/* * Real Audio 1.0 (14.4K) * Copyright (c) 2003 the ffmpeg project * * 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 */ #include "avcodec.h" #include "bitstream.h" #include "ra144.h" #define NBLOCKS 4 /* number of segments within a block */ #define BLOCKSIZE 40 /* (quarter) block size in 16-bit words (80 bytes) */ #define HALFBLOCK 20 /* BLOCKSIZE/2 */ #define BUFFERSIZE 146 /* for do_output */ typedef struct { unsigned int old_energy; ///< previous frame energy /* the swapped buffers */ unsigned int lpc_tables[2][10]; unsigned int *lpc_coef; ///< LPC coefficients unsigned int *lpc_coef_old; ///< previous frame LPC coefficients unsigned int lpc_refl_rms; unsigned int lpc_refl_rms_old; unsigned int buffer[5]; uint16_t adapt_cb[148]; ///< adaptive codebook } RA144Context; static int ra144_decode_init(AVCodecContext * avctx) { RA144Context *ractx = avctx->priv_data; ractx->lpc_coef = ractx->lpc_tables[0]; ractx->lpc_coef_old = ractx->lpc_tables[1]; return 0; } /** * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an * odd way to make the output identical to the binary decoder. */ static int t_sqrt(unsigned int x) { int s = 0; while (x > 0xfff) { s++; x = x >> 2; } return (ff_sqrt(x << 20) << s) << 2; } /** * Evaluate the LPC filter coefficients from the reflection coefficients. * Does the inverse of the eval_refl() function. */ static void eval_coefs(const int *refl, int *coefs) { int buffer[10]; int *b1 = buffer; int *b2 = coefs; int x, y; for (x=0; x < 10; x++) { b1[x] = refl[x] << 4; for (y=0; y < x; y++) b1[y] = ((refl[x] * b2[x-y-1]) >> 12) + b2[y]; FFSWAP(int *, b1, b2); } for (x=0; x < 10; x++) coefs[x] >>= 4; } /* rotate block */ static void rotate_block(const int16_t *source, int16_t *target, int offset) { int i=0, k=0; source += BUFFERSIZE - offset; while (i> 8)) * factor; } /* multiply/add wavetable */ static void add_wav(int n, int skip_first, int *m, const int16_t *s1, const int8_t *s2, const int8_t *s3, int16_t *dest) { int i; int v[3]; v[0] = 0; for (i=!skip_first; i<3; i++) v[i] = (gain_val_tab[n][i] * m[i]) >> (gain_exp_tab[n][i] + 1); for (i=0; i < BLOCKSIZE; i++) dest[i] = ((*(s1++))*v[0] + (*(s2++))*v[1] + (*(s3++))*v[2]) >> 12; } static void lpc_filter(const int16_t *lpc_coefs, const int16_t *adapt_coef, void *out, int *statbuf, int len) { int x, i; uint16_t work[50]; int16_t *ptr = work; memcpy(work, statbuf,20); memcpy(work + 10, adapt_coef, len * 2); for (i=0; i>= 12; new_val = ptr[10] - sum; if (new_val < -32768 || new_val > 32767) { memset(out, 0, len * 2); memset(statbuf, 0, 20); return; } ptr[10] = new_val; ptr++; } memcpy(out, work+10, len * 2); memcpy(statbuf, work + 40, 20); } static unsigned int rescale_rms(int rms, int energy) { return (rms * energy) >> 10; } static unsigned int rms(const int *data) { int x; unsigned int res = 0x10000; int b = 0; for (x=0; x<10; x++) { res = (((0x1000000 - (*data) * (*data)) >> 12) * res) >> 12; if (res == 0) return 0; while (res <= 0x3fff) { b++; res <<= 2; } data++; } if (res > 0) res = t_sqrt(res); res >>= (b + 10); return res; } /* do quarter-block output */ static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs, unsigned int gval, int16_t *output_buffer, GetBitContext *gb) { uint16_t buffer_a[40]; uint16_t *block; int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none int gain = get_bits(gb, 8); int cb1_idx = get_bits(gb, 7); int cb2_idx = get_bits(gb, 7); int m[3]; if (cba_idx) { cba_idx += HALFBLOCK - 1; rotate_block(ractx->adapt_cb, buffer_a, cba_idx); m[0] = irms(buffer_a, gval) >> 12; } else { m[0] = 0; } m[1] = ((cb1_base[cb1_idx] >> 4) * gval) >> 8; m[2] = ((cb2_base[cb2_idx] >> 4) * gval) >> 8; memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE, (BUFFERSIZE - BLOCKSIZE) * 2); block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE; add_wav(gain, cba_idx, m, buffer_a, cb1_vects[cb1_idx], cb2_vects[cb2_idx], block); lpc_filter(lpc_coefs, block, output_buffer, ractx->buffer, BLOCKSIZE); } static void int_to_int16(int16_t *out, const int *inp) { int i; for (i=0; i<30; i++) *(out++) = *(inp++); } /** * Evaluate the reflection coefficients from the filter coefficients. * Does the inverse of the eval_coefs() function. * * @return 1 if one of the reflection coefficients is of magnitude greater than * 4095, 0 if not. */ static int eval_refl(const int16_t *coefs, int *refl, RA144Context *ractx) { int retval = 0; int b, c, i; unsigned int u; int buffer1[10]; int buffer2[10]; int *bp1 = buffer1; int *bp2 = buffer2; for (i=0; i < 10; i++) buffer2[i] = coefs[i]; u = refl[9] = bp2[9]; if (u + 0x1000 > 0x1fff) { av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n"); return 0; } for (c=8; c >= 0; c--) { if (u == 0x1000) u++; if (u == 0xfffff000) u--; b = 0x1000-((u * u) >> 12); if (b == 0) b++; for (u=0; u<=c; u++) bp1[u] = ((bp2[u] - ((refl[c+1] * bp2[c-u]) >> 12)) * (0x1000000 / b)) >> 12; refl[c] = u = bp1[c]; if ((u + 0x1000) > 0x1fff) retval = 1; FFSWAP(int *, bp1, bp2); } return retval; } static int interp(RA144Context *ractx, int16_t *out, int block_num, int copynew, int energy) { int work[10]; int a = block_num + 1; int b = NBLOCKS - a; int x; // Interpolate block coefficients from the this frame forth block and // last frame forth block for (x=0; x<30; x++) out[x] = (a * ractx->lpc_coef[x] + b * ractx->lpc_coef_old[x])>> 2; if (eval_refl(out, work, ractx)) { // The interpolated coefficients are unstable, copy either new or old // coefficients if (copynew) { int_to_int16(out, ractx->lpc_coef); return rescale_rms(ractx->lpc_refl_rms, energy); } else { int_to_int16(out, ractx->lpc_coef_old); return rescale_rms(ractx->lpc_refl_rms_old, energy); } } else { return rescale_rms(rms(work), energy); } } /* Uncompress one block (20 bytes -> 160*2 bytes) */ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, int *data_size, const uint8_t * buf, int buf_size) { static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; unsigned int refl_rms[4]; // RMS of the reflection coefficients uint16_t block_coefs[4][30]; // LPC coefficients of each sub-block unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame int i, c; int16_t *data = vdata; unsigned int energy; RA144Context *ractx = avctx->priv_data; GetBitContext gb; if(buf_size < 20) { av_log(avctx, AV_LOG_ERROR, "Frame too small (%d bytes). Truncated file?\n", buf_size); *data_size = 0; return buf_size; } init_get_bits(&gb, buf, 20 * 8); for (i=0; i<10; i++) // "<< 1"? Doesn't this make one value out of two of the table useless? lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i]) << 1]; eval_coefs(lpc_refl, ractx->lpc_coef); ractx->lpc_refl_rms = rms(lpc_refl); energy = energy_tab[get_bits(&gb, 5) << 1]; // Useless table entries? refl_rms[0] = interp(ractx, block_coefs[0], 0, 0, ractx->old_energy); refl_rms[1] = interp(ractx, block_coefs[1], 1, energy > ractx->old_energy, t_sqrt(energy*ractx->old_energy) >> 12); refl_rms[2] = interp(ractx, block_coefs[2], 2, 1, energy); refl_rms[3] = rescale_rms(ractx->lpc_refl_rms, energy); int_to_int16(block_coefs[3], ractx->lpc_coef); /* do output */ for (c=0; c<4; c++) { do_output_subblock(ractx, block_coefs[c], refl_rms[c], data, &gb); for (i=0; iold_energy = energy; ractx->lpc_refl_rms_old = ractx->lpc_refl_rms; FFSWAP(unsigned int *, ractx->lpc_coef_old, ractx->lpc_coef); *data_size = 2*160; return 20; } AVCodec ra_144_decoder = { "real_144", CODEC_TYPE_AUDIO, CODEC_ID_RA_144, sizeof(RA144Context), ra144_decode_init, NULL, NULL, ra144_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), };