X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fra288.c;h=ddbda1de4cc3d50b2d58dc27496cfcb7080e0ed8;hb=2467d8d9eaae52eb8e18e276a44a15d1d8fd7f97;hp=228125825d00a606f49795d48bd78b101b4619e7;hpb=a3896c63775e82d2d06766db682e816872a8c0e1;p=ffmpeg diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c index 228125825d0..ddbda1de4cc 100644 --- a/libavcodec/ra288.c +++ b/libavcodec/ra288.c @@ -2,257 +2,222 @@ * 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" +#include "dsputil.h" -typedef struct { - float history[8]; - float output[40]; - float pr1[36]; - float pr2[10]; - int phase; - - float st1a[111], st1b[37], st1[37]; - float st2a[38], st2b[11], st2[11]; - float sb[41]; - float lhist[10]; -} Real288_internal; - -static inline float scalar_product_float(const float * v1, const float * v2, - int size) -{ - float res = 0.; - - while (size--) - res += *v1++ * *v2++; - - return res; -} - -/* Decode and produce output */ -static void decode(Real288_internal *glob, float gain, int cb_coef) -{ - int x, y; - double sumsum; - float sum, buffer[5]; - - memmove(glob->sb + 5, glob->sb, 36 * sizeof(*glob->sb)); - - for (x=4; x >= 0; x--) - glob->sb[x] = -scalar_product_float(glob->sb + x + 1, glob->pr1, 36); - - /* convert log and do rms */ - sum = 32. - scalar_product_float(glob->pr2, glob->lhist, 10); +#define MAX_BACKWARD_FILTER_ORDER 36 +#define MAX_BACKWARD_FILTER_LEN 40 +#define MAX_BACKWARD_FILTER_NONREC 35 - sum = av_clipf(sum, 0, 60); - - sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*f */ - - for (x=0; x < 5; x++) - buffer[x] = codetable[cb_coef][x] * sumsum; +#define RA288_BLOCK_SIZE 5 +#define RA288_BLOCKS_PER_FRAME 32 - sum = scalar_product_float(buffer, buffer, 5) / 5; +typedef struct { + DSPContext dsp; + DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A) + DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB) - sum = FFMAX(sum, 1); + /** speech data history (spec: SB). + * Its first 70 coefficients are updated only at backward filtering. + */ + float sp_hist[111]; - /* shift and store */ - memmove(glob->lhist, glob->lhist - 1, 10 * sizeof(*glob->lhist)); + /// speech part of the gain autocorrelation (spec: REXP) + float sp_rec[37]; - *glob->lhist = glob->history[glob->phase] = 10 * log10(sum) - 32; + /** log-gain history (spec: SBLG). + * Its first 28 coefficients are updated only at backward filtering. + */ + float gain_hist[38]; - for (x=1; x < 5; x++) - for (y=x-1; y >= 0; y--) - buffer[x] -= glob->pr1[x-y-1] * buffer[y]; + /// recursive part of the gain autocorrelation (spec: REXPLG) + float gain_rec[11]; +} RA288Context; - /* output */ - for (x=0; x < 5; x++) { - glob->output[glob->phase*5+x] = glob->sb[4-x] = - av_clipf(glob->sb[4-x] + buffer[x], -4095, 4095); - } +static av_cold int ra288_decode_init(AVCodecContext *avctx) +{ + RA288Context *ractx = avctx->priv_data; + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + dsputil_init(&ractx->dsp, avctx); + return 0; } -/* column multiply */ -static void colmult(float *tgt, const float *m1, const float *m2, int n) +static void convolve(float *tgt, const float *src, int len, int n) { - while (n--) - *(tgt++) = (*(m1++)) * (*(m2++)); + for (; n >= 0; n--) + tgt[n] = ff_dot_productf(src, src - n, len); + } -/** - * Converts autocorrelation coefficients to LPC coefficients using the - * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification. - * - * @return 1 if success, 0 if fail - */ -static int eval_lpc_coeffs(const float *in, float *tgt, int n) +static void decode(RA288Context *ractx, float gain, int cb_coef) { - int x, y; - double f0, f1, f2; + int i; + double sumsum; + float sum, buffer[5]; + float *block = ractx->sp_hist + 70 + 36; // current block + float *gain_block = ractx->gain_hist + 28; - if (in[n] == 0) - return 0; + memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); - if ((f0 = *in) <= 0) - return 0; + /* block 46 of G.728 spec */ + sum = 32.; + for (i=0; i < 10; i++) + sum -= gain_block[9-i] * ractx->gain_lpc[i]; - in--; // To avoid a -1 subtraction in the inner loop + /* block 47 of G.728 spec */ + sum = av_clipf(sum, 0, 60); - for (x=1; x <= n; x++) { - f1 = in[x+1]; + /* 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)); - for (y=0; y < x - 1; y++) - f1 += in[x-y]*tgt[y]; + for (i=0; i < 5; i++) + buffer[i] = codetable[cb_coef][i] * sumsum; - tgt[x-1] = f2 = -f1/f0; - for (y=0; y < x >> 1; y++) { - float temp = tgt[y] + tgt[x-y-2]*f2; - tgt[x-y-2] += tgt[y]*f2; - tgt[y] = temp; - } - if ((f0 += f1*f2) < 0) - return 0; - } + sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); - return 1; -} + sum = FFMAX(sum, 1); -/* product sum (lsf) */ -static void prodsum(float *tgt, const float *src, int len, int n) -{ - for (; n >= 0; n--) - tgt[n] = scalar_product_float(src, src - n, len); + /* shift and store */ + memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); + + gain_block[9] = 10 * log10(sum) - 32; + ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); } /** - * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification. + * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. * - * @param order the order of the filter - * @param n the length of the input - * @param non_rec the number of non recursive samples - * @param out the filter output - * @param in pointer to the input of the filter - * @param hist pointer to the input history of the filter. It is updated by - * this function. - * @param out pointer to the non recursive part of the output + * @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, const float *in, - float *out, float *hist, float *out2, - const float *window) +static void do_hybrid_window(RA288Context *ractx, + int order, int n, int non_rec, float *out, + float *hist, float *out2, const float *window) { - unsigned int x; - float buffer1[37]; - float buffer2[37]; - float work[111]; + int i; + float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; + float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; + LOCAL_ALIGNED_16(float, work)[FFALIGN(MAX_BACKWARD_FILTER_ORDER + + MAX_BACKWARD_FILTER_LEN + + MAX_BACKWARD_FILTER_NONREC, 8)]; - /* update history */ - memmove(hist , hist + n, (order + non_rec)*sizeof(*hist)); - memcpy (hist + order + non_rec, in , n *sizeof(*hist)); + ractx->dsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 8)); - colmult(work, window, hist, order + n + non_rec); + convolve(buffer1, work + order , n , order); + convolve(buffer2, work + order + n, non_rec, order); - prodsum(buffer1, work + order , n , order); - prodsum(buffer2, work + order + n, non_rec, order); - - for (x=0; x <= order; x++) { - out2[x] = out2[x] * 0.5625 + buffer1[x]; - out [x] = out2[x] + buffer2[x]; + for (i=0; i <= order; i++) { + out2[i] = out2[i] * 0.5625 + buffer1[i]; + out [i] = out2[i] + buffer2[i]; } - /* Multiply by the white noise correcting factor (WNCF) */ + /* 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(RA288Context *ractx, + float *hist, float *rec, const float *window, + float *lpc, const float *tab, + int order, int n, int non_rec, int move_size) { - float buffer1[40], temp1[37]; - float buffer2[8], temp2[11]; - - memcpy(buffer1 , glob->output + 20, 20*sizeof(*buffer1)); - memcpy(buffer1 + 20, glob->output , 20*sizeof(*buffer1)); - - do_hybrid_window(36, 40, 35, buffer1, temp1, glob->st1a, glob->st1b, - syn_window); - - if (eval_lpc_coeffs(temp1, glob->st1, 36)) - colmult(glob->pr1, glob->st1, table1a, 36); + float temp[MAX_BACKWARD_FILTER_ORDER+1]; - memcpy(buffer2 , glob->history + 4, 4*sizeof(*buffer2)); - memcpy(buffer2 + 4, glob->history , 4*sizeof(*buffer2)); + do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); - do_hybrid_window(10, 8, 20, buffer2, temp2, glob->st2a, glob->st2b, - gain_window); + if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) + ractx->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8)); - if (eval_lpc_coeffs(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, out_size; + RA288Context *ractx = avctx->priv_data; GetBitContext gb; if (buf_size < avctx->block_align) { av_log(avctx, AV_LOG_ERROR, "Error! Input buffer is too small [%d<%d]\n", buf_size, avctx->block_align); - return 0; + return AVERROR_INVALIDDATA; + } + + 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++) { + for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) { float gain = amptable[get_bits(&gb, 3)]; - int cb_coef = get_bits(&gb, 6 + (x&1)); - glob->phase = x & 7; - decode(glob, gain, cb_coef); + int cb_coef = get_bits(&gb, 6 + (i&1)); + + decode(ractx, gain, cb_coef); - for (y=0; y < 5; y++) - *(out++) = 8 * glob->output[glob->phase*5 + y]; + memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out)); + out += RA288_BLOCK_SIZE; - if (glob->phase == 3) - update(glob); + if ((i & 7) == 3) { + backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, + ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); + + backward_filter(ractx, 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)"), };