/* * The simplest AC-3 encoder * Copyright (c) 2000 Fabrice Bellard * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser 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 */ /** * @file * The simplest AC-3 encoder. */ //#define DEBUG #include "libavcore/audioconvert.h" #include "libavutil/crc.h" #include "avcodec.h" #include "put_bits.h" #include "ac3.h" #include "audioconvert.h" #define MDCT_NBITS 9 #define MDCT_SAMPLES (1 << MDCT_NBITS) /** Scale a float value by 2^bits and convert to an integer. */ #define SCALE_FLOAT(a, bits) lrintf((a) * (float)(1 << (bits))) /** Scale a float value by 2^15, convert to an integer, and clip to int16_t range. */ #define FIX15(a) av_clip_int16(SCALE_FLOAT(a, 15)) /** * Compex number. * Used in fixed-point MDCT calculation. */ typedef struct IComplex { int16_t re,im; } IComplex; /** * AC-3 encoder private context. */ typedef struct AC3EncodeContext { PutBitContext pb; ///< bitstream writer context int bitstream_id; ///< bitstream id (bsid) int bitstream_mode; ///< bitstream mode (bsmod) int bit_rate; ///< target bit rate, in bits-per-second int sample_rate; ///< sampling frequency, in Hz int frame_size_min; ///< minimum frame size in case rounding is necessary int frame_size; ///< current frame size in bytes int frame_size_code; ///< frame size code (frmsizecod) int bits_written; ///< bit count (used to avg. bitrate) int samples_written; ///< sample count (used to avg. bitrate) int fbw_channels; ///< number of full-bandwidth channels (nfchans) int channels; ///< total number of channels (nchans) int lfe_on; ///< indicates if there is an LFE channel (lfeon) int lfe_channel; ///< channel index of the LFE channel int channel_mode; ///< channel mode (acmod) const uint8_t *channel_map; ///< channel map used to reorder channels int bandwidth_code[AC3_MAX_CHANNELS]; ///< bandwidth code (0 to 60) (chbwcod) int nb_coefs[AC3_MAX_CHANNELS]; /* bitrate allocation control */ int slow_gain_code; ///< slow gain code (sgaincod) int slow_decay_code; ///< slow decay code (sdcycod) int fast_decay_code; ///< fast decay code (fdcycod) int db_per_bit_code; ///< dB/bit code (dbpbcod) int floor_code; ///< floor code (floorcod) AC3BitAllocParameters bit_alloc; ///< bit allocation parameters int coarse_snr_offset; ///< coarse SNR offsets (csnroffst) int fast_gain_code[AC3_MAX_CHANNELS]; ///< fast gain codes (signal-to-mask ratio) (fgaincod) int fine_snr_offset[AC3_MAX_CHANNELS]; ///< fine SNR offsets (fsnroffst) /* mantissa encoding */ int mant1_cnt, mant2_cnt, mant4_cnt; ///< mantissa counts for bap=1,2,4 int16_t last_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< last 256 samples from previous frame } AC3EncodeContext; /** MDCT and FFT tables */ static int16_t costab[64]; static int16_t sintab[64]; static int16_t xcos1[128]; static int16_t xsin1[128]; /** * Initialize FFT tables. * @param ln log2(FFT size) */ static av_cold void fft_init(int ln) { int i, n, n2; float alpha; n = 1 << ln; n2 = n >> 1; for (i = 0; i < n2; i++) { alpha = 2.0 * M_PI * i / n; costab[i] = FIX15(cos(alpha)); sintab[i] = FIX15(sin(alpha)); } } /** * Initialize MDCT tables. * @param nbits log2(MDCT size) */ static av_cold void mdct_init(int nbits) { int i, n, n4; n = 1 << nbits; n4 = n >> 2; fft_init(nbits - 2); for (i = 0; i < n4; i++) { float alpha = 2.0 * M_PI * (i + 1.0 / 8.0) / n; xcos1[i] = FIX15(-cos(alpha)); xsin1[i] = FIX15(-sin(alpha)); } } /** Butterfly op */ #define BF(pre, pim, qre, qim, pre1, pim1, qre1, qim1) \ { \ int ax, ay, bx, by; \ bx = pre1; \ by = pim1; \ ax = qre1; \ ay = qim1; \ pre = (bx + ax) >> 1; \ pim = (by + ay) >> 1; \ qre = (bx - ax) >> 1; \ qim = (by - ay) >> 1; \ } /** Complex multiply */ #define CMUL(pre, pim, are, aim, bre, bim) \ { \ pre = (MUL16(are, bre) - MUL16(aim, bim)) >> 15; \ pim = (MUL16(are, bim) + MUL16(bre, aim)) >> 15; \ } /** * Calculate a 2^n point complex FFT on 2^ln points. * @param z complex input/output samples * @param ln log2(FFT size) */ static void fft(IComplex *z, int ln) { int j, l, np, np2; int nblocks, nloops; register IComplex *p,*q; int tmp_re, tmp_im; np = 1 << ln; /* reverse */ for (j = 0; j < np; j++) { int k = av_reverse[j] >> (8 - ln); if (k < j) FFSWAP(IComplex, z[k], z[j]); } /* pass 0 */ p = &z[0]; j = np >> 1; do { BF(p[0].re, p[0].im, p[1].re, p[1].im, p[0].re, p[0].im, p[1].re, p[1].im); p += 2; } while (--j); /* pass 1 */ p = &z[0]; j = np >> 2; do { BF(p[0].re, p[0].im, p[2].re, p[2].im, p[0].re, p[0].im, p[2].re, p[2].im); BF(p[1].re, p[1].im, p[3].re, p[3].im, p[1].re, p[1].im, p[3].im, -p[3].re); p+=4; } while (--j); /* pass 2 .. ln-1 */ nblocks = np >> 3; nloops = 1 << 2; np2 = np >> 1; do { p = z; q = z + nloops; for (j = 0; j < nblocks; j++) { BF(p->re, p->im, q->re, q->im, p->re, p->im, q->re, q->im); p++; q++; for(l = nblocks; l < np2; l += nblocks) { CMUL(tmp_re, tmp_im, costab[l], -sintab[l], q->re, q->im); BF(p->re, p->im, q->re, q->im, p->re, p->im, tmp_re, tmp_im); p++; q++; } p += nloops; q += nloops; } nblocks = nblocks >> 1; nloops = nloops << 1; } while (nblocks); } /** * Calculate a 512-point MDCT * @param out 256 output frequency coefficients * @param in 512 windowed input audio samples */ static void mdct512(int32_t *out, int16_t *in) { int i, re, im, re1, im1; int16_t rot[MDCT_SAMPLES]; IComplex x[MDCT_SAMPLES/4]; /* shift to simplify computations */ for (i = 0; i < MDCT_SAMPLES/4; i++) rot[i] = -in[i + 3*MDCT_SAMPLES/4]; for (;i < MDCT_SAMPLES; i++) rot[i] = in[i - MDCT_SAMPLES/4]; /* pre rotation */ for (i = 0; i < MDCT_SAMPLES/4; i++) { re = ((int)rot[ 2*i] - (int)rot[MDCT_SAMPLES -1-2*i]) >> 1; im = -((int)rot[MDCT_SAMPLES/2+2*i] - (int)rot[MDCT_SAMPLES/2-1-2*i]) >> 1; CMUL(x[i].re, x[i].im, re, im, -xcos1[i], xsin1[i]); } fft(x, MDCT_NBITS - 2); /* post rotation */ for (i = 0; i < MDCT_SAMPLES/4; i++) { re = x[i].re; im = x[i].im; CMUL(re1, im1, re, im, xsin1[i], xcos1[i]); out[ 2*i] = im1; out[MDCT_SAMPLES/2-1-2*i] = re1; } } /** * Calculate the log2() of the maximum absolute value in an array. * @param tab input array * @param n number of values in the array * @return log2(max(abs(tab[]))) */ static int log2_tab(int16_t *tab, int n) { int i, v; v = 0; for (i = 0; i < n; i++) v |= abs(tab[i]); return av_log2(v); } /** * Left-shift each value in an array by a specified amount. * @param tab input array * @param n number of values in the array * @param lshift left shift amount. a negative value means right shift. */ static void lshift_tab(int16_t *tab, int n, int lshift) { int i; if (lshift > 0) { for(i = 0; i < n; i++) tab[i] <<= lshift; } else if (lshift < 0) { lshift = -lshift; for (i = 0; i < n; i++) tab[i] >>= lshift; } } /** * Calculate the sum of absolute differences (SAD) between 2 sets of exponents. */ static int calc_exp_diff(uint8_t *exp1, uint8_t *exp2, int n) { int sum, i; sum = 0; for (i = 0; i < n; i++) sum += abs(exp1[i] - exp2[i]); return sum; } /** * Exponent Difference Threshold. * New exponents are sent if their SAD exceed this number. */ #define EXP_DIFF_THRESHOLD 1000 /** * Calculate exponent strategies for all blocks in a single channel. */ static void compute_exp_strategy_ch(uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS], uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS], int ch, int is_lfe) { int blk, blk1; int exp_diff; /* estimate if the exponent variation & decide if they should be reused in the next frame */ exp_strategy[0][ch] = EXP_NEW; for (blk = 1; blk < AC3_MAX_BLOCKS; blk++) { exp_diff = calc_exp_diff(exp[blk][ch], exp[blk-1][ch], AC3_MAX_COEFS); if (exp_diff > EXP_DIFF_THRESHOLD) exp_strategy[blk][ch] = EXP_NEW; else exp_strategy[blk][ch] = EXP_REUSE; } if (is_lfe) return; /* now select the encoding strategy type : if exponents are often recoded, we use a coarse encoding */ blk = 0; while (blk < AC3_MAX_BLOCKS) { blk1 = blk + 1; while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE) blk1++; switch (blk1 - blk) { case 1: exp_strategy[blk][ch] = EXP_D45; break; case 2: case 3: exp_strategy[blk][ch] = EXP_D25; break; default: exp_strategy[blk][ch] = EXP_D15; break; } blk = blk1; } } /** * Set each encoded exponent in a block to the minimum of itself and the * exponent in the same frequency bin of a following block. * exp[i] = min(exp[i], exp1[i] */ static void exponent_min(uint8_t exp[AC3_MAX_COEFS], uint8_t exp1[AC3_MAX_COEFS], int n) { int i; for (i = 0; i < n; i++) { if (exp1[i] < exp[i]) exp[i] = exp1[i]; } } /** * Update the exponents so that they are the ones the decoder will decode. * @return the number of bits used to encode the exponents. */ static int encode_exponents_blk_ch(uint8_t encoded_exp[AC3_MAX_COEFS], uint8_t exp[AC3_MAX_COEFS], int nb_exps, int exp_strategy) { int group_size, nb_groups, i, j, k, exp_min; uint8_t exp1[AC3_MAX_COEFS]; group_size = exp_strategy + (exp_strategy == EXP_D45); nb_groups = ((nb_exps + (group_size * 3) - 4) / (3 * group_size)) * 3; /* for each group, compute the minimum exponent */ exp1[0] = exp[0]; /* DC exponent is handled separately */ k = 1; for (i = 1; i <= nb_groups; i++) { exp_min = exp[k]; assert(exp_min >= 0 && exp_min <= 24); for (j = 1; j < group_size; j++) { if (exp[k+j] < exp_min) exp_min = exp[k+j]; } exp1[i] = exp_min; k += group_size; } /* constraint for DC exponent */ if (exp1[0] > 15) exp1[0] = 15; /* decrease the delta between each groups to within 2 so that they can be differentially encoded */ for (i = 1; i <= nb_groups; i++) exp1[i] = FFMIN(exp1[i], exp1[i-1] + 2); for (i = nb_groups-1; i >= 0; i--) exp1[i] = FFMIN(exp1[i], exp1[i+1] + 2); /* now we have the exponent values the decoder will see */ encoded_exp[0] = exp1[0]; k = 1; for (i = 1; i <= nb_groups; i++) { for (j = 0; j < group_size; j++) encoded_exp[k+j] = exp1[i]; k += group_size; } return 4 + (nb_groups / 3) * 7; } /** * Calculate the number of bits needed to encode a set of mantissas. */ static int compute_mantissa_size(AC3EncodeContext *s, uint8_t *m, int nb_coefs) { int bits, mant, i; bits = 0; for (i = 0; i < nb_coefs; i++) { mant = m[i]; switch (mant) { case 0: /* nothing */ break; case 1: /* 3 mantissa in 5 bits */ if (s->mant1_cnt == 0) bits += 5; if (++s->mant1_cnt == 3) s->mant1_cnt = 0; break; case 2: /* 3 mantissa in 7 bits */ if (s->mant2_cnt == 0) bits += 7; if (++s->mant2_cnt == 3) s->mant2_cnt = 0; break; case 3: bits += 3; break; case 4: /* 2 mantissa in 7 bits */ if (s->mant4_cnt == 0) bits += 7; if (++s->mant4_cnt == 2) s->mant4_cnt = 0; break; case 14: bits += 14; break; case 15: bits += 16; break; default: bits += mant - 1; break; } } return bits; } /** * Calculate masking curve based on the final exponents. * Also calculate the power spectral densities to use in future calculations. */ static void bit_alloc_masking(AC3EncodeContext *s, uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS], uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS], int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS], int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS]) { int blk, ch; int16_t band_psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS]; for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { for (ch = 0; ch < s->channels; ch++) { if(exp_strategy[blk][ch] == EXP_REUSE) { memcpy(psd[blk][ch], psd[blk-1][ch], AC3_MAX_COEFS*sizeof(psd[0][0][0])); memcpy(mask[blk][ch], mask[blk-1][ch], AC3_CRITICAL_BANDS*sizeof(mask[0][0][0])); } else { ff_ac3_bit_alloc_calc_psd(encoded_exp[blk][ch], 0, s->nb_coefs[ch], psd[blk][ch], band_psd[blk][ch]); ff_ac3_bit_alloc_calc_mask(&s->bit_alloc, band_psd[blk][ch], 0, s->nb_coefs[ch], ff_ac3_fast_gain_tab[s->fast_gain_code[ch]], ch == s->lfe_channel, DBA_NONE, 0, NULL, NULL, NULL, mask[blk][ch]); } } } } /** * Run the bit allocation with a given SNR offset. * This calculates the bit allocation pointers that will be used to determine * the quantization of each mantissa. * @return the number of remaining bits (positive or negative) if the given * SNR offset is used to quantize the mantissas. */ static int bit_alloc(AC3EncodeContext *s, int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS], int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS], uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS], int frame_bits, int coarse_snr_offset, int fine_snr_offset) { int blk, ch; int snr_offset; snr_offset = (((coarse_snr_offset - 15) << 4) + fine_snr_offset) << 2; for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { s->mant1_cnt = 0; s->mant2_cnt = 0; s->mant4_cnt = 0; for (ch = 0; ch < s->channels; ch++) { ff_ac3_bit_alloc_calc_bap(mask[blk][ch], psd[blk][ch], 0, s->nb_coefs[ch], snr_offset, s->bit_alloc.floor, ff_ac3_bap_tab, bap[blk][ch]); frame_bits += compute_mantissa_size(s, bap[blk][ch], s->nb_coefs[ch]); } } return 8 * s->frame_size - frame_bits; } #define SNR_INC1 4 /** * Perform bit allocation search. * Finds the SNR offset value that maximizes quality and fits in the specified * frame size. Output is the SNR offset and a set of bit allocation pointers * used to quantize the mantissas. */ static int compute_bit_allocation(AC3EncodeContext *s, uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS], uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS], uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS], int frame_bits) { int blk, ch; int coarse_snr_offset, fine_snr_offset; uint8_t bap1[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS]; int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS]; int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS]; static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 }; /* init default parameters */ s->slow_decay_code = 2; s->fast_decay_code = 1; s->slow_gain_code = 1; s->db_per_bit_code = 2; s->floor_code = 4; for (ch = 0; ch < s->channels; ch++) s->fast_gain_code[ch] = 4; /* compute real values */ s->bit_alloc.slow_decay = ff_ac3_slow_decay_tab[s->slow_decay_code] >> s->bit_alloc.sr_shift; s->bit_alloc.fast_decay = ff_ac3_fast_decay_tab[s->fast_decay_code] >> s->bit_alloc.sr_shift; s->bit_alloc.slow_gain = ff_ac3_slow_gain_tab[s->slow_gain_code]; s->bit_alloc.db_per_bit = ff_ac3_db_per_bit_tab[s->db_per_bit_code]; s->bit_alloc.floor = ff_ac3_floor_tab[s->floor_code]; /* header size */ frame_bits += 65; // if (s->channel_mode == 2) // frame_bits += 2; frame_bits += frame_bits_inc[s->channel_mode]; /* audio blocks */ for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { frame_bits += s->fbw_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */ if (s->channel_mode == AC3_CHMODE_STEREO) { frame_bits++; /* rematstr */ if (!blk) frame_bits += 4; } frame_bits += 2 * s->fbw_channels; /* chexpstr[2] * c */ if (s->lfe_on) frame_bits++; /* lfeexpstr */ for (ch = 0; ch < s->fbw_channels; ch++) { if (exp_strategy[blk][ch] != EXP_REUSE) frame_bits += 6 + 2; /* chbwcod[6], gainrng[2] */ } frame_bits++; /* baie */ frame_bits++; /* snr */ frame_bits += 2; /* delta / skip */ } frame_bits++; /* cplinu for block 0 */ /* bit alloc info */ /* sdcycod[2], fdcycod[2], sgaincod[2], dbpbcod[2], floorcod[3] */ /* csnroffset[6] */ /* (fsnoffset[4] + fgaincod[4]) * c */ frame_bits += 2*4 + 3 + 6 + s->channels * (4 + 3); /* auxdatae, crcrsv */ frame_bits += 2; /* CRC */ frame_bits += 16; /* calculate psd and masking curve before doing bit allocation */ bit_alloc_masking(s, encoded_exp, exp_strategy, psd, mask); /* now the big work begins : do the bit allocation. Modify the snr offset until we can pack everything in the requested frame size */ coarse_snr_offset = s->coarse_snr_offset; while (coarse_snr_offset >= 0 && bit_alloc(s, mask, psd, bap, frame_bits, coarse_snr_offset, 0) < 0) coarse_snr_offset -= SNR_INC1; if (coarse_snr_offset < 0) { av_log(NULL, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n"); return -1; } while (coarse_snr_offset + SNR_INC1 <= 63 && bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset + SNR_INC1, 0) >= 0) { coarse_snr_offset += SNR_INC1; memcpy(bap, bap1, sizeof(bap1)); } while (coarse_snr_offset + 1 <= 63 && bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset + 1, 0) >= 0) { coarse_snr_offset++; memcpy(bap, bap1, sizeof(bap1)); } fine_snr_offset = 0; while (fine_snr_offset + SNR_INC1 <= 15 && bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset, fine_snr_offset + SNR_INC1) >= 0) { fine_snr_offset += SNR_INC1; memcpy(bap, bap1, sizeof(bap1)); } while (fine_snr_offset + 1 <= 15 && bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset, fine_snr_offset + 1) >= 0) { fine_snr_offset++; memcpy(bap, bap1, sizeof(bap1)); } s->coarse_snr_offset = coarse_snr_offset; for (ch = 0; ch < s->channels; ch++) s->fine_snr_offset[ch] = fine_snr_offset; return 0; } /** * Write the AC-3 frame header to the output bitstream. */ static void output_frame_header(AC3EncodeContext *s, unsigned char *frame) { init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE); put_bits(&s->pb, 16, 0x0b77); /* frame header */ put_bits(&s->pb, 16, 0); /* crc1: will be filled later */ put_bits(&s->pb, 2, s->bit_alloc.sr_code); put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min) / 2); put_bits(&s->pb, 5, s->bitstream_id); put_bits(&s->pb, 3, s->bitstream_mode); put_bits(&s->pb, 3, s->channel_mode); if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO) put_bits(&s->pb, 2, 1); /* XXX -4.5 dB */ if (s->channel_mode & 0x04) put_bits(&s->pb, 2, 1); /* XXX -6 dB */ if (s->channel_mode == AC3_CHMODE_STEREO) put_bits(&s->pb, 2, 0); /* surround not indicated */ put_bits(&s->pb, 1, s->lfe_on); /* LFE */ put_bits(&s->pb, 5, 31); /* dialog norm: -31 db */ put_bits(&s->pb, 1, 0); /* no compression control word */ put_bits(&s->pb, 1, 0); /* no lang code */ put_bits(&s->pb, 1, 0); /* no audio production info */ put_bits(&s->pb, 1, 0); /* no copyright */ put_bits(&s->pb, 1, 1); /* original bitstream */ put_bits(&s->pb, 1, 0); /* no time code 1 */ put_bits(&s->pb, 1, 0); /* no time code 2 */ put_bits(&s->pb, 1, 0); /* no additional bit stream info */ } /** * Symmetric quantization on 'levels' levels. */ static inline int sym_quant(int c, int e, int levels) { int v; if (c >= 0) { v = (levels * (c << e)) >> 24; v = (v + 1) >> 1; v = (levels >> 1) + v; } else { v = (levels * ((-c) << e)) >> 24; v = (v + 1) >> 1; v = (levels >> 1) - v; } assert (v >= 0 && v < levels); return v; } /** * Asymmetric quantization on 2^qbits levels. */ static inline int asym_quant(int c, int e, int qbits) { int lshift, m, v; lshift = e + qbits - 24; if (lshift >= 0) v = c << lshift; else v = c >> (-lshift); /* rounding */ v = (v + 1) >> 1; m = (1 << (qbits-1)); if (v >= m) v = m - 1; assert(v >= -m); return v & ((1 << qbits)-1); } /** * Write one audio block to the output bitstream. */ static void output_audio_block(AC3EncodeContext *s, uint8_t exp_strategy[AC3_MAX_CHANNELS], uint8_t encoded_exp[AC3_MAX_CHANNELS][AC3_MAX_COEFS], uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS], int32_t mdct_coef[AC3_MAX_CHANNELS][AC3_MAX_COEFS], int8_t exp_shift[AC3_MAX_CHANNELS], int block_num) { int ch, nb_groups, group_size, i, baie, rbnd; uint8_t *p; uint16_t qmant[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; int exp0, exp1; int mant1_cnt, mant2_cnt, mant4_cnt; uint16_t *qmant1_ptr, *qmant2_ptr, *qmant4_ptr; int delta0, delta1, delta2; for (ch = 0; ch < s->fbw_channels; ch++) put_bits(&s->pb, 1, 0); /* no block switching */ for (ch = 0; ch < s->fbw_channels; ch++) put_bits(&s->pb, 1, 1); /* no dither */ put_bits(&s->pb, 1, 0); /* no dynamic range */ if (!block_num) { put_bits(&s->pb, 1, 1); /* coupling strategy present */ put_bits(&s->pb, 1, 0); /* no coupling strategy */ } else { put_bits(&s->pb, 1, 0); /* no new coupling strategy */ } if (s->channel_mode == AC3_CHMODE_STEREO) { if (!block_num) { /* first block must define rematrixing (rematstr) */ put_bits(&s->pb, 1, 1); /* dummy rematrixing rematflg(1:4)=0 */ for (rbnd = 0; rbnd < 4; rbnd++) put_bits(&s->pb, 1, 0); } else { /* no matrixing (but should be used in the future) */ put_bits(&s->pb, 1, 0); } } /* exponent strategy */ for (ch = 0; ch < s->fbw_channels; ch++) put_bits(&s->pb, 2, exp_strategy[ch]); if (s->lfe_on) put_bits(&s->pb, 1, exp_strategy[s->lfe_channel]); /* bandwidth */ for (ch = 0; ch < s->fbw_channels; ch++) { if (exp_strategy[ch] != EXP_REUSE) put_bits(&s->pb, 6, s->bandwidth_code[ch]); } /* exponents */ for (ch = 0; ch < s->channels; ch++) { if (exp_strategy[ch] == EXP_REUSE) continue; group_size = exp_strategy[ch] + (exp_strategy[ch] == EXP_D45); nb_groups = (s->nb_coefs[ch] + (group_size * 3) - 4) / (3 * group_size); p = encoded_exp[ch]; /* first exponent */ exp1 = *p++; put_bits(&s->pb, 4, exp1); /* next ones are delta encoded */ for (i = 0; i < nb_groups; i++) { /* merge three delta in one code */ exp0 = exp1; exp1 = p[0]; p += group_size; delta0 = exp1 - exp0 + 2; exp0 = exp1; exp1 = p[0]; p += group_size; delta1 = exp1 - exp0 + 2; exp0 = exp1; exp1 = p[0]; p += group_size; delta2 = exp1 - exp0 + 2; put_bits(&s->pb, 7, ((delta0 * 5 + delta1) * 5) + delta2); } if (ch != s->lfe_channel) put_bits(&s->pb, 2, 0); /* no gain range info */ } /* bit allocation info */ baie = (block_num == 0); put_bits(&s->pb, 1, baie); if (baie) { put_bits(&s->pb, 2, s->slow_decay_code); put_bits(&s->pb, 2, s->fast_decay_code); put_bits(&s->pb, 2, s->slow_gain_code); put_bits(&s->pb, 2, s->db_per_bit_code); put_bits(&s->pb, 3, s->floor_code); } /* snr offset */ put_bits(&s->pb, 1, baie); if (baie) { put_bits(&s->pb, 6, s->coarse_snr_offset); for (ch = 0; ch < s->channels; ch++) { put_bits(&s->pb, 4, s->fine_snr_offset[ch]); put_bits(&s->pb, 3, s->fast_gain_code[ch]); } } put_bits(&s->pb, 1, 0); /* no delta bit allocation */ put_bits(&s->pb, 1, 0); /* no data to skip */ /* mantissa encoding : we use two passes to handle the grouping. A one pass method may be faster, but it would necessitate to modify the output stream. */ /* first pass: quantize */ mant1_cnt = mant2_cnt = mant4_cnt = 0; qmant1_ptr = qmant2_ptr = qmant4_ptr = NULL; for (ch = 0; ch < s->channels; ch++) { int b, c, e, v; for (i = 0; i < s->nb_coefs[ch]; i++) { c = mdct_coef[ch][i]; e = encoded_exp[ch][i] - exp_shift[ch]; b = bap[ch][i]; switch (b) { case 0: v = 0; break; case 1: v = sym_quant(c, e, 3); switch (mant1_cnt) { case 0: qmant1_ptr = &qmant[ch][i]; v = 9 * v; mant1_cnt = 1; break; case 1: *qmant1_ptr += 3 * v; mant1_cnt = 2; v = 128; break; default: *qmant1_ptr += v; mant1_cnt = 0; v = 128; break; } break; case 2: v = sym_quant(c, e, 5); switch (mant2_cnt) { case 0: qmant2_ptr = &qmant[ch][i]; v = 25 * v; mant2_cnt = 1; break; case 1: *qmant2_ptr += 5 * v; mant2_cnt = 2; v = 128; break; default: *qmant2_ptr += v; mant2_cnt = 0; v = 128; break; } break; case 3: v = sym_quant(c, e, 7); break; case 4: v = sym_quant(c, e, 11); switch (mant4_cnt) { case 0: qmant4_ptr = &qmant[ch][i]; v = 11 * v; mant4_cnt = 1; break; default: *qmant4_ptr += v; mant4_cnt = 0; v = 128; break; } break; case 5: v = sym_quant(c, e, 15); break; case 14: v = asym_quant(c, e, 14); break; case 15: v = asym_quant(c, e, 16); break; default: v = asym_quant(c, e, b - 1); break; } qmant[ch][i] = v; } } /* second pass : output the values */ for (ch = 0; ch < s->channels; ch++) { int b, q; for (i = 0; i < s->nb_coefs[ch]; i++) { q = qmant[ch][i]; b = bap[ch][i]; switch (b) { case 0: break; case 1: if (q != 128) put_bits(&s->pb, 5, q); break; case 2: if (q != 128) put_bits(&s->pb, 7, q); break; case 3: put_bits(&s->pb, 3, q); break; case 4: if (q != 128) put_bits(&s->pb, 7, q); break; case 14: put_bits(&s->pb, 14, q); break; case 15: put_bits(&s->pb, 16, q); break; default: put_bits(&s->pb, b-1, q); break; } } } } /** CRC-16 Polynomial */ #define CRC16_POLY ((1 << 0) | (1 << 2) | (1 << 15) | (1 << 16)) static unsigned int mul_poly(unsigned int a, unsigned int b, unsigned int poly) { unsigned int c; c = 0; while (a) { if (a & 1) c ^= b; a = a >> 1; b = b << 1; if (b & (1 << 16)) b ^= poly; } return c; } static unsigned int pow_poly(unsigned int a, unsigned int n, unsigned int poly) { unsigned int r; r = 1; while (n) { if (n & 1) r = mul_poly(r, a, poly); a = mul_poly(a, a, poly); n >>= 1; } return r; } /** * Fill the end of the frame with 0's and compute the two CRCs. */ static void output_frame_end(AC3EncodeContext *s) { int frame_size, frame_size_58, pad_bytes, crc1, crc2, crc_inv; uint8_t *frame; frame_size = s->frame_size; /* frame size in words */ /* align to 8 bits */ flush_put_bits(&s->pb); /* add zero bytes to reach the frame size */ frame = s->pb.buf; pad_bytes = s->frame_size - (put_bits_ptr(&s->pb) - frame) - 2; assert(pad_bytes >= 0); if (pad_bytes > 0) memset(put_bits_ptr(&s->pb), 0, pad_bytes); /* Now we must compute both crcs : this is not so easy for crc1 because it is at the beginning of the data... */ frame_size_58 = ((frame_size >> 2) + (frame_size >> 4)) << 1; crc1 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, frame + 4, frame_size_58 - 4)); /* XXX: could precompute crc_inv */ crc_inv = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY); crc1 = mul_poly(crc_inv, crc1, CRC16_POLY); AV_WB16(frame + 2, crc1); crc2 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, frame + frame_size_58, frame_size - frame_size_58 - 2)); AV_WB16(frame + frame_size - 2, crc2); } /** * Encode a single AC-3 frame. */ static int ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame, int buf_size, void *data) { AC3EncodeContext *s = avctx->priv_data; const int16_t *samples = data; int v; int blk, blk1, blk2, ch, i; int16_t planar_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE+AC3_FRAME_SIZE]; int16_t windowed_samples[AC3_WINDOW_SIZE]; int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS]; uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS]; uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS]; uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS]; uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS]; int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS]; int frame_bits; /* deinterleave and remap input samples */ for (ch = 0; ch < s->channels; ch++) { const int16_t *sptr; int sinc; /* copy last 256 samples of previous frame to the start of the current frame */ memcpy(&planar_samples[ch][0], s->last_samples[ch], AC3_BLOCK_SIZE * sizeof(planar_samples[0][0])); /* deinterleave */ sinc = s->channels; sptr = samples + s->channel_map[ch]; for (i = AC3_BLOCK_SIZE; i < AC3_FRAME_SIZE+AC3_BLOCK_SIZE; i++) { planar_samples[ch][i] = *sptr; sptr += sinc; } /* save last 256 samples for next frame */ memcpy(s->last_samples[ch], &planar_samples[ch][6* AC3_BLOCK_SIZE], AC3_BLOCK_SIZE * sizeof(planar_samples[0][0])); } frame_bits = 0; for (ch = 0; ch < s->channels; ch++) { /* fixed mdct to the six sub blocks & exponent computation */ for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { int16_t *input_samples = &planar_samples[ch][blk * AC3_BLOCK_SIZE]; /* apply the MDCT window */ for (i = 0; i < AC3_BLOCK_SIZE; i++) { windowed_samples[i] = MUL16(input_samples[i], ff_ac3_window[i]) >> 15; windowed_samples[AC3_WINDOW_SIZE-i-1] = MUL16(input_samples[AC3_WINDOW_SIZE-i-1], ff_ac3_window[i]) >> 15; } /* Normalize the samples to use the maximum available precision */ v = 14 - log2_tab(windowed_samples, AC3_WINDOW_SIZE); if (v < 0) v = 0; exp_shift[blk][ch] = v - 9; lshift_tab(windowed_samples, AC3_WINDOW_SIZE, v); /* do the MDCT */ mdct512(mdct_coef[blk][ch], windowed_samples); /* compute "exponents". We take into account the normalization there */ for (i = 0; i < AC3_MAX_COEFS; i++) { int e; v = abs(mdct_coef[blk][ch][i]); if (v == 0) e = 24; else { e = 23 - av_log2(v) + exp_shift[blk][ch]; if (e >= 24) { e = 24; mdct_coef[blk][ch][i] = 0; } } exp[blk][ch][i] = e; } } compute_exp_strategy_ch(exp_strategy, exp, ch, ch == s->lfe_channel); /* compute the exponents as the decoder will see them. The EXP_REUSE case must be handled carefully : we select the min of the exponents */ blk = 0; while (blk < AC3_MAX_BLOCKS) { blk1 = blk + 1; while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE) { exponent_min(exp[blk][ch], exp[blk1][ch], s->nb_coefs[ch]); blk1++; } frame_bits += encode_exponents_blk_ch(encoded_exp[blk][ch], exp[blk][ch], s->nb_coefs[ch], exp_strategy[blk][ch]); /* copy encoded exponents for reuse case */ for (blk2 = blk+1; blk2 < blk1; blk2++) { memcpy(encoded_exp[blk2][ch], encoded_exp[blk][ch], s->nb_coefs[ch] * sizeof(uint8_t)); } blk = blk1; } } /* adjust for fractional frame sizes */ while (s->bits_written >= s->bit_rate && s->samples_written >= s->sample_rate) { s->bits_written -= s->bit_rate; s->samples_written -= s->sample_rate; } s->frame_size = s->frame_size_min + 2 * (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate); s->bits_written += s->frame_size * 8; s->samples_written += AC3_FRAME_SIZE; compute_bit_allocation(s, bap, encoded_exp, exp_strategy, frame_bits); /* everything is known... let's output the frame */ output_frame_header(s, frame); for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { output_audio_block(s, exp_strategy[blk], encoded_exp[blk], bap[blk], mdct_coef[blk], exp_shift[blk], blk); } output_frame_end(s); return s->frame_size; } /** * Finalize encoding and free any memory allocated by the encoder. */ static av_cold int ac3_encode_close(AVCodecContext *avctx) { av_freep(&avctx->coded_frame); return 0; } /** * Set channel information during initialization. */ static av_cold int set_channel_info(AC3EncodeContext *s, int channels, int64_t *channel_layout) { int ch_layout; if (channels < 1 || channels > AC3_MAX_CHANNELS) return AVERROR(EINVAL); if ((uint64_t)*channel_layout > 0x7FF) return AVERROR(EINVAL); ch_layout = *channel_layout; if (!ch_layout) ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL); if (av_get_channel_layout_nb_channels(ch_layout) != channels) return AVERROR(EINVAL); s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY); s->channels = channels; s->fbw_channels = channels - s->lfe_on; s->lfe_channel = s->lfe_on ? s->fbw_channels : -1; if (s->lfe_on) ch_layout -= AV_CH_LOW_FREQUENCY; switch (ch_layout) { case AV_CH_LAYOUT_MONO: s->channel_mode = AC3_CHMODE_MONO; break; case AV_CH_LAYOUT_STEREO: s->channel_mode = AC3_CHMODE_STEREO; break; case AV_CH_LAYOUT_SURROUND: s->channel_mode = AC3_CHMODE_3F; break; case AV_CH_LAYOUT_2_1: s->channel_mode = AC3_CHMODE_2F1R; break; case AV_CH_LAYOUT_4POINT0: s->channel_mode = AC3_CHMODE_3F1R; break; case AV_CH_LAYOUT_QUAD: case AV_CH_LAYOUT_2_2: s->channel_mode = AC3_CHMODE_2F2R; break; case AV_CH_LAYOUT_5POINT0: case AV_CH_LAYOUT_5POINT0_BACK: s->channel_mode = AC3_CHMODE_3F2R; break; default: return AVERROR(EINVAL); } s->channel_map = ff_ac3_enc_channel_map[s->channel_mode][s->lfe_on]; *channel_layout = ch_layout; if (s->lfe_on) *channel_layout |= AV_CH_LOW_FREQUENCY; return 0; } static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s) { int i, ret; /* validate channel layout */ if (!avctx->channel_layout) { av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The " "encoder will guess the layout, but it " "might be incorrect.\n"); } ret = set_channel_info(s, avctx->channels, &avctx->channel_layout); if (ret) { av_log(avctx, AV_LOG_ERROR, "invalid channel layout\n"); return ret; } /* validate sample rate */ for (i = 0; i < 9; i++) { if ((ff_ac3_sample_rate_tab[i / 3] >> (i % 3)) == avctx->sample_rate) break; } if (i == 9) { av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n"); return AVERROR(EINVAL); } s->sample_rate = avctx->sample_rate; s->bit_alloc.sr_shift = i % 3; s->bit_alloc.sr_code = i / 3; /* validate bit rate */ for (i = 0; i < 19; i++) { if ((ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift)*1000 == avctx->bit_rate) break; } if (i == 19) { av_log(avctx, AV_LOG_ERROR, "invalid bit rate\n"); return AVERROR(EINVAL); } s->bit_rate = avctx->bit_rate; s->frame_size_code = i << 1; return 0; } /** * Set bandwidth for all channels. * The user can optionally supply a cutoff frequency. Otherwise an appropriate * default value will be used. */ static av_cold void set_bandwidth(AC3EncodeContext *s, int cutoff) { int ch, bw_code; if (cutoff) { /* calculate bandwidth based on user-specified cutoff frequency */ int fbw_coeffs; cutoff = av_clip(cutoff, 1, s->sample_rate >> 1); fbw_coeffs = cutoff * 2 * AC3_MAX_COEFS / s->sample_rate; bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60); } else { /* use default bandwidth setting */ /* XXX: should compute the bandwidth according to the frame size, so that we avoid annoying high frequency artifacts */ bw_code = 50; } /* set number of coefficients for each channel */ for (ch = 0; ch < s->fbw_channels; ch++) { s->bandwidth_code[ch] = bw_code; s->nb_coefs[ch] = bw_code * 3 + 73; } if (s->lfe_on) s->nb_coefs[s->lfe_channel] = 7; /* LFE channel always has 7 coefs */ } /** * Initialize the encoder. */ static av_cold int ac3_encode_init(AVCodecContext *avctx) { AC3EncodeContext *s = avctx->priv_data; int ret; avctx->frame_size = AC3_FRAME_SIZE; ac3_common_init(); ret = validate_options(avctx, s); if (ret) return ret; s->bitstream_id = 8 + s->bit_alloc.sr_shift; s->bitstream_mode = 0; /* complete main audio service */ s->frame_size_min = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code]; s->bits_written = 0; s->samples_written = 0; s->frame_size = s->frame_size_min; set_bandwidth(s, avctx->cutoff); /* initial snr offset */ s->coarse_snr_offset = 40; mdct_init(9); avctx->coded_frame= avcodec_alloc_frame(); avctx->coded_frame->key_frame= 1; return 0; } #ifdef TEST /*************************************************************************/ /* TEST */ #include "libavutil/lfg.h" #define FN (MDCT_SAMPLES/4) static void fft_test(AVLFG *lfg) { IComplex in[FN], in1[FN]; int k, n, i; float sum_re, sum_im, a; for (i = 0; i < FN; i++) { in[i].re = av_lfg_get(lfg) % 65535 - 32767; in[i].im = av_lfg_get(lfg) % 65535 - 32767; in1[i] = in[i]; } fft(in, 7); /* do it by hand */ for (k = 0; k < FN; k++) { sum_re = 0; sum_im = 0; for (n = 0; n < FN; n++) { a = -2 * M_PI * (n * k) / FN; sum_re += in1[n].re * cos(a) - in1[n].im * sin(a); sum_im += in1[n].re * sin(a) + in1[n].im * cos(a); } av_log(NULL, AV_LOG_DEBUG, "%3d: %6d,%6d %6.0f,%6.0f\n", k, in[k].re, in[k].im, sum_re / FN, sum_im / FN); } } static void mdct_test(AVLFG *lfg) { int16_t input[MDCT_SAMPLES]; int32_t output[AC3_MAX_COEFS]; float input1[MDCT_SAMPLES]; float output1[AC3_MAX_COEFS]; float s, a, err, e, emax; int i, k, n; for (i = 0; i < MDCT_SAMPLES; i++) { input[i] = (av_lfg_get(lfg) % 65535 - 32767) * 9 / 10; input1[i] = input[i]; } mdct512(output, input); /* do it by hand */ for (k = 0; k < AC3_MAX_COEFS; k++) { s = 0; for (n = 0; n < MDCT_SAMPLES; n++) { a = (2*M_PI*(2*n+1+MDCT_SAMPLES/2)*(2*k+1) / (4 * MDCT_SAMPLES)); s += input1[n] * cos(a); } output1[k] = -2 * s / MDCT_SAMPLES; } err = 0; emax = 0; for (i = 0; i < AC3_MAX_COEFS; i++) { av_log(NULL, AV_LOG_DEBUG, "%3d: %7d %7.0f\n", i, output[i], output1[i]); e = output[i] - output1[i]; if (e > emax) emax = e; err += e * e; } av_log(NULL, AV_LOG_DEBUG, "err2=%f emax=%f\n", err / AC3_MAX_COEFS, emax); } int main(void) { AVLFG lfg; av_log_set_level(AV_LOG_DEBUG); mdct_init(9); fft_test(&lfg); mdct_test(&lfg); return 0; } #endif /* TEST */ AVCodec ac3_encoder = { "ac3", AVMEDIA_TYPE_AUDIO, CODEC_ID_AC3, sizeof(AC3EncodeContext), ac3_encode_init, ac3_encode_frame, ac3_encode_close, NULL, .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), .channel_layouts = (const int64_t[]){ AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_2_1, AV_CH_LAYOUT_SURROUND, AV_CH_LAYOUT_2_2, AV_CH_LAYOUT_QUAD, AV_CH_LAYOUT_4POINT0, AV_CH_LAYOUT_5POINT0, AV_CH_LAYOUT_5POINT0_BACK, (AV_CH_LAYOUT_MONO | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_STEREO | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_2_1 | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_SURROUND | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_2_2 | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_QUAD | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_4POINT0 | AV_CH_LOW_FREQUENCY), AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_5POINT1_BACK, 0 }, };