Mercurial > libavcodec.hg
view ra144.c @ 11588:480a999ce2c4 libavcodec
Add support for PS sync extensions.
author | alexc |
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date | Thu, 08 Apr 2010 06:04:41 +0000 |
parents | 8a4984c5cacc |
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/* * Real Audio 1.0 (14.4K) * * Copyright (c) 2008 Vitor Sessak * Copyright (c) 2003 Nick Kurshev * Based on public domain decoder at http://www.honeypot.net/audio * * 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 "libavutil/intmath.h" #include "avcodec.h" #include "get_bits.h" #include "ra144.h" #include "celp_filters.h" #define NBLOCKS 4 ///< number of subblocks within a block #define BLOCKSIZE 40 ///< subblock size in 16-bit words #define BUFFERSIZE 146 ///< the size of the adaptive codebook typedef struct { AVCodecContext *avctx; unsigned int old_energy; ///< previous frame energy unsigned int lpc_tables[2][10]; /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame * and lpc_coef[1] of the previous one. */ unsigned int *lpc_coef[2]; unsigned int lpc_refl_rms[2]; /** The current subblock padded by the last 10 values of the previous one. */ int16_t curr_sblock[50]; /** Adaptive codebook, its size is two units bigger to avoid a * buffer overflow. */ uint16_t adapt_cb[146+2]; } RA144Context; static av_cold int ra144_decode_init(AVCodecContext * avctx) { RA144Context *ractx = avctx->priv_data; ractx->avctx = avctx; ractx->lpc_coef[0] = ractx->lpc_tables[0]; ractx->lpc_coef[1] = ractx->lpc_tables[1]; avctx->sample_fmt = SAMPLE_FMT_S16; 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 = 2; while (x > 0xfff) { s++; x >>= 2; } return ff_sqrt(x << 20) << s; } /** * Evaluate the LPC filter coefficients from the reflection coefficients. * Does the inverse of the eval_refl() function. */ static void eval_coefs(int *coefs, const int *refl) { int buffer[10]; int *b1 = buffer; int *b2 = coefs; int i, j; for (i=0; i < 10; i++) { b1[i] = refl[i] << 4; for (j=0; j < i; j++) b1[j] = ((refl[i] * b2[i-j-1]) >> 12) + b2[j]; FFSWAP(int *, b1, b2); } for (i=0; i < 10; i++) coefs[i] >>= 4; } /** * Copy the last offset values of *source to *target. If those values are not * enough to fill the target buffer, fill it with another copy of those values. */ static void copy_and_dup(int16_t *target, const int16_t *source, int offset) { source += BUFFERSIZE - offset; memcpy(target, source, FFMIN(BLOCKSIZE, offset)*sizeof(*target)); if (offset < BLOCKSIZE) memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target)); } /** inverse root mean square */ static int irms(const int16_t *data) { unsigned int i, sum = 0; for (i=0; i < BLOCKSIZE; i++) sum += data[i] * data[i]; if (sum == 0) return 0; /* OOPS - division by zero */ return 0x20000000 / (t_sqrt(sum) >> 8); } static void add_wav(int16_t *dest, int n, int skip_first, int *m, const int16_t *s1, const int8_t *s2, const int8_t *s3) { 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]; if (v[0]) { for (i=0; i < BLOCKSIZE; i++) dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12; } else { for (i=0; i < BLOCKSIZE; i++) dest[i] = ( s2[i]*v[1] + s3[i]*v[2]) >> 12; } } static unsigned int rescale_rms(unsigned int rms, unsigned int energy) { return (rms * energy) >> 10; } static unsigned int rms(const int *data) { int i; unsigned int res = 0x10000; int b = 10; for (i=0; i < 10; i++) { res = (((0x1000000 - data[i]*data[i]) >> 12) * res) >> 12; if (res == 0) return 0; while (res <= 0x3fff) { b++; res <<= 2; } } return t_sqrt(res) >> b; } static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs, int gval, 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 += BLOCKSIZE/2 - 1; copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx); m[0] = (irms(buffer_a) * gval) >> 12; } else { m[0] = 0; } m[1] = (cb1_base[cb1_idx] * gval) >> 8; m[2] = (cb2_base[cb2_idx] * gval) >> 8; memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE, (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb)); block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE; add_wav(block, gain, cba_idx, m, cba_idx? buffer_a: NULL, cb1_vects[cb1_idx], cb2_vects[cb2_idx]); memcpy(ractx->curr_sblock, ractx->curr_sblock + 40, 10*sizeof(*ractx->curr_sblock)); if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs, block, BLOCKSIZE, 10, 1, 0xfff)) memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock)); } static void int_to_int16(int16_t *out, const int *inp) { int i; for (i=0; i < 10; 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 greater than * 4095, 0 if not. */ static int eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx) { int b, i, j; int buffer1[10]; int buffer2[10]; int *bp1 = buffer1; int *bp2 = buffer2; for (i=0; i < 10; i++) buffer2[i] = coefs[i]; refl[9] = bp2[9]; if ((unsigned) bp2[9] + 0x1000 > 0x1fff) { av_log(avctx, AV_LOG_ERROR, "Overflow. Broken sample?\n"); return 1; } for (i=8; i >= 0; i--) { b = 0x1000-((bp2[i+1] * bp2[i+1]) >> 12); if (!b) b = -2; for (j=0; j <= i; j++) bp1[j] = ((bp2[j] - ((refl[i+1] * bp2[i-j]) >> 12)) * (0x1000000 / b)) >> 12; if ((unsigned) bp1[i] + 0x1000 > 0x1fff) return 1; refl[i] = bp1[i]; FFSWAP(int *, bp1, bp2); } return 0; } static int interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy) { int work[10]; int b = NBLOCKS - a; int i; // Interpolate block coefficients from the this frame's forth block and // last frame's forth block. for (i=0; i<10; i++) out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2; if (eval_refl(work, out, ractx->avctx)) { // The interpolated coefficients are unstable, copy either new or old // coefficients. int_to_int16(out, ractx->lpc_coef[copyold]); return rescale_rms(ractx->lpc_refl_rms[copyold], 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, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->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][10]; // LPC coefficients of each sub-block unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame int i, j; int16_t *data = vdata; unsigned int energy; RA144Context *ractx = avctx->priv_data; GetBitContext gb; if (*data_size < 2*160) return -1; 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++) lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])]; eval_coefs(ractx->lpc_coef[0], lpc_refl); ractx->lpc_refl_rms[0] = rms(lpc_refl); energy = energy_tab[get_bits(&gb, 5)]; refl_rms[0] = interp(ractx, block_coefs[0], 1, 1, ractx->old_energy); refl_rms[1] = interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy, t_sqrt(energy*ractx->old_energy) >> 12); refl_rms[2] = interp(ractx, block_coefs[2], 3, 0, energy); refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy); int_to_int16(block_coefs[3], ractx->lpc_coef[0]); for (i=0; i < 4; i++) { do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb); for (j=0; j < BLOCKSIZE; j++) *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2); } ractx->old_energy = energy; ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0]; FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); *data_size = 2*160; return 20; } AVCodec ra_144_decoder = { "real_144", AVMEDIA_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)"), };