayer.org> @@ -176,7 +176,7 @@ static void ao_fill_gap (ao_instance_t *this, uint32_t pts_len) { num_frames = pts_len * this->frames_per_kpts / 1024; - if (this->mode == AO_CAP_MODE_AC3) return; /* FIXME */ + if (this->mode == AO_CAP_MODE_A52) return; /* FIXME */ printf ("audio_out: inserting %d 0-frames to fill a gap of %d pts\n",num_frames, pts_len); @@ -264,8 +264,8 @@ static int ao_write(ao_instance_t *this, else delay = 0; - /* External AC3 decoder delay correction */ - if (this->mode==AO_CAP_MODE_AC3) + /* External A52 decoder delay correction */ + if (this->mode==AO_CAP_MODE_A52) delay+=10; buffer_vpts += delay * 1024 / this->frames_per_kpts; @@ -298,13 +298,13 @@ static int ao_write(ao_instance_t *this, /* * resample and output frames */ - if (this->mode == AO_CAP_MODE_AC3) + if (this->mode == AO_CAP_MODE_A52) bDropPackage=0; if (!bDropPackage) { int num_output_frames = (double) num_frames * this->frame_rate_factor; - if ((!this->do_resample) && (this->mode != AO_CAP_MODE_AC3)) { + if ((!this->do_resample) && (this->mode != AO_CAP_MODE_A52)) { xprintf (VERBOSE|AUDIO, "audio_out: writing without resampling\n"); this->driver->write (this->driver, output_frames, num_output_frames ); @@ -334,7 +334,7 @@ static int ao_write(ao_instance_t *this, this->frame_buffer, num_output_frames); this->driver->write(this->driver, this->frame_buffer, num_output_frames); break; - case AO_CAP_MODE_AC3: + case AO_CAP_MODE_A52: num_output_frames = (num_frames+8)/4; this->frame_buffer[0] = 0xf872; /* spdif syncword */ this->frame_buffer[1] = 0x4e1f; /* ............. */ diff --git a/src/xine-engine/audio_out.h b/src/xine-engine/audio_out.h index 047529492..eee66ccff 100644 --- a/src/xine-engine/audio_out.h +++ b/src/xine-engine/audio_out.h @@ -17,7 +17,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA * - * $Id: audio_out.h,v 1.11 2001/08/24 01:05:31 guenter Exp $ + * $Id: audio_out.h,v 1.12 2001/08/29 00:51:58 guenter Exp $ */ #ifndef HAVE_AUDIO_OUT_H #define HAVE_AUDIO_OUT_H @@ -196,7 +196,7 @@ ao_instance_t *ao_new_instance (ao_driver_t *driver, metronom_t *metronom, confi */ #define AO_CAP_NOCAP 0x00000000 /* driver has no capabilities */ -#define AO_CAP_MODE_AC3 0x00000001 /* driver supports AC3 output */ +#define AO_CAP_MODE_A52 0x00000001 /* driver supports A/52 output */ #define AO_CAP_MODE_AC5 0x00000002 /* driver supports AC5 output */ /* 1 sample == 2 bytes (C) */ #define AO_CAP_MODE_MONO 0x00000004 /* driver supports mono output */ |