X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fra288.c;h=2166d8ca6159bb61d880b10f47943308492756b8;hb=99975966c319f8995f20c8db56a5cf77cfeb69ee;hp=6c0d7815c8f41c0265a1a18bf735551fc236ef9d;hpb=f38deb4452e89e397e6b6aeef9dda31c279857fd;p=ffmpeg diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c index 6c0d7815c8f..2166d8ca615 100644 --- a/libavcodec/ra288.c +++ b/libavcodec/ra288.c @@ -2,217 +2,174 @@ * RealAudio 2.0 (28.8K) * Copyright (c) 2003 the ffmpeg project * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avcodec.h" #define ALT_BITSTREAM_READER_LE -#include "bitstream.h" +#include "get_bits.h" #include "ra288.h" +#include "lpc.h" +#include "celp_math.h" +#include "celp_filters.h" -typedef struct { - float history[8]; - float output[40]; - float pr1[36]; - float pr2[10]; - int phase, phasep; - - float st1a[111], st1b[37], st1[37]; - float st2a[38], st2b[11], st2[11]; - float sb[41]; - float lhist[10]; -} Real288_internal; - -/* Decode and produce output */ -static void decode(Real288_internal *glob, int amp_coef, int cb_coef) -{ - unsigned int x, y; - float f; - double sum, sumsum; - float *p1, *p2; - float buffer[5]; - const float *table; +#define MAX_BACKWARD_FILTER_ORDER 36 +#define MAX_BACKWARD_FILTER_LEN 40 +#define MAX_BACKWARD_FILTER_NONREC 35 - for (x=36; x--; glob->sb[x+5] = glob->sb[x]); +#define RA288_BLOCK_SIZE 5 +#define RA288_BLOCKS_PER_FRAME 32 - for (x=5; x--;) { - p1 = glob->sb+x; - p2 = glob->pr1; - for (sum=0, y=36; y--; sum -= (*(++p1))*(*(p2++))); +typedef struct { + float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) + float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) - glob->sb[x] = sum; - } + /** speech data history (spec: SB). + * Its first 70 coefficients are updated only at backward filtering. + */ + float sp_hist[111]; - f = amptable[amp_coef]; - table = codetable + cb_coef * 5; + /// speech part of the gain autocorrelation (spec: REXP) + float sp_rec[37]; - /* convert log and do rms */ - for (sum=32, x=10; x--; sum -= glob->pr2[x] * glob->lhist[x]); + /** log-gain history (spec: SBLG). + * Its first 28 coefficients are updated only at backward filtering. + */ + float gain_hist[38]; - if (sum < 0) - sum = 0; - else if (sum > 60) - sum = 60; + /// recursive part of the gain autocorrelation (spec: REXPLG) + float gain_rec[11]; +} RA288Context; - sumsum = exp(sum * 0.1151292546497) * f; /* pow(10.0,sum/20)*f */ +static av_cold int ra288_decode_init(AVCodecContext *avctx) +{ + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + return 0; +} - for (sum=0, x=5; x--;) { - buffer[x] = table[x] * sumsum; - sum += buffer[x] * buffer[x]; - } +static void apply_window(float *tgt, const float *m1, const float *m2, int n) +{ + while (n--) + *tgt++ = *m1++ * *m2++; +} - if ((sum /= 5) < 1) - sum = 1; +static void convolve(float *tgt, const float *src, int len, int n) +{ + for (; n >= 0; n--) + tgt[n] = ff_dot_productf(src, src - n, len); - /* shift and store */ - for (x=10; --x; glob->lhist[x] = glob->lhist[x-1]); +} - *glob->lhist = glob->history[glob->phase] = 10 * log10(sum) - 32; +static void decode(RA288Context *ractx, float gain, int cb_coef) +{ + int i; + double sumsum; + float sum, buffer[5]; + float *block = ractx->sp_hist + 70 + 36; // current block + float *gain_block = ractx->gain_hist + 28; - for (x=1; x < 5; x++) - for (y=x; y--; buffer[x] -= glob->pr1[x-y-1] * buffer[y]); + memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); - /* output */ - for (x=0; x < 5; x++) { - f = glob->sb[4-x] + buffer[x]; + /* block 46 of G.728 spec */ + sum = 32.; + for (i=0; i < 10; i++) + sum -= gain_block[9-i] * ractx->gain_lpc[i]; - if (f > 4095) - f = 4095; - else if (f < -4095) - f = -4095; + /* block 47 of G.728 spec */ + sum = av_clipf(sum, 0, 60); - glob->output[glob->phasep+x] = glob->sb[4-x] = f; - } -} + /* block 48 of G.728 spec */ + /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ + sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); -/* column multiply */ -static void colmult(float *tgt, float *m1, const float *m2, int n) -{ - while (n--) - *(tgt++) = (*(m1++)) * (*(m2++)); -} + for (i=0; i < 5; i++) + buffer[i] = codetable[cb_coef][i] * sumsum; -static int pred(float *in, float *tgt, int n) -{ - int x, y; - float *p1, *p2; - double f0, f1, f2; - float temp; + sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); - if (in[n] == 0) - return 0; + sum = FFMAX(sum, 1); - if ((f0 = *in) <= 0) - return 0; + /* shift and store */ + memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); - for (x=1 ; ; x++) { - if (n < x) - return 1; - - p1 = in + x; - p2 = tgt; - f1 = *(p1--); - for (y=x; --y; f1 += (*(p1--))*(*(p2++))); - - p1 = tgt + x - 1; - p2 = tgt; - *(p1--) = f2 = -f1/f0; - for (y=x >> 1; y--;) { - temp = *p2 + *p1 * f2; - *(p1--) += *p2 * f2; - *(p2++) = temp; - } - if ((f0 += f1*f2) < 0) - return 0; - } -} + gain_block[9] = 10 * log10(sum) - 32; -/* product sum (lsf) */ -static void prodsum(float *tgt, float *src, int len, int n) -{ - unsigned int x; - float *p1, *p2; - double sum; - - while (n >= 0) { - p1 = (p2 = src) - n; - for (sum=0, x=len; x--; sum += (*p1++) * (*p2++)); - tgt[n--] = sum; - } + ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); } -static void co(int n, int i, int j, float *in, float *out, float *st1, - float *st2, const float *table) +/** + * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. + * + * @param order filter order + * @param n input length + * @param non_rec number of non-recursive samples + * @param out filter output + * @param hist pointer to the input history of the filter + * @param out pointer to the non-recursive part of the output + * @param out2 pointer to the recursive part of the output + * @param window pointer to the windowing function table + */ +static void do_hybrid_window(int order, int n, int non_rec, float *out, + float *hist, float *out2, const float *window) { - int a, b, c; - unsigned int x; - float *fp; - float buffer1[37]; - float buffer2[37]; - float work[111]; - - /* rotate and multiply */ - c = (b = (a = n + i) + j) - i; - fp = st1 + i; - for (x=0; x < b; x++) { - if (x == c) - fp=in; - work[x] = *(table++) * (*(st1++) = *(fp++)); - } + int i; + float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; + float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; + float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC]; - prodsum(buffer1, work + n, i, n); - prodsum(buffer2, work + a, j, n); + apply_window(work, window, hist, order + n + non_rec); - for (x=0;x<=n;x++) { - *st2 = *st2 * (0.5625) + buffer1[x]; - out[x] = *(st2++) + buffer2[x]; + convolve(buffer1, work + order , n , order); + convolve(buffer2, work + order + n, non_rec, order); + + for (i=0; i <= order; i++) { + out2[i] = out2[i] * 0.5625 + buffer1[i]; + out [i] = out2[i] + buffer2[i]; } - *out *= 1.00390625; /* to prevent clipping */ + + /* Multiply by the white noise correcting factor (WNCF). */ + *out *= 257./256.; } -static void update(Real288_internal *glob) +/** + * Backward synthesis filter, find the LPC coefficients from past speech data. + */ +static void backward_filter(float *hist, float *rec, const float *window, + float *lpc, const float *tab, + int order, int n, int non_rec, int move_size) { - int x,y; - float buffer1[40], temp1[37]; - float buffer2[8], temp2[11]; - - for (x=0, y=glob->phasep+5; x < 40; buffer1[x++] = glob->output[(y++)%40]); - - co(36, 40, 35, buffer1, temp1, glob->st1a, glob->st1b, table1); + float temp[MAX_BACKWARD_FILTER_ORDER+1]; - if (pred(temp1, glob->st1, 36)) - colmult(glob->pr1, glob->st1, table1a, 36); + do_hybrid_window(order, n, non_rec, temp, hist, rec, window); - for (x=0, y=glob->phase + 1; x < 8; buffer2[x++] = glob->history[(y++) % 8]); + if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) + apply_window(lpc, lpc, tab, order); - co(10, 8, 20, buffer2, temp2, glob->st2a, glob->st2b, table2); - - if (pred(temp2, glob->st2, 10)) - colmult(glob->pr2, glob->st2, table2a, 10); + memmove(hist, hist + n, move_size*sizeof(*hist)); } -/* Decode a block (celp) */ static int ra288_decode_frame(AVCodecContext * avctx, void *data, - int *data_size, const uint8_t * buf, - int buf_size) + int *data_size, AVPacket *avpkt) { - int16_t *out = data; - int x, y; - Real288_internal *glob = avctx->priv_data; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + float *out = data; + int i, j, out_size; + RA288Context *ractx = avctx->priv_data; GetBitContext gb; if (buf_size < avctx->block_align) { @@ -222,33 +179,43 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data, return 0; } + out_size = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME * + av_get_bytes_per_sample(avctx->sample_fmt); + if (*data_size < out_size) { + av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); + return AVERROR(EINVAL); + } + init_get_bits(&gb, buf, avctx->block_align * 8); - for (x=0; x < 32; x++) { - int amp_coef = get_bits(&gb, 3); - int cb_coef = get_bits(&gb, 6 + (x&1)); - glob->phasep = (glob->phase = x & 7) * 5; - decode(glob, amp_coef, cb_coef); + for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) { + float gain = amptable[get_bits(&gb, 3)]; + int cb_coef = get_bits(&gb, 6 + (i&1)); + + decode(ractx, gain, cb_coef); + + for (j=0; j < RA288_BLOCK_SIZE; j++) + *(out++) = ractx->sp_hist[70 + 36 + j]; - for (y=0; y<5; *(out++) = 8 * glob->output[glob->phasep+(y++)]); + if ((i & 7) == 3) { + backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, + ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); - if (glob->phase == 3) - update(glob); + backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, + ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); + } } - *data_size = (char *)out - (char *)data; + *data_size = out_size; return avctx->block_align; } -AVCodec ra_288_decoder = -{ - "real_288", - CODEC_TYPE_AUDIO, - CODEC_ID_RA_288, - sizeof(Real288_internal), - NULL, - NULL, - NULL, - ra288_decode_frame, - .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), +AVCodec ff_ra_288_decoder = { + .name = "real_288", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_RA_288, + .priv_data_size = sizeof(RA288Context), + .init = ra288_decode_init, + .decode = ra288_decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), };