/* * 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 * QCELP decoder * @author Reynaldo H. Verdejo Pinochet * @remark FFmpeg merging spearheaded by Kenan Gillet * @remark Development mentored by Benjamin Larson */ #include #include "libavutil/channel_layout.h" #include "avcodec.h" #include "internal.h" #include "get_bits.h" #include "dsputil.h" #include "qcelpdata.h" #include "celp_filters.h" #include "acelp_filters.h" #include "acelp_vectors.h" #include "lsp.h" #undef NDEBUG #include typedef enum { I_F_Q = -1, /**< insufficient frame quality */ SILENCE, RATE_OCTAVE, RATE_QUARTER, RATE_HALF, RATE_FULL } qcelp_packet_rate; typedef struct { AVFrame avframe; GetBitContext gb; qcelp_packet_rate bitrate; QCELPFrame frame; /**< unpacked data frame */ uint8_t erasure_count; uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */ float prev_lspf[10]; float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */ float pitch_synthesis_filter_mem[303]; float pitch_pre_filter_mem[303]; float rnd_fir_filter_mem[180]; float formant_mem[170]; float last_codebook_gain; int prev_g1[2]; int prev_bitrate; float pitch_gain[4]; uint8_t pitch_lag[4]; uint16_t first16bits; uint8_t warned_buf_mismatch_bitrate; /* postfilter */ float postfilter_synth_mem[10]; float postfilter_agc_mem; float postfilter_tilt_mem; } QCELPContext; /** * Initialize the speech codec according to the specification. * * TIA/EIA/IS-733 2.4.9 */ static av_cold int qcelp_decode_init(AVCodecContext *avctx) { QCELPContext *q = avctx->priv_data; int i; avctx->channels = 1; avctx->channel_layout = AV_CH_LAYOUT_MONO; avctx->sample_fmt = AV_SAMPLE_FMT_FLT; for (i = 0; i < 10; i++) q->prev_lspf[i] = (i + 1) / 11.; avcodec_get_frame_defaults(&q->avframe); avctx->coded_frame = &q->avframe; return 0; } /** * Decode the 10 quantized LSP frequencies from the LSPV/LSP * transmission codes of any bitrate and check 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, smooth, erasure_coeff; const float *predictors; if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) { 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->frame.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 { 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. ff_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->frame.lspv[i]][0] * 0.0001; lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.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; } /** * Convert codebook transmission codes to GAIN and INDEX. * * @param q the context * @param gain array holding the decoded gain * * TIA/EIA/IS-733 2.4.6.2 */ static void decode_gain_and_index(QCELPContext *q, float *gain) { int i, subframes_count, g1[16]; float slope; if (q->bitrate >= RATE_QUARTER) { switch (q->bitrate) { case RATE_FULL: subframes_count = 16; break; case RATE_HALF: subframes_count = 4; break; default: subframes_count = 5; } for (i = 0; i < subframes_count; i++) { g1[i] = 4 * q->frame.cbgain[i]; if (q->bitrate == RATE_FULL && !((i + 1) & 3)) { g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32); } gain[i] = qcelp_g12ga[g1[i]]; if (q->frame.cbsign[i]) { gain[i] = -gain[i]; q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127; } } q->prev_g1[0] = g1[i - 2]; q->prev_g1[1] = g1[i - 1]; q->last_codebook_gain = qcelp_g12ga[g1[i - 1]]; if (q->bitrate == RATE_QUARTER) { // Provide smoothing of the unvoiced excitation energy. gain[7] = gain[4]; gain[6] = 0.4 * gain[3] + 0.6 * gain[4]; gain[5] = gain[3]; gain[4] = 0.8 * gain[2] + 0.2 * gain[3]; gain[3] = 0.2 * gain[1] + 0.8 * gain[2]; gain[2] = gain[1]; gain[1] = 0.6 * gain[0] + 0.4 * gain[1]; } } else if (q->bitrate != SILENCE) { if (q->bitrate == RATE_OCTAVE) { g1[0] = 2 * q->frame.cbgain[0] + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54); subframes_count = 8; } else { assert(q->bitrate == I_F_Q); g1[0] = q->prev_g1[1]; switch (q->erasure_count) { case 1 : break; case 2 : g1[0] -= 1; break; case 3 : g1[0] -= 2; break; default: g1[0] -= 6; } if (g1[0] < 0) g1[0] = 0; subframes_count = 4; } // This interpolation is done to produce smoother background noise. slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count; for (i = 1; i <= subframes_count; i++) gain[i - 1] = q->last_codebook_gain + slope * i; q->last_codebook_gain = gain[i - 2]; q->prev_g1[0] = q->prev_g1[1]; q->prev_g1[1] = g1[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, diff, prev_diff = 0; for (i = 1; i < 5; i++) { 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; } /** * Compute 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(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->frame.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->frame.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->frame.lspv[4]) << 14 | (0x003F & q->frame.lspv[3]) << 8 | (0x0060 & q->frame.lspv[2]) << 1 | (0x0007 & q->frame.lspv[1]) << 3 | (0x0038 & q->frame.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; case SILENCE: memset(cdn_vector, 0, 160 * sizeof(float)); break; } } /** * Apply generic gain control. * * @param v_out output vector * @param v_in gain-controlled vector * @param v_ref vector to control gain of * * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6 */ static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in) { int i; for (i = 0; i < 160; i += 40) ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, ff_scalarproduct_float_c(v_ref + i, v_ref + i, 40), 40); } /** * Apply filter in pitch-subframe steps. * * @param memory buffer for the previous state of the filter * - must be able to contain 303 elements * - the 143 first elements are from the previous state * - the next 160 are for output * @param v_in input filter vector * @param gain per-subframe gain array, each element is between 0.0 and 2.0 * @param lag per-subframe lag array, each element is * - between 16 and 143 if its corresponding pfrac is 0, * - between 16 and 139 otherwise * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 * otherwise * * @return filter output vector */ static const float *do_pitchfilter(float memory[303], const float v_in[160], const float gain[4], const uint8_t *lag, const uint8_t pfrac[4]) { int i, j; float *v_lag, *v_out; const float *v_len; v_out = memory + 143; // Output vector starts at memory[143]. for (i = 0; i < 4; i++) { if (gain[i]) { v_lag = memory + 143 + 40 * i - lag[i]; for (v_len = v_in + 40; v_in < v_len; v_in++) { if (pfrac[i]) { // If it is a fractional lag... for (j = 0, *v_out = 0.; j < 4; j++) *v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]); } else *v_out = *v_lag; *v_out = *v_in + gain[i] * *v_out; v_lag++; v_out++; } } else { memcpy(v_out, v_in, 40 * sizeof(float)); v_in += 40; v_out += 40; } } memmove(memory, memory + 160, 143 * sizeof(float)); return memory + 143; } /** * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector. * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2 * * @param q the context * @param cdn_vector the scaled codebook vector */ static void apply_pitch_filters(QCELPContext *q, float *cdn_vector) { int i; const float *v_synthesis_filtered, *v_pre_filtered; if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE || (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) { if (q->bitrate >= RATE_HALF) { // Compute gain & lag for the whole frame. for (i = 0; i < 4; i++) { q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0; q->pitch_lag[i] = q->frame.plag[i] + 16; } } else { float max_pitch_gain; if (q->bitrate == I_F_Q) { if (q->erasure_count < 3) max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1); else max_pitch_gain = 0.0; } else { assert(q->bitrate == SILENCE); max_pitch_gain = 1.0; } for (i = 0; i < 4; i++) q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain); memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac)); } // pitch synthesis filter v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem, cdn_vector, q->pitch_gain, q->pitch_lag, q->frame.pfrac); // pitch prefilter update for (i = 0; i < 4; i++) q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0); v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem, v_synthesis_filtered, q->pitch_gain, q->pitch_lag, q->frame.pfrac); apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered); } else { memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17, 143 * sizeof(float)); memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float)); memset(q->pitch_gain, 0, sizeof(q->pitch_gain)); memset(q->pitch_lag, 0, sizeof(q->pitch_lag)); } } /** * Reconstruct LPC coefficients from the line spectral pair frequencies * and perform bandwidth expansion. * * @param lspf line spectral pair frequencies * @param lpc linear predictive coding coefficients * * @note: bandwidth_expansion_coeff could be precalculated into a table * but it seems to be slower on x86 * * TIA/EIA/IS-733 2.4.3.3.5 */ static void lspf2lpc(const float *lspf, float *lpc) { double lsp[10]; double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF; int i; for (i = 0; i < 10; i++) lsp[i] = cos(M_PI * lspf[i]); ff_acelp_lspd2lpc(lsp, lpc, 5); for (i = 0; i < 10; i++) { lpc[i] *= bandwidth_expansion_coeff; bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF; } } /** * Interpolate LSP frequencies and compute LPC coefficients * for a given bitrate & pitch subframe. * * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2 * * @param q the context * @param curr_lspf LSP frequencies vector of the current frame * @param lpc float vector for the resulting LPC * @param subframe_num frame number in decoded stream */ static void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc, const int subframe_num) { float interpolated_lspf[10]; float weight; if (q->bitrate >= 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) { ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, weight, 1.0 - weight, 10); lspf2lpc(interpolated_lspf, lpc); } else if (q->bitrate >= RATE_QUARTER || (q->bitrate == I_F_Q && !subframe_num)) lspf2lpc(curr_lspf, lpc); else if (q->bitrate == SILENCE && !subframe_num) lspf2lpc(q->prev_lspf, lpc); } static qcelp_packet_rate 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 I_F_Q; } /** * Determine the bitrate from the frame size and/or the first byte of the frame. * * @param avctx the AV codec context * @param buf_size length of the buffer * @param buf the bufffer * * @return the bitrate on success, * I_F_Q if the bitrate cannot be satisfactorily determined * * TIA/EIA/IS-733 2.4.8.7.1 */ static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size, const uint8_t **buf) { qcelp_packet_rate bitrate; if ((bitrate = buf_size2bitrate(buf_size)) >= 0) { if (bitrate > **buf) { QCELPContext *q = avctx->priv_data; if (!q->warned_buf_mismatch_bitrate) { av_log(avctx, AV_LOG_WARNING, "Claimed bitrate and buffer size mismatch.\n"); q->warned_buf_mismatch_bitrate = 1; } bitrate = **buf; } else if (bitrate < **buf) { av_log(avctx, AV_LOG_ERROR, "Buffer is too small for the claimed bitrate.\n"); return I_F_Q; } (*buf)++; } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) { av_log(avctx, AV_LOG_WARNING, "Bitrate byte is missing, guessing the bitrate from packet size.\n"); } else return I_F_Q; if (bitrate == SILENCE) { //FIXME: Remove experimental warning when tested with samples. av_log_ask_for_sample(avctx, "'Blank frame handling is experimental."); } return bitrate; } 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); } static void postfilter(QCELPContext *q, float *samples, float *lpc) { static const float pow_0_775[10] = { 0.775000, 0.600625, 0.465484, 0.360750, 0.279582, 0.216676, 0.167924, 0.130141, 0.100859, 0.078166 }, pow_0_625[10] = { 0.625000, 0.390625, 0.244141, 0.152588, 0.095367, 0.059605, 0.037253, 0.023283, 0.014552, 0.009095 }; float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160]; int n; for (n = 0; n < 10; n++) { lpc_s[n] = lpc[n] * pow_0_625[n]; lpc_p[n] = lpc[n] * pow_0_775[n]; } ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s, q->formant_mem + 10, 160, 10); memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10); ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10); memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10); ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160); ff_adaptive_gain_control(samples, pole_out + 10, ff_scalarproduct_float_c(q->formant_mem + 10, q->formant_mem + 10, 160), 160, 0.9375, &q->postfilter_agc_mem); } static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; QCELPContext *q = avctx->priv_data; float *outbuffer; int i, ret; float quantized_lspf[10], lpc[10]; float gain[16]; float *formant_mem; /* get output buffer */ q->avframe.nb_samples = 160; if ((ret = avctx->get_buffer(avctx, &q->avframe)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } outbuffer = (float *)q->avframe.data[0]; if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) { warn_insufficient_frame_quality(avctx, "bitrate cannot be determined."); goto erasure; } if (q->bitrate == RATE_OCTAVE && (q->first16bits = AV_RB16(buf)) == 0xFFFF) { warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on."); goto erasure; } if (q->bitrate > SILENCE) { const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate]; const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] + qcelp_unpacking_bitmaps_lengths[q->bitrate]; uint8_t *unpacked_data = (uint8_t *)&q->frame; init_get_bits(&q->gb, buf, 8 * buf_size); memset(&q->frame, 0, sizeof(QCELPFrame)); for (; bitmaps < bitmaps_end; bitmaps++) unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos; // Check for erasures/blanks on rates 1, 1/4 and 1/8. if (q->frame.reserved) { warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area."); goto erasure; } if (q->bitrate == RATE_QUARTER && codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) { warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed."); goto erasure; } if (q->bitrate >= RATE_HALF) { for (i = 0; i < 4; i++) { if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) { warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter."); goto erasure; } } } } decode_gain_and_index(q, gain); compute_svector(q, gain, outbuffer); if (decode_lspf(q, quantized_lspf) < 0) { warn_insufficient_frame_quality(avctx, "Badly received packets in frame."); goto erasure; } apply_pitch_filters(q, outbuffer); if (q->bitrate == I_F_Q) { erasure: q->bitrate = I_F_Q; q->erasure_count++; decode_gain_and_index(q, gain); compute_svector(q, gain, outbuffer); decode_lspf(q, quantized_lspf); apply_pitch_filters(q, outbuffer); } else q->erasure_count = 0; formant_mem = q->formant_mem + 10; for (i = 0; i < 4; i++) { interpolate_lpc(q, quantized_lspf, lpc, i); ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40, 10); formant_mem += 40; } // postfilter, as per TIA/EIA/IS-733 2.4.8.6 postfilter(q, outbuffer, lpc); memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float)); memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf)); q->prev_bitrate = q->bitrate; *got_frame_ptr = 1; *(AVFrame *)data = q->avframe; return buf_size; } AVCodec ff_qcelp_decoder = { .name = "qcelp", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_QCELP, .init = qcelp_decode_init, .decode = qcelp_decode_frame, .capabilities = CODEC_CAP_DR1, .priv_data_size = sizeof(QCELPContext), .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"), };