Mercurial > libavcodec.hg
view ac3dec.c @ 6323:e6da66f378c7 libavcodec
mpegvideo.h has two function declarations with the 'inline' specifier
but no definition for those functions. The C standard requires a
definition to appear in the same translation unit for any function
declared with 'inline'. Most of the files including mpegvideo.h do not
define those functions. Fix this by removing the 'inline' specifiers
from the header.
patch by Uoti Urpala
author | diego |
---|---|
date | Sun, 03 Feb 2008 17:54:30 +0000 |
parents | a35b838ab955 |
children | de7502093922 |
line wrap: on
line source
/* * AC-3 Audio Decoder * This code is developed as part of Google Summer of Code 2006 Program. * * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com). * Copyright (c) 2007 Justin Ruggles * * Portions of this code are derived from liba52 * http://liba52.sourceforge.net * Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org> * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <stdio.h> #include <stddef.h> #include <math.h> #include <string.h> #include "avcodec.h" #include "ac3_parser.h" #include "bitstream.h" #include "crc.h" #include "dsputil.h" #include "random.h" /** * Table of bin locations for rematrixing bands * reference: Section 7.5.2 Rematrixing : Frequency Band Definitions */ static const uint8_t rematrix_band_tab[5] = { 13, 25, 37, 61, 253 }; /** * table for exponent to scale_factor mapping * scale_factors[i] = 2 ^ -i */ static float scale_factors[25]; /** table for grouping exponents */ static uint8_t exp_ungroup_tab[128][3]; /** tables for ungrouping mantissas */ static float b1_mantissas[32][3]; static float b2_mantissas[128][3]; static float b3_mantissas[8]; static float b4_mantissas[128][2]; static float b5_mantissas[16]; /** * Quantization table: levels for symmetric. bits for asymmetric. * reference: Table 7.18 Mapping of bap to Quantizer */ static const uint8_t quantization_tab[16] = { 0, 3, 5, 7, 11, 15, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16 }; /** dynamic range table. converts codes to scale factors. */ static float dynamic_range_tab[256]; /** Adjustments in dB gain */ #define LEVEL_MINUS_3DB 0.7071067811865476 #define LEVEL_MINUS_4POINT5DB 0.5946035575013605 #define LEVEL_MINUS_6DB 0.5000000000000000 #define LEVEL_MINUS_9DB 0.3535533905932738 #define LEVEL_ZERO 0.0000000000000000 #define LEVEL_ONE 1.0000000000000000 static const float gain_levels[6] = { LEVEL_ZERO, LEVEL_ONE, LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB, LEVEL_MINUS_9DB }; /** * Table for center mix levels * reference: Section 5.4.2.4 cmixlev */ static const uint8_t center_levels[4] = { 2, 3, 4, 3 }; /** * Table for surround mix levels * reference: Section 5.4.2.5 surmixlev */ static const uint8_t surround_levels[4] = { 2, 4, 0, 4 }; /** * Table for default stereo downmixing coefficients * reference: Section 7.8.2 Downmixing Into Two Channels */ static const uint8_t ac3_default_coeffs[8][5][2] = { { { 1, 0 }, { 0, 1 }, }, { { 2, 2 }, }, { { 1, 0 }, { 0, 1 }, }, { { 1, 0 }, { 3, 3 }, { 0, 1 }, }, { { 1, 0 }, { 0, 1 }, { 4, 4 }, }, { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 5, 5 }, }, { { 1, 0 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, }, { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, }, }; /* override ac3.h to include coupling channel */ #undef AC3_MAX_CHANNELS #define AC3_MAX_CHANNELS 7 #define CPL_CH 0 #define AC3_OUTPUT_LFEON 8 typedef struct { int channel_mode; ///< channel mode (acmod) int block_switch[AC3_MAX_CHANNELS]; ///< block switch flags int dither_flag[AC3_MAX_CHANNELS]; ///< dither flags int dither_all; ///< true if all channels are dithered int cpl_in_use; ///< coupling in use int channel_in_cpl[AC3_MAX_CHANNELS]; ///< channel in coupling int phase_flags_in_use; ///< phase flags in use int phase_flags[18]; ///< phase flags int cpl_band_struct[18]; ///< coupling band structure int num_rematrixing_bands; ///< number of rematrixing bands int rematrixing_flags[4]; ///< rematrixing flags int exp_strategy[AC3_MAX_CHANNELS]; ///< exponent strategies int snr_offset[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets int fast_gain[AC3_MAX_CHANNELS]; ///< fast gain values (signal-to-mask ratio) int dba_mode[AC3_MAX_CHANNELS]; ///< delta bit allocation mode int dba_nsegs[AC3_MAX_CHANNELS]; ///< number of delta segments uint8_t dba_offsets[AC3_MAX_CHANNELS][8]; ///< delta segment offsets uint8_t dba_lengths[AC3_MAX_CHANNELS][8]; ///< delta segment lengths uint8_t dba_values[AC3_MAX_CHANNELS][8]; ///< delta values for each segment int sample_rate; ///< sample frequency, in Hz int bit_rate; ///< stream bit rate, in bits-per-second int frame_size; ///< current frame size, in bytes int channels; ///< number of total channels int fbw_channels; ///< number of full-bandwidth channels int lfe_on; ///< lfe channel in use int lfe_ch; ///< index of LFE channel int output_mode; ///< output channel configuration int out_channels; ///< number of output channels int center_mix_level; ///< Center mix level index int surround_mix_level; ///< Surround mix level index float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients float dynamic_range[2]; ///< dynamic range float cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates int num_cpl_bands; ///< number of coupling bands int num_cpl_subbands; ///< number of coupling sub bands int start_freq[AC3_MAX_CHANNELS]; ///< start frequency bin int end_freq[AC3_MAX_CHANNELS]; ///< end frequency bin AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters int8_t dexps[AC3_MAX_CHANNELS][256]; ///< decoded exponents uint8_t bap[AC3_MAX_CHANNELS][256]; ///< bit allocation pointers int16_t psd[AC3_MAX_CHANNELS][256]; ///< scaled exponents int16_t band_psd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients /* For IMDCT. */ MDCTContext imdct_512; ///< for 512 sample IMDCT MDCTContext imdct_256; ///< for 256 sample IMDCT DSPContext dsp; ///< for optimization float add_bias; ///< offset for float_to_int16 conversion float mul_bias; ///< scaling for float_to_int16 conversion DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients /* Miscellaneous. */ GetBitContext gbc; ///< bitstream reader AVRandomState dith_state; ///< for dither generation AVCodecContext *avctx; ///< parent context } AC3DecodeContext; /** * Symmetrical Dequantization * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization * Tables 7.19 to 7.23 */ static inline float symmetric_dequant(int code, int levels) { return (code - (levels >> 1)) * (2.0f / levels); } /* * Initialize tables at runtime. */ static void ac3_tables_init(void) { int i; /* generate grouped mantissa tables reference: Section 7.3.5 Ungrouping of Mantissas */ for(i=0; i<32; i++) { /* bap=1 mantissas */ b1_mantissas[i][0] = symmetric_dequant( i / 9 , 3); b1_mantissas[i][1] = symmetric_dequant((i % 9) / 3, 3); b1_mantissas[i][2] = symmetric_dequant((i % 9) % 3, 3); } for(i=0; i<128; i++) { /* bap=2 mantissas */ b2_mantissas[i][0] = symmetric_dequant( i / 25 , 5); b2_mantissas[i][1] = symmetric_dequant((i % 25) / 5, 5); b2_mantissas[i][2] = symmetric_dequant((i % 25) % 5, 5); /* bap=4 mantissas */ b4_mantissas[i][0] = symmetric_dequant(i / 11, 11); b4_mantissas[i][1] = symmetric_dequant(i % 11, 11); } /* generate ungrouped mantissa tables reference: Tables 7.21 and 7.23 */ for(i=0; i<7; i++) { /* bap=3 mantissas */ b3_mantissas[i] = symmetric_dequant(i, 7); } for(i=0; i<15; i++) { /* bap=5 mantissas */ b5_mantissas[i] = symmetric_dequant(i, 15); } /* generate dynamic range table reference: Section 7.7.1 Dynamic Range Control */ for(i=0; i<256; i++) { int v = (i >> 5) - ((i >> 7) << 3) - 5; dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20); } /* generate scale factors for exponents and asymmetrical dequantization reference: Section 7.3.2 Expansion of Mantissas for Asymmetric Quantization */ for (i = 0; i < 25; i++) scale_factors[i] = pow(2.0, -i); /* generate exponent tables reference: Section 7.1.3 Exponent Decoding */ for(i=0; i<128; i++) { exp_ungroup_tab[i][0] = i / 25; exp_ungroup_tab[i][1] = (i % 25) / 5; exp_ungroup_tab[i][2] = (i % 25) % 5; } } /** * AVCodec initialization */ static int ac3_decode_init(AVCodecContext *avctx) { AC3DecodeContext *s = avctx->priv_data; s->avctx = avctx; ac3_common_init(); ac3_tables_init(); ff_mdct_init(&s->imdct_256, 8, 1); ff_mdct_init(&s->imdct_512, 9, 1); ff_kbd_window_init(s->window, 5.0, 256); dsputil_init(&s->dsp, avctx); av_init_random(0, &s->dith_state); /* set bias values for float to int16 conversion */ if(s->dsp.float_to_int16 == ff_float_to_int16_c) { s->add_bias = 385.0f; s->mul_bias = 1.0f; } else { s->add_bias = 0.0f; s->mul_bias = 32767.0f; } /* allow downmixing to stereo or mono */ if (avctx->channels > 0 && avctx->request_channels > 0 && avctx->request_channels < avctx->channels && avctx->request_channels <= 2) { avctx->channels = avctx->request_channels; } return 0; } /** * Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream. * GetBitContext within AC3DecodeContext must point to * start of the synchronized ac3 bitstream. */ static int ac3_parse_header(AC3DecodeContext *s) { AC3HeaderInfo hdr; GetBitContext *gbc = &s->gbc; int err, i; err = ff_ac3_parse_header(gbc->buffer, &hdr); if(err) return err; if(hdr.bitstream_id > 10) return AC3_PARSE_ERROR_BSID; /* get decoding parameters from header info */ s->bit_alloc_params.sr_code = hdr.sr_code; s->channel_mode = hdr.channel_mode; s->lfe_on = hdr.lfe_on; s->bit_alloc_params.sr_shift = hdr.sr_shift; s->sample_rate = hdr.sample_rate; s->bit_rate = hdr.bit_rate; s->channels = hdr.channels; s->fbw_channels = s->channels - s->lfe_on; s->lfe_ch = s->fbw_channels + 1; s->frame_size = hdr.frame_size; /* set default output to all source channels */ s->out_channels = s->channels; s->output_mode = s->channel_mode; if(s->lfe_on) s->output_mode |= AC3_OUTPUT_LFEON; /* set default mix levels */ s->center_mix_level = 3; // -4.5dB s->surround_mix_level = 4; // -6.0dB /* skip over portion of header which has already been read */ skip_bits(gbc, 16); // skip the sync_word skip_bits(gbc, 16); // skip crc1 skip_bits(gbc, 8); // skip fscod and frmsizecod skip_bits(gbc, 11); // skip bsid, bsmod, and acmod if(s->channel_mode == AC3_CHMODE_STEREO) { skip_bits(gbc, 2); // skip dsurmod } else { if((s->channel_mode & 1) && s->channel_mode != AC3_CHMODE_MONO) s->center_mix_level = center_levels[get_bits(gbc, 2)]; if(s->channel_mode & 4) s->surround_mix_level = surround_levels[get_bits(gbc, 2)]; } skip_bits1(gbc); // skip lfeon /* read the rest of the bsi. read twice for dual mono mode. */ i = !(s->channel_mode); do { skip_bits(gbc, 5); // skip dialog normalization if (get_bits1(gbc)) skip_bits(gbc, 8); //skip compression if (get_bits1(gbc)) skip_bits(gbc, 8); //skip language code if (get_bits1(gbc)) skip_bits(gbc, 7); //skip audio production information } while (i--); skip_bits(gbc, 2); //skip copyright bit and original bitstream bit /* skip the timecodes (or extra bitstream information for Alternate Syntax) TODO: read & use the xbsi1 downmix levels */ if (get_bits1(gbc)) skip_bits(gbc, 14); //skip timecode1 / xbsi1 if (get_bits1(gbc)) skip_bits(gbc, 14); //skip timecode2 / xbsi2 /* skip additional bitstream info */ if (get_bits1(gbc)) { i = get_bits(gbc, 6); do { skip_bits(gbc, 8); } while(i--); } return 0; } /** * Set stereo downmixing coefficients based on frame header info. * reference: Section 7.8.2 Downmixing Into Two Channels */ static void set_downmix_coeffs(AC3DecodeContext *s) { int i; float cmix = gain_levels[s->center_mix_level]; float smix = gain_levels[s->surround_mix_level]; for(i=0; i<s->fbw_channels; i++) { s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]]; s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]]; } if(s->channel_mode > 1 && s->channel_mode & 1) { s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix; } if(s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) { int nf = s->channel_mode - 2; s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB; } if(s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) { int nf = s->channel_mode - 4; s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix; } } /** * Decode the grouped exponents according to exponent strategy. * reference: Section 7.1.3 Exponent Decoding */ static void decode_exponents(GetBitContext *gbc, int exp_strategy, int ngrps, uint8_t absexp, int8_t *dexps) { int i, j, grp, group_size; int dexp[256]; int expacc, prevexp; /* unpack groups */ group_size = exp_strategy + (exp_strategy == EXP_D45); for(grp=0,i=0; grp<ngrps; grp++) { expacc = get_bits(gbc, 7); dexp[i++] = exp_ungroup_tab[expacc][0]; dexp[i++] = exp_ungroup_tab[expacc][1]; dexp[i++] = exp_ungroup_tab[expacc][2]; } /* convert to absolute exps and expand groups */ prevexp = absexp; for(i=0; i<ngrps*3; i++) { prevexp = av_clip(prevexp + dexp[i]-2, 0, 24); for(j=0; j<group_size; j++) { dexps[(i*group_size)+j] = prevexp; } } } /** * Generate transform coefficients for each coupled channel in the coupling * range using the coupling coefficients and coupling coordinates. * reference: Section 7.4.3 Coupling Coordinate Format */ static void uncouple_channels(AC3DecodeContext *s) { int i, j, ch, bnd, subbnd; subbnd = -1; i = s->start_freq[CPL_CH]; for(bnd=0; bnd<s->num_cpl_bands; bnd++) { do { subbnd++; for(j=0; j<12; j++) { for(ch=1; ch<=s->fbw_channels; ch++) { if(s->channel_in_cpl[ch]) { s->transform_coeffs[ch][i] = s->transform_coeffs[CPL_CH][i] * s->cpl_coords[ch][bnd] * 8.0f; if (ch == 2 && s->phase_flags[bnd]) s->transform_coeffs[ch][i] = -s->transform_coeffs[ch][i]; } } i++; } } while(s->cpl_band_struct[subbnd]); } } /** * Grouped mantissas for 3-level 5-level and 11-level quantization */ typedef struct { float b1_mant[3]; float b2_mant[3]; float b4_mant[2]; int b1ptr; int b2ptr; int b4ptr; } mant_groups; /** * Get the transform coefficients for a particular channel * reference: Section 7.3 Quantization and Decoding of Mantissas */ static int get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m) { GetBitContext *gbc = &s->gbc; int i, gcode, tbap, start, end; uint8_t *exps; uint8_t *bap; float *coeffs; exps = s->dexps[ch_index]; bap = s->bap[ch_index]; coeffs = s->transform_coeffs[ch_index]; start = s->start_freq[ch_index]; end = s->end_freq[ch_index]; for (i = start; i < end; i++) { tbap = bap[i]; switch (tbap) { case 0: coeffs[i] = ((av_random(&s->dith_state) & 0xFFFF) / 65535.0f) - 0.5f; break; case 1: if(m->b1ptr > 2) { gcode = get_bits(gbc, 5); m->b1_mant[0] = b1_mantissas[gcode][0]; m->b1_mant[1] = b1_mantissas[gcode][1]; m->b1_mant[2] = b1_mantissas[gcode][2]; m->b1ptr = 0; } coeffs[i] = m->b1_mant[m->b1ptr++]; break; case 2: if(m->b2ptr > 2) { gcode = get_bits(gbc, 7); m->b2_mant[0] = b2_mantissas[gcode][0]; m->b2_mant[1] = b2_mantissas[gcode][1]; m->b2_mant[2] = b2_mantissas[gcode][2]; m->b2ptr = 0; } coeffs[i] = m->b2_mant[m->b2ptr++]; break; case 3: coeffs[i] = b3_mantissas[get_bits(gbc, 3)]; break; case 4: if(m->b4ptr > 1) { gcode = get_bits(gbc, 7); m->b4_mant[0] = b4_mantissas[gcode][0]; m->b4_mant[1] = b4_mantissas[gcode][1]; m->b4ptr = 0; } coeffs[i] = m->b4_mant[m->b4ptr++]; break; case 5: coeffs[i] = b5_mantissas[get_bits(gbc, 4)]; break; default: /* asymmetric dequantization */ coeffs[i] = get_sbits(gbc, quantization_tab[tbap]) * scale_factors[quantization_tab[tbap]-1]; break; } coeffs[i] *= scale_factors[exps[i]]; } return 0; } /** * Remove random dithering from coefficients with zero-bit mantissas * reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0) */ static void remove_dithering(AC3DecodeContext *s) { int ch, i; int end=0; float *coeffs; uint8_t *bap; for(ch=1; ch<=s->fbw_channels; ch++) { if(!s->dither_flag[ch]) { coeffs = s->transform_coeffs[ch]; bap = s->bap[ch]; if(s->channel_in_cpl[ch]) end = s->start_freq[CPL_CH]; else end = s->end_freq[ch]; for(i=0; i<end; i++) { if(!bap[i]) coeffs[i] = 0.0f; } if(s->channel_in_cpl[ch]) { bap = s->bap[CPL_CH]; for(; i<s->end_freq[CPL_CH]; i++) { if(!bap[i]) coeffs[i] = 0.0f; } } } } } /** * Get the transform coefficients. */ static int get_transform_coeffs(AC3DecodeContext *s) { int ch, end; int got_cplchan = 0; mant_groups m; m.b1ptr = m.b2ptr = m.b4ptr = 3; for (ch = 1; ch <= s->channels; ch++) { /* transform coefficients for full-bandwidth channel */ if (get_transform_coeffs_ch(s, ch, &m)) return -1; /* tranform coefficients for coupling channel come right after the coefficients for the first coupled channel*/ if (s->channel_in_cpl[ch]) { if (!got_cplchan) { if (get_transform_coeffs_ch(s, CPL_CH, &m)) { av_log(s->avctx, AV_LOG_ERROR, "error in decoupling channels\n"); return -1; } uncouple_channels(s); got_cplchan = 1; } end = s->end_freq[CPL_CH]; } else { end = s->end_freq[ch]; } do s->transform_coeffs[ch][end] = 0; while(++end < 256); } /* if any channel doesn't use dithering, zero appropriate coefficients */ if(!s->dither_all) remove_dithering(s); return 0; } /** * Stereo rematrixing. * reference: Section 7.5.4 Rematrixing : Decoding Technique */ static void do_rematrixing(AC3DecodeContext *s) { int bnd, i; int end, bndend; float tmp0, tmp1; end = FFMIN(s->end_freq[1], s->end_freq[2]); for(bnd=0; bnd<s->num_rematrixing_bands; bnd++) { if(s->rematrixing_flags[bnd]) { bndend = FFMIN(end, rematrix_band_tab[bnd+1]); for(i=rematrix_band_tab[bnd]; i<bndend; i++) { tmp0 = s->transform_coeffs[1][i]; tmp1 = s->transform_coeffs[2][i]; s->transform_coeffs[1][i] = tmp0 + tmp1; s->transform_coeffs[2][i] = tmp0 - tmp1; } } } } /** * Perform the 256-point IMDCT */ static void do_imdct_256(AC3DecodeContext *s, int chindex) { int i, k; DECLARE_ALIGNED_16(float, x[128]); FFTComplex z[2][64]; float *o_ptr = s->tmp_output; for(i=0; i<2; i++) { /* de-interleave coefficients */ for(k=0; k<128; k++) { x[k] = s->transform_coeffs[chindex][2*k+i]; } /* run standard IMDCT */ s->imdct_256.fft.imdct_calc(&s->imdct_256, o_ptr, x, s->tmp_imdct); /* reverse the post-rotation & reordering from standard IMDCT */ for(k=0; k<32; k++) { z[i][32+k].re = -o_ptr[128+2*k]; z[i][32+k].im = -o_ptr[2*k]; z[i][31-k].re = o_ptr[2*k+1]; z[i][31-k].im = o_ptr[128+2*k+1]; } } /* apply AC-3 post-rotation & reordering */ for(k=0; k<64; k++) { o_ptr[ 2*k ] = -z[0][ k].im; o_ptr[ 2*k+1] = z[0][63-k].re; o_ptr[128+2*k ] = -z[0][ k].re; o_ptr[128+2*k+1] = z[0][63-k].im; o_ptr[256+2*k ] = -z[1][ k].re; o_ptr[256+2*k+1] = z[1][63-k].im; o_ptr[384+2*k ] = z[1][ k].im; o_ptr[384+2*k+1] = -z[1][63-k].re; } } /** * Inverse MDCT Transform. * Convert frequency domain coefficients to time-domain audio samples. * reference: Section 7.9.4 Transformation Equations */ static inline void do_imdct(AC3DecodeContext *s) { int ch; int channels; /* Don't perform the IMDCT on the LFE channel unless it's used in the output */ channels = s->fbw_channels; if(s->output_mode & AC3_OUTPUT_LFEON) channels++; for (ch=1; ch<=channels; ch++) { if (s->block_switch[ch]) { do_imdct_256(s, ch); } else { s->imdct_512.fft.imdct_calc(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch], s->tmp_imdct); } /* For the first half of the block, apply the window, add the delay from the previous block, and send to output */ s->dsp.vector_fmul_add_add(s->output[ch-1], s->tmp_output, s->window, s->delay[ch-1], 0, 256, 1); /* For the second half of the block, apply the window and store the samples to delay, to be combined with the next block */ s->dsp.vector_fmul_reverse(s->delay[ch-1], s->tmp_output+256, s->window, 256); } } /** * Downmix the output to mono or stereo. */ static void ac3_downmix(AC3DecodeContext *s) { int i, j; float v0, v1, s0, s1; for(i=0; i<256; i++) { v0 = v1 = s0 = s1 = 0.0f; for(j=0; j<s->fbw_channels; j++) { v0 += s->output[j][i] * s->downmix_coeffs[j][0]; v1 += s->output[j][i] * s->downmix_coeffs[j][1]; s0 += s->downmix_coeffs[j][0]; s1 += s->downmix_coeffs[j][1]; } v0 /= s0; v1 /= s1; if(s->output_mode == AC3_CHMODE_MONO) { s->output[0][i] = (v0 + v1) * LEVEL_MINUS_3DB; } else if(s->output_mode == AC3_CHMODE_STEREO) { s->output[0][i] = v0; s->output[1][i] = v1; } } } /** * Parse an audio block from AC-3 bitstream. */ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk) { int fbw_channels = s->fbw_channels; int channel_mode = s->channel_mode; int i, bnd, seg, ch; GetBitContext *gbc = &s->gbc; uint8_t bit_alloc_stages[AC3_MAX_CHANNELS]; memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS); /* block switch flags */ for (ch = 1; ch <= fbw_channels; ch++) s->block_switch[ch] = get_bits1(gbc); /* dithering flags */ s->dither_all = 1; for (ch = 1; ch <= fbw_channels; ch++) { s->dither_flag[ch] = get_bits1(gbc); if(!s->dither_flag[ch]) s->dither_all = 0; } /* dynamic range */ i = !(s->channel_mode); do { if(get_bits1(gbc)) { s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)]-1.0) * s->avctx->drc_scale)+1.0; } else if(blk == 0) { s->dynamic_range[i] = 1.0f; } } while(i--); /* coupling strategy */ if (get_bits1(gbc)) { memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); s->cpl_in_use = get_bits1(gbc); if (s->cpl_in_use) { /* coupling in use */ int cpl_begin_freq, cpl_end_freq; /* determine which channels are coupled */ for (ch = 1; ch <= fbw_channels; ch++) s->channel_in_cpl[ch] = get_bits1(gbc); /* phase flags in use */ if (channel_mode == AC3_CHMODE_STEREO) s->phase_flags_in_use = get_bits1(gbc); /* coupling frequency range and band structure */ cpl_begin_freq = get_bits(gbc, 4); cpl_end_freq = get_bits(gbc, 4); if (3 + cpl_end_freq - cpl_begin_freq < 0) { av_log(s->avctx, AV_LOG_ERROR, "3+cplendf = %d < cplbegf = %d\n", 3+cpl_end_freq, cpl_begin_freq); return -1; } s->num_cpl_bands = s->num_cpl_subbands = 3 + cpl_end_freq - cpl_begin_freq; s->start_freq[CPL_CH] = cpl_begin_freq * 12 + 37; s->end_freq[CPL_CH] = cpl_end_freq * 12 + 73; for (bnd = 0; bnd < s->num_cpl_subbands - 1; bnd++) { if (get_bits1(gbc)) { s->cpl_band_struct[bnd] = 1; s->num_cpl_bands--; } } s->cpl_band_struct[s->num_cpl_subbands-1] = 0; } else { /* coupling not in use */ for (ch = 1; ch <= fbw_channels; ch++) s->channel_in_cpl[ch] = 0; } } /* coupling coordinates */ if (s->cpl_in_use) { int cpl_coords_exist = 0; for (ch = 1; ch <= fbw_channels; ch++) { if (s->channel_in_cpl[ch]) { if (get_bits1(gbc)) { int master_cpl_coord, cpl_coord_exp, cpl_coord_mant; cpl_coords_exist = 1; master_cpl_coord = 3 * get_bits(gbc, 2); for (bnd = 0; bnd < s->num_cpl_bands; bnd++) { cpl_coord_exp = get_bits(gbc, 4); cpl_coord_mant = get_bits(gbc, 4); if (cpl_coord_exp == 15) s->cpl_coords[ch][bnd] = cpl_coord_mant / 16.0f; else s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16.0f) / 32.0f; s->cpl_coords[ch][bnd] *= scale_factors[cpl_coord_exp + master_cpl_coord]; } } } } /* phase flags */ if (channel_mode == AC3_CHMODE_STEREO && cpl_coords_exist) { for (bnd = 0; bnd < s->num_cpl_bands; bnd++) { s->phase_flags[bnd] = s->phase_flags_in_use? get_bits1(gbc) : 0; } } } /* stereo rematrixing strategy and band structure */ if (channel_mode == AC3_CHMODE_STEREO) { if (get_bits1(gbc)) { s->num_rematrixing_bands = 4; if(s->cpl_in_use && s->start_freq[CPL_CH] <= 61) s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37); for(bnd=0; bnd<s->num_rematrixing_bands; bnd++) s->rematrixing_flags[bnd] = get_bits1(gbc); } } /* exponent strategies for each channel */ s->exp_strategy[CPL_CH] = EXP_REUSE; s->exp_strategy[s->lfe_ch] = EXP_REUSE; for (ch = !s->cpl_in_use; ch <= s->channels; ch++) { if(ch == s->lfe_ch) s->exp_strategy[ch] = get_bits(gbc, 1); else s->exp_strategy[ch] = get_bits(gbc, 2); if(s->exp_strategy[ch] != EXP_REUSE) bit_alloc_stages[ch] = 3; } /* channel bandwidth */ for (ch = 1; ch <= fbw_channels; ch++) { s->start_freq[ch] = 0; if (s->exp_strategy[ch] != EXP_REUSE) { int prev = s->end_freq[ch]; if (s->channel_in_cpl[ch]) s->end_freq[ch] = s->start_freq[CPL_CH]; else { int bandwidth_code = get_bits(gbc, 6); if (bandwidth_code > 60) { av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60", bandwidth_code); return -1; } s->end_freq[ch] = bandwidth_code * 3 + 73; } if(blk > 0 && s->end_freq[ch] != prev) memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); } } s->start_freq[s->lfe_ch] = 0; s->end_freq[s->lfe_ch] = 7; /* decode exponents for each channel */ for (ch = !s->cpl_in_use; ch <= s->channels; ch++) { if (s->exp_strategy[ch] != EXP_REUSE) { int group_size, num_groups; group_size = 3 << (s->exp_strategy[ch] - 1); if(ch == CPL_CH) num_groups = (s->end_freq[ch] - s->start_freq[ch]) / group_size; else if(ch == s->lfe_ch) num_groups = 2; else num_groups = (s->end_freq[ch] + group_size - 4) / group_size; s->dexps[ch][0] = get_bits(gbc, 4) << !ch; decode_exponents(gbc, s->exp_strategy[ch], num_groups, s->dexps[ch][0], &s->dexps[ch][s->start_freq[ch]+!!ch]); if(ch != CPL_CH && ch != s->lfe_ch) skip_bits(gbc, 2); /* skip gainrng */ } } /* bit allocation information */ if (get_bits1(gbc)) { s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift; s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift; s->bit_alloc_params.slow_gain = ff_ac3_slow_gain_tab[get_bits(gbc, 2)]; s->bit_alloc_params.db_per_bit = ff_ac3_db_per_bit_tab[get_bits(gbc, 2)]; s->bit_alloc_params.floor = ff_ac3_floor_tab[get_bits(gbc, 3)]; for(ch=!s->cpl_in_use; ch<=s->channels; ch++) { bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } } /* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */ if (get_bits1(gbc)) { int csnr; csnr = (get_bits(gbc, 6) - 15) << 4; for (ch = !s->cpl_in_use; ch <= s->channels; ch++) { /* snr offset and fast gain */ s->snr_offset[ch] = (csnr + get_bits(gbc, 4)) << 2; s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)]; } memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); } /* coupling leak information */ if (s->cpl_in_use && get_bits1(gbc)) { s->bit_alloc_params.cpl_fast_leak = get_bits(gbc, 3); s->bit_alloc_params.cpl_slow_leak = get_bits(gbc, 3); bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2); } /* delta bit allocation information */ if (get_bits1(gbc)) { /* delta bit allocation exists (strategy) */ for (ch = !s->cpl_in_use; ch <= fbw_channels; ch++) { s->dba_mode[ch] = get_bits(gbc, 2); if (s->dba_mode[ch] == DBA_RESERVED) { av_log(s->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n"); return -1; } bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } /* channel delta offset, len and bit allocation */ for (ch = !s->cpl_in_use; ch <= fbw_channels; ch++) { if (s->dba_mode[ch] == DBA_NEW) { s->dba_nsegs[ch] = get_bits(gbc, 3); for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) { s->dba_offsets[ch][seg] = get_bits(gbc, 5); s->dba_lengths[ch][seg] = get_bits(gbc, 4); s->dba_values[ch][seg] = get_bits(gbc, 3); } } } } else if(blk == 0) { for(ch=0; ch<=s->channels; ch++) { s->dba_mode[ch] = DBA_NONE; } } /* Bit allocation */ for(ch=!s->cpl_in_use; ch<=s->channels; ch++) { if(bit_alloc_stages[ch] > 2) { /* Exponent mapping into PSD and PSD integration */ ff_ac3_bit_alloc_calc_psd(s->dexps[ch], s->start_freq[ch], s->end_freq[ch], s->psd[ch], s->band_psd[ch]); } if(bit_alloc_stages[ch] > 1) { /* Compute excitation function, Compute masking curve, and Apply delta bit allocation */ ff_ac3_bit_alloc_calc_mask(&s->bit_alloc_params, s->band_psd[ch], s->start_freq[ch], s->end_freq[ch], s->fast_gain[ch], (ch == s->lfe_ch), s->dba_mode[ch], s->dba_nsegs[ch], s->dba_offsets[ch], s->dba_lengths[ch], s->dba_values[ch], s->mask[ch]); } if(bit_alloc_stages[ch] > 0) { /* Compute bit allocation */ ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch], s->start_freq[ch], s->end_freq[ch], s->snr_offset[ch], s->bit_alloc_params.floor, s->bap[ch]); } } /* unused dummy data */ if (get_bits1(gbc)) { int skipl = get_bits(gbc, 9); while(skipl--) skip_bits(gbc, 8); } /* unpack the transform coefficients this also uncouples channels if coupling is in use. */ if (get_transform_coeffs(s)) { av_log(s->avctx, AV_LOG_ERROR, "Error in routine get_transform_coeffs\n"); return -1; } /* recover coefficients if rematrixing is in use */ if(s->channel_mode == AC3_CHMODE_STEREO) do_rematrixing(s); /* apply scaling to coefficients (headroom, dynrng) */ for(ch=1; ch<=s->channels; ch++) { float gain = 2.0f * s->mul_bias; if(s->channel_mode == AC3_CHMODE_DUALMONO) { gain *= s->dynamic_range[ch-1]; } else { gain *= s->dynamic_range[0]; } for(i=0; i<s->end_freq[ch]; i++) { s->transform_coeffs[ch][i] *= gain; } } do_imdct(s); /* downmix output if needed */ if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) && s->fbw_channels == s->out_channels)) { ac3_downmix(s); } /* convert float to 16-bit integer */ for(ch=0; ch<s->out_channels; ch++) { for(i=0; i<256; i++) { s->output[ch][i] += s->add_bias; } s->dsp.float_to_int16(s->int_output[ch], s->output[ch], 256); } return 0; } /** * Decode a single AC-3 frame. */ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, uint8_t *buf, int buf_size) { AC3DecodeContext *s = avctx->priv_data; int16_t *out_samples = (int16_t *)data; int i, blk, ch, err; /* initialize the GetBitContext with the start of valid AC-3 Frame */ init_get_bits(&s->gbc, buf, buf_size * 8); /* parse the syncinfo */ err = ac3_parse_header(s); if(err) { switch(err) { case AC3_PARSE_ERROR_SYNC: av_log(avctx, AV_LOG_ERROR, "frame sync error\n"); break; case AC3_PARSE_ERROR_BSID: av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n"); break; case AC3_PARSE_ERROR_SAMPLE_RATE: av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n"); break; case AC3_PARSE_ERROR_FRAME_SIZE: av_log(avctx, AV_LOG_ERROR, "invalid frame size\n"); break; default: av_log(avctx, AV_LOG_ERROR, "invalid header\n"); break; } return -1; } /* check that reported frame size fits in input buffer */ if(s->frame_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); return -1; } /* check for crc mismatch */ if(avctx->error_resilience >= FF_ER_CAREFUL) { if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) { av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n"); return -1; } /* TODO: error concealment */ } avctx->sample_rate = s->sample_rate; avctx->bit_rate = s->bit_rate; /* channel config */ s->out_channels = s->channels; if (avctx->request_channels > 0 && avctx->request_channels <= 2 && avctx->request_channels < s->channels) { s->out_channels = avctx->request_channels; s->output_mode = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO; } avctx->channels = s->out_channels; /* set downmixing coefficients if needed */ if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) && s->fbw_channels == s->out_channels)) { set_downmix_coeffs(s); } /* parse the audio blocks */ for (blk = 0; blk < NB_BLOCKS; blk++) { if (ac3_parse_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n"); *data_size = 0; return s->frame_size; } for (i = 0; i < 256; i++) for (ch = 0; ch < s->out_channels; ch++) *(out_samples++) = s->int_output[ch][i]; } *data_size = NB_BLOCKS * 256 * avctx->channels * sizeof (int16_t); return s->frame_size; } /** * Uninitialize the AC-3 decoder. */ static int ac3_decode_end(AVCodecContext *avctx) { AC3DecodeContext *s = avctx->priv_data; ff_mdct_end(&s->imdct_512); ff_mdct_end(&s->imdct_256); return 0; } AVCodec ac3_decoder = { .name = "ac3", .type = CODEC_TYPE_AUDIO, .id = CODEC_ID_AC3, .priv_data_size = sizeof (AC3DecodeContext), .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, };