/* * QCELP decoder * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet * * 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 qcelpdec.c * QCELP decoder * @author Reynaldo H. Verdejo Pinochet * @remark FFmpeg merging spearheaded by Kenan Gillet */ #include #include "avcodec.h" #include "bitstream.h" #include "qcelp.h" #include "qcelpdata.h" #include "celp_math.h" #include "celp_filters.h" #undef NDEBUG #include static void weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length) { int i; for(i=0; ipriv_data; int i; avctx->sample_fmt = SAMPLE_FMT_FLT; for (i = 0; i < 10; i++) q->prev_lspf[i] = (i + 1) / 11.; return 0; } /** * Decodes the 10 quantized LSP frequencies from the LSPV/LSP * transmission codes of any bitrate and checks for badly received packets. * * @param q the context * @param lspf line spectral pair frequencies * * @return 0 on success, -1 if the packet is badly received * * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3 */ static int decode_lspf(QCELPContext *q, float *lspf) { int i; float tmp_lspf; if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) { float smooth; const float *predictors = (q->prev_bitrate != RATE_OCTAVE && q->prev_bitrate != I_F_Q ? q->prev_lspf : q->predictor_lspf); if(q->bitrate == RATE_OCTAVE) { q->octave_count++; for(i=0; i<10; i++) { q->predictor_lspf[i] = lspf[i] = (q->lspv[i] ? QCELP_LSP_SPREAD_FACTOR : -QCELP_LSP_SPREAD_FACTOR) + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11); } smooth = (q->octave_count < 10 ? .875 : 0.1); }else { float erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR; assert(q->bitrate == I_F_Q); if(q->erasure_count > 1) erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7); for(i=0; i<10; i++) { q->predictor_lspf[i] = lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11 + erasure_coeff * predictors[i]; } smooth = 0.125; } // Check the stability of the LSP frequencies. lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR); for(i=1; i<10; i++) lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR)); lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR)); for(i=9; i>0; i--) lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR)); // Low-pass filter the LSP frequencies. weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10); }else { q->octave_count = 0; tmp_lspf = 0.; for(i=0; i<5 ; i++) { lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->lspv[i]][0] * 0.0001; lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->lspv[i]][1] * 0.0001; } // Check for badly received packets. if(q->bitrate == RATE_QUARTER) { if(lspf[9] <= .70 || lspf[9] >= .97) return -1; for(i=3; i<10; i++) if(fabs(lspf[i] - lspf[i-2]) < .08) return -1; }else { if(lspf[9] <= .66 || lspf[9] >= .985) return -1; for(i=4; i<10; i++) if (fabs(lspf[i] - lspf[i-4]) < .0931) return -1; } } return 0; } /** * If the received packet is Rate 1/4 a further sanity check is made of the * codebook gain. * * @param cbgain the unpacked cbgain array * @return -1 if the sanity check fails, 0 otherwise * * TIA/EIA/IS-733 2.4.8.7.3 */ static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain) { int i, prev_diff=0; for(i=1; i<5; i++) { int diff = cbgain[i] - cbgain[i-1]; if(FFABS(diff) > 10) return -1; else if(FFABS(diff - prev_diff) > 12) return -1; prev_diff = diff; } return 0; } /** * Computes the scaled codebook vector Cdn From INDEX and GAIN * for all rates. * * The specification lacks some information here. * * TIA/EIA/IS-733 has an omission on the codebook index determination * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says * you have to subtract the decoded index parameter from the given scaled * codebook vector index 'n' to get the desired circular codebook index, but * it does not mention that you have to clamp 'n' to [0-9] in order to get * RI-compliant results. * * The reason for this mistake seems to be the fact they forgot to mention you * have to do these calculations per codebook subframe and adjust given * equation values accordingly. * * @param q the context * @param gain array holding the 4 pitch subframe gain values * @param cdn_vector array for the generated scaled codebook vector */ static void compute_svector(const QCELPContext *q, const float *gain, float *cdn_vector) { int i, j, k; uint16_t cbseed, cindex; float *rnd, tmp_gain, fir_filter_value; switch(q->bitrate) { case RATE_FULL: for(i=0; i<16; i++) { tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO; cindex = -q->cindex[i]; for(j=0; j<10; j++) *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127]; } break; case RATE_HALF: for(i=0; i<4; i++) { tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO; cindex = -q->cindex[i]; for (j = 0; j < 40; j++) *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127]; } break; case RATE_QUARTER: cbseed = (0x0003 & q->lspv[4])<<14 | (0x003F & q->lspv[3])<< 8 | (0x0060 & q->lspv[2])<< 1 | (0x0007 & q->lspv[1])<< 3 | (0x0038 & q->lspv[0])>> 3 ; rnd = q->rnd_fir_filter_mem + 20; for(i=0; i<8; i++) { tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0); for(k=0; k<20; k++) { cbseed = 521 * cbseed + 259; *rnd = (int16_t)cbseed; // FIR filter fir_filter_value = 0.0; for(j=0; j<10; j++) fir_filter_value += qcelp_rnd_fir_coefs[j ] * (rnd[-j ] + rnd[-20+j]); fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10]; *cdn_vector++ = tmp_gain * fir_filter_value; rnd++; } } memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float)); break; case RATE_OCTAVE: cbseed = q->first16bits; for(i=0; i<8; i++) { tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0); for(j=0; j<20; j++) { cbseed = 521 * cbseed + 259; *cdn_vector++ = tmp_gain * (int16_t)cbseed; } } break; case I_F_Q: cbseed = -44; // random codebook index for(i=0; i<4; i++) { tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO; for(j=0; j<40; j++) *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127]; } break; } } /** * Apply generic gain control. * * @param v_out output vector * @param v_in gain-controlled vector * @param v_ref vector to control gain of * * FIXME: If v_ref is a zero vector, it energy is zero * and the behavior of the gain control is * undefined in the specs. * * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6 */ static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in) { int i, j, len; float scalefactor; for(i=0, j=0; i<4; i++) { scalefactor = ff_dot_productf(v_in + j, v_in + j, 40); if(scalefactor) scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40) / scalefactor); else av_log_missing_feature(NULL, "Zero energy for gain control", 1); for(len=j+40; jbitrate >= RATE_QUARTER) weight = 0.25 * (subframe_num + 1); else if(q->bitrate == RATE_OCTAVE && !subframe_num) weight = 0.625; else weight = 1.0; if(weight != 1.0) { weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, weight, 1.0 - weight, 10); qcelp_lspf2lpc(interpolated_lspf, lpc); }else if(q->bitrate >= RATE_QUARTER || (q->bitrate == I_F_Q && !subframe_num)) qcelp_lspf2lpc(curr_lspf, lpc); } static int buf_size2bitrate(const int buf_size) { switch(buf_size) { case 35: return RATE_FULL; case 17: return RATE_HALF; case 8: return RATE_QUARTER; case 4: return RATE_OCTAVE; case 1: return SILENCE; } return -1; } static void warn_insufficient_frame_quality(AVCodecContext *avctx, const char *message) { av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number, message); } AVCodec qcelp_decoder = { .name = "qcelp", .type = CODEC_TYPE_AUDIO, .id = CODEC_ID_QCELP, .init = qcelp_decode_init, .decode = qcelp_decode_frame, .priv_data_size = sizeof(QCELPContext), .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"), };