/* * AMR wideband decoder * Copyright (c) 2010 Marcelo Galvao Povoa * * 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 * AMR wideband decoder */ #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/float_dsp.h" #include "libavutil/lfg.h" #include "avcodec.h" #include "lsp.h" #include "celp_filters.h" #include "celp_math.h" #include "acelp_filters.h" #include "acelp_vectors.h" #include "acelp_pitch_delay.h" #include "internal.h" #define AMR_USE_16BIT_TABLES #include "amr.h" #include "amrwbdata.h" #include "mips/amrwbdec_mips.h" typedef struct AMRWBContext { AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream enum Mode fr_cur_mode; ///< mode index of current frame uint8_t fr_quality; ///< frame quality index (FQI) float isf_cur[LP_ORDER]; ///< working ISF vector from current frame float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame double isp[4][LP_ORDER]; ///< ISP vectors from current frame double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history float *excitation; ///< points to current excitation in excitation_buf[] float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset" uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters float demph_mem[1]; ///< previous value in the de-emphasis filter float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter AVLFG prng; ///< random number generator for white noise excitation uint8_t first_frame; ///< flag active during decoding of the first frame ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs CELPMContext celpm_ctx; ///< context for fixed point math operations } AMRWBContext; static av_cold int amrwb_decode_init(AVCodecContext *avctx) { AMRWBContext *ctx = avctx->priv_data; int i; if (avctx->channels > 1) { avpriv_report_missing_feature(avctx, "multi-channel AMR"); return AVERROR_PATCHWELCOME; } avctx->channels = 1; avctx->channel_layout = AV_CH_LAYOUT_MONO; if (!avctx->sample_rate) avctx->sample_rate = 16000; avctx->sample_fmt = AV_SAMPLE_FMT_FLT; av_lfg_init(&ctx->prng, 1); ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1]; ctx->first_frame = 1; for (i = 0; i < LP_ORDER; i++) ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15)); for (i = 0; i < 4; i++) ctx->prediction_error[i] = MIN_ENERGY; ff_acelp_filter_init(&ctx->acelpf_ctx); ff_acelp_vectors_init(&ctx->acelpv_ctx); ff_celp_filter_init(&ctx->celpf_ctx); ff_celp_math_init(&ctx->celpm_ctx); return 0; } /** * Decode the frame header in the "MIME/storage" format. This format * is simpler and does not carry the auxiliary frame information. * * @param[in] ctx The Context * @param[in] buf Pointer to the input buffer * * @return The decoded header length in bytes */ static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf) { /* Decode frame header (1st octet) */ ctx->fr_cur_mode = buf[0] >> 3 & 0x0F; ctx->fr_quality = (buf[0] & 0x4) == 0x4; return 1; } /** * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only). * * @param[in] ind Array of 5 indexes * @param[out] isf_q Buffer for isf_q[LP_ORDER] */ static void decode_isf_indices_36b(uint16_t *ind, float *isf_q) { int i; for (i = 0; i < 9; i++) isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); for (i = 0; i < 7; i++) isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); for (i = 0; i < 5; i++) isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15)); for (i = 0; i < 4; i++) isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15)); for (i = 0; i < 7; i++) isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15)); } /** * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode). * * @param[in] ind Array of 7 indexes * @param[out] isf_q Buffer for isf_q[LP_ORDER] */ static void decode_isf_indices_46b(uint16_t *ind, float *isf_q) { int i; for (i = 0; i < 9; i++) isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); for (i = 0; i < 7; i++) isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); for (i = 0; i < 3; i++) isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15)); for (i = 0; i < 3; i++) isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15)); for (i = 0; i < 3; i++) isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15)); for (i = 0; i < 3; i++) isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15)); for (i = 0; i < 4; i++) isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15)); } /** * Apply mean and past ISF values using the prediction factor. * Updates past ISF vector. * * @param[in,out] isf_q Current quantized ISF * @param[in,out] isf_past Past quantized ISF */ static void isf_add_mean_and_past(float *isf_q, float *isf_past) { int i; float tmp; for (i = 0; i < LP_ORDER; i++) { tmp = isf_q[i]; isf_q[i] += isf_mean[i] * (1.0f / (1 << 15)); isf_q[i] += PRED_FACTOR * isf_past[i]; isf_past[i] = tmp; } } /** * Interpolate the fourth ISP vector from current and past frames * to obtain an ISP vector for each subframe. * * @param[in,out] isp_q ISPs for each subframe * @param[in] isp4_past Past ISP for subframe 4 */ static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past) { int i, k; for (k = 0; k < 3; k++) { float c = isfp_inter[k]; for (i = 0; i < LP_ORDER; i++) isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i]; } } /** * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes). * Calculate integer lag and fractional lag always using 1/4 resolution. * In 1st and 3rd subframes the index is relative to last subframe integer lag. * * @param[out] lag_int Decoded integer pitch lag * @param[out] lag_frac Decoded fractional pitch lag * @param[in] pitch_index Adaptive codebook pitch index * @param[in,out] base_lag_int Base integer lag used in relative subframes * @param[in] subframe Current subframe index (0 to 3) */ static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index, uint8_t *base_lag_int, int subframe) { if (subframe == 0 || subframe == 2) { if (pitch_index < 376) { *lag_int = (pitch_index + 137) >> 2; *lag_frac = pitch_index - (*lag_int << 2) + 136; } else if (pitch_index < 440) { *lag_int = (pitch_index + 257 - 376) >> 1; *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2; /* the actual resolution is 1/2 but expressed as 1/4 */ } else { *lag_int = pitch_index - 280; *lag_frac = 0; } /* minimum lag for next subframe */ *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0), AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15); // XXX: the spec states clearly that *base_lag_int should be // the nearest integer to *lag_int (minus 8), but the ref code // actually always uses its floor, I'm following the latter } else { *lag_int = (pitch_index + 1) >> 2; *lag_frac = pitch_index - (*lag_int << 2); *lag_int += *base_lag_int; } } /** * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes. * The description is analogous to decode_pitch_lag_high, but in 6k60 the * relative index is used for all subframes except the first. */ static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index, uint8_t *base_lag_int, int subframe, enum Mode mode) { if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) { if (pitch_index < 116) { *lag_int = (pitch_index + 69) >> 1; *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2; } else { *lag_int = pitch_index - 24; *lag_frac = 0; } // XXX: same problem as before *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0), AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15); } else { *lag_int = (pitch_index + 1) >> 1; *lag_frac = (pitch_index - (*lag_int << 1)) * 2; *lag_int += *base_lag_int; } } /** * Find the pitch vector by interpolating the past excitation at the * pitch delay, which is obtained in this function. * * @param[in,out] ctx The context * @param[in] amr_subframe Current subframe data * @param[in] subframe Current subframe index (0 to 3) */ static void decode_pitch_vector(AMRWBContext *ctx, const AMRWBSubFrame *amr_subframe, const int subframe) { int pitch_lag_int, pitch_lag_frac; int i; float *exc = ctx->excitation; enum Mode mode = ctx->fr_cur_mode; if (mode <= MODE_8k85) { decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap, &ctx->base_pitch_lag, subframe, mode); } else decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap, &ctx->base_pitch_lag, subframe); ctx->pitch_lag_int = pitch_lag_int; pitch_lag_int += pitch_lag_frac > 0; /* Calculate the pitch vector by interpolating the past excitation at the pitch lag using a hamming windowed sinc function */ ctx->acelpf_ctx.acelp_interpolatef(exc, exc + 1 - pitch_lag_int, ac_inter, 4, pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4), LP_ORDER, AMRWB_SFR_SIZE + 1); /* Check which pitch signal path should be used * 6k60 and 8k85 modes have the ltp flag set to 0 */ if (amr_subframe->ltp) { memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float)); } else { for (i = 0; i < AMRWB_SFR_SIZE; i++) ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] + 0.18 * exc[i + 1]; memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float)); } } /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */ #define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len)) /** Get the bit at specified position */ #define BIT_POS(x, p) (((x) >> (p)) & 1) /** * The next six functions decode_[i]p_track decode exactly i pulses * positions and amplitudes (-1 or 1) in a subframe track using * an encoded pulse indexing (TS 26.190 section 5.8.2). * * The results are given in out[], in which a negative number means * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ). * * @param[out] out Output buffer (writes i elements) * @param[in] code Pulse index (no. of bits varies, see below) * @param[in] m (log2) Number of potential positions * @param[in] off Offset for decoded positions */ static inline void decode_1p_track(int *out, int code, int m, int off) { int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits out[0] = BIT_POS(code, m) ? -pos : pos; } static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits { int pos0 = BIT_STR(code, m, m) + off; int pos1 = BIT_STR(code, 0, m) + off; out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0; out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1; out[1] = pos0 > pos1 ? -out[1] : out[1]; } static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits { int half_2p = BIT_POS(code, 2*m - 1) << (m - 1); decode_2p_track(out, BIT_STR(code, 0, 2*m - 1), m - 1, off + half_2p); decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off); } static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits { int half_4p, subhalf_2p; int b_offset = 1 << (m - 1); switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */ case 0: /* 0 pulses in A, 4 pulses in B or vice versa */ half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2); decode_2p_track(out, BIT_STR(code, 0, 2*m - 3), m - 2, off + half_4p + subhalf_2p); decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1), m - 1, off + half_4p); break; case 1: /* 1 pulse in A, 3 pulses in B */ decode_1p_track(out, BIT_STR(code, 3*m - 2, m), m - 1, off); decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2), m - 1, off + b_offset); break; case 2: /* 2 pulses in each half */ decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1), m - 1, off); decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1), m - 1, off + b_offset); break; case 3: /* 3 pulses in A, 1 pulse in B */ decode_3p_track(out, BIT_STR(code, m, 3*m - 2), m - 1, off); decode_1p_track(out + 3, BIT_STR(code, 0, m), m - 1, off + b_offset); break; } } static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits { int half_3p = BIT_POS(code, 5*m - 1) << (m - 1); decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2), m - 1, off + half_3p); decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off); } static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits { int b_offset = 1 << (m - 1); /* which half has more pulses in cases 0 to 2 */ int half_more = BIT_POS(code, 6*m - 5) << (m - 1); int half_other = b_offset - half_more; switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */ case 0: /* 0 pulses in A, 6 pulses in B or vice versa */ decode_1p_track(out, BIT_STR(code, 0, m), m - 1, off + half_more); decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5), m - 1, off + half_more); break; case 1: /* 1 pulse in A, 5 pulses in B or vice versa */ decode_1p_track(out, BIT_STR(code, 0, m), m - 1, off + half_other); decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5), m - 1, off + half_more); break; case 2: /* 2 pulses in A, 4 pulses in B or vice versa */ decode_2p_track(out, BIT_STR(code, 0, 2*m - 1), m - 1, off + half_other); decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4), m - 1, off + half_more); break; case 3: /* 3 pulses in A, 3 pulses in B */ decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2), m - 1, off); decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2), m - 1, off + b_offset); break; } } /** * Decode the algebraic codebook index to pulse positions and signs, * then construct the algebraic codebook vector. * * @param[out] fixed_vector Buffer for the fixed codebook excitation * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only) * @param[in] pulse_lo LSBs part of the pulse index array * @param[in] mode Mode of the current frame */ static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi, const uint16_t *pulse_lo, const enum Mode mode) { /* sig_pos stores for each track the decoded pulse position indexes * (1-based) multiplied by its corresponding amplitude (+1 or -1) */ int sig_pos[4][6]; int spacing = (mode == MODE_6k60) ? 2 : 4; int i, j; switch (mode) { case MODE_6k60: for (i = 0; i < 2; i++) decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1); break; case MODE_8k85: for (i = 0; i < 4; i++) decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1); break; case MODE_12k65: for (i = 0; i < 4; i++) decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1); break; case MODE_14k25: for (i = 0; i < 2; i++) decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1); for (i = 2; i < 4; i++) decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1); break; case MODE_15k85: for (i = 0; i < 4; i++) decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1); break; case MODE_18k25: for (i = 0; i < 4; i++) decode_4p_track(sig_pos[i], (int) pulse_lo[i] + ((int) pulse_hi[i] << 14), 4, 1); break; case MODE_19k85: for (i = 0; i < 2; i++) decode_5p_track(sig_pos[i], (int) pulse_lo[i] + ((int) pulse_hi[i] << 10), 4, 1); for (i = 2; i < 4; i++) decode_4p_track(sig_pos[i], (int) pulse_lo[i] + ((int) pulse_hi[i] << 14), 4, 1); break; case MODE_23k05: case MODE_23k85: for (i = 0; i < 4; i++) decode_6p_track(sig_pos[i], (int) pulse_lo[i] + ((int) pulse_hi[i] << 11), 4, 1); break; } memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE); for (i = 0; i < 4; i++) for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) { int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i; fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0; } } /** * Decode pitch gain and fixed gain correction factor. * * @param[in] vq_gain Vector-quantized index for gains * @param[in] mode Mode of the current frame * @param[out] fixed_gain_factor Decoded fixed gain correction factor * @param[out] pitch_gain Decoded pitch gain */ static void decode_gains(const uint8_t vq_gain, const enum Mode mode, float *fixed_gain_factor, float *pitch_gain) { const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] : qua_gain_7b[vq_gain]); *pitch_gain = gains[0] * (1.0f / (1 << 14)); *fixed_gain_factor = gains[1] * (1.0f / (1 << 11)); } /** * Apply pitch sharpening filters to the fixed codebook vector. * * @param[in] ctx The context * @param[in,out] fixed_vector Fixed codebook excitation */ // XXX: Spec states this procedure should be applied when the pitch // lag is less than 64, but this checking seems absent in reference and AMR-NB static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector) { int i; /* Tilt part */ for (i = AMRWB_SFR_SIZE - 1; i != 0; i--) fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef; /* Periodicity enhancement part */ for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++) fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85; } /** * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced). * * @param[in] p_vector, f_vector Pitch and fixed excitation vectors * @param[in] p_gain, f_gain Pitch and fixed gains * @param[in] ctx The context */ // XXX: There is something wrong with the precision here! The magnitudes // of the energies are not correct. Please check the reference code carefully static float voice_factor(float *p_vector, float p_gain, float *f_vector, float f_gain, CELPMContext *ctx) { double p_ener = (double) ctx->dot_productf(p_vector, p_vector, AMRWB_SFR_SIZE) * p_gain * p_gain; double f_ener = (double) ctx->dot_productf(f_vector, f_vector, AMRWB_SFR_SIZE) * f_gain * f_gain; return (p_ener - f_ener) / (p_ener + f_ener + 0.01); } /** * Reduce fixed vector sparseness by smoothing with one of three IR filters, * also known as "adaptive phase dispersion". * * @param[in] ctx The context * @param[in,out] fixed_vector Unfiltered fixed vector * @param[out] buf Space for modified vector if necessary * * @return The potentially overwritten filtered fixed vector address */ static float *anti_sparseness(AMRWBContext *ctx, float *fixed_vector, float *buf) { int ir_filter_nr; if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes return fixed_vector; if (ctx->pitch_gain[0] < 0.6) { ir_filter_nr = 0; // strong filtering } else if (ctx->pitch_gain[0] < 0.9) { ir_filter_nr = 1; // medium filtering } else ir_filter_nr = 2; // no filtering /* detect 'onset' */ if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) { if (ir_filter_nr < 2) ir_filter_nr++; } else { int i, count = 0; for (i = 0; i < 6; i++) if (ctx->pitch_gain[i] < 0.6) count++; if (count > 2) ir_filter_nr = 0; if (ir_filter_nr > ctx->prev_ir_filter_nr + 1) ir_filter_nr--; } /* update ir filter strength history */ ctx->prev_ir_filter_nr = ir_filter_nr; ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85); if (ir_filter_nr < 2) { int i; const float *coef = ir_filters_lookup[ir_filter_nr]; /* Circular convolution code in the reference * decoder was modified to avoid using one * extra array. The filtered vector is given by: * * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) } */ memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE); for (i = 0; i < AMRWB_SFR_SIZE; i++) if (fixed_vector[i]) ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i], AMRWB_SFR_SIZE); fixed_vector = buf; } return fixed_vector; } /** * Calculate a stability factor {teta} based on distance between * current and past isf. A value of 1 shows maximum signal stability. */ static float stability_factor(const float *isf, const float *isf_past) { int i; float acc = 0.0; for (i = 0; i < LP_ORDER - 1; i++) acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]); // XXX: This part is not so clear from the reference code // the result is more accurate changing the "/ 256" to "* 512" return FFMAX(0.0, 1.25 - acc * 0.8 * 512); } /** * Apply a non-linear fixed gain smoothing in order to reduce * fluctuation in the energy of excitation. * * @param[in] fixed_gain Unsmoothed fixed gain * @param[in,out] prev_tr_gain Previous threshold gain (updated) * @param[in] voice_fac Frame voicing factor * @param[in] stab_fac Frame stability factor * * @return The smoothed gain */ static float noise_enhancer(float fixed_gain, float *prev_tr_gain, float voice_fac, float stab_fac) { float sm_fac = 0.5 * (1 - voice_fac) * stab_fac; float g0; // XXX: the following fixed-point constants used to in(de)crement // gain by 1.5dB were taken from the reference code, maybe it could // be simpler if (fixed_gain < *prev_tr_gain) { g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain * (6226 * (1.0f / (1 << 15)))); // +1.5 dB } else g0 = FFMAX(*prev_tr_gain, fixed_gain * (27536 * (1.0f / (1 << 15)))); // -1.5 dB *prev_tr_gain = g0; // update next frame threshold return sm_fac * g0 + (1 - sm_fac) * fixed_gain; } /** * Filter the fixed_vector to emphasize the higher frequencies. * * @param[in,out] fixed_vector Fixed codebook vector * @param[in] voice_fac Frame voicing factor */ static void pitch_enhancer(float *fixed_vector, float voice_fac) { int i; float cpe = 0.125 * (1 + voice_fac); float last = fixed_vector[0]; // holds c(i - 1) fixed_vector[0] -= cpe * fixed_vector[1]; for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) { float cur = fixed_vector[i]; fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]); last = cur; } fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last; } /** * Conduct 16th order linear predictive coding synthesis from excitation. * * @param[in] ctx Pointer to the AMRWBContext * @param[in] lpc Pointer to the LPC coefficients * @param[out] excitation Buffer for synthesis final excitation * @param[in] fixed_gain Fixed codebook gain for synthesis * @param[in] fixed_vector Algebraic codebook vector * @param[in,out] samples Pointer to the output samples and memory */ static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation, float fixed_gain, const float *fixed_vector, float *samples) { ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector, ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE); /* emphasize pitch vector contribution in low bitrate modes */ if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) { int i; float energy = ctx->celpm_ctx.dot_productf(excitation, excitation, AMRWB_SFR_SIZE); // XXX: Weird part in both ref code and spec. A unknown parameter // {beta} seems to be identical to the current pitch gain float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0]; for (i = 0; i < AMRWB_SFR_SIZE; i++) excitation[i] += pitch_factor * ctx->pitch_vector[i]; ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy, AMRWB_SFR_SIZE); } ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation, AMRWB_SFR_SIZE, LP_ORDER); } /** * Apply to synthesis a de-emphasis filter of the form: * H(z) = 1 / (1 - m * z^-1) * * @param[out] out Output buffer * @param[in] in Input samples array with in[-1] * @param[in] m Filter coefficient * @param[in,out] mem State from last filtering */ static void de_emphasis(float *out, float *in, float m, float mem[1]) { int i; out[0] = in[0] + m * mem[0]; for (i = 1; i < AMRWB_SFR_SIZE; i++) out[i] = in[i] + out[i - 1] * m; mem[0] = out[AMRWB_SFR_SIZE - 1]; } /** * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using * a FIR interpolation filter. Uses past data from before *in address. * * @param[out] out Buffer for interpolated signal * @param[in] in Current signal data (length 0.8*o_size) * @param[in] o_size Output signal length * @param[in] ctx The context */ static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx) { const float *in0 = in - UPS_FIR_SIZE + 1; int i, j, k; int int_part = 0, frac_part; i = 0; for (j = 0; j < o_size / 5; j++) { out[i] = in[int_part]; frac_part = 4; i++; for (k = 1; k < 5; k++) { out[i] = ctx->dot_productf(in0 + int_part, upsample_fir[4 - frac_part], UPS_MEM_SIZE); int_part++; frac_part--; i++; } } } /** * Calculate the high-band gain based on encoded index (23k85 mode) or * on the low-band speech signal and the Voice Activity Detection flag. * * @param[in] ctx The context * @param[in] synth LB speech synthesis at 12.8k * @param[in] hb_idx Gain index for mode 23k85 only * @param[in] vad VAD flag for the frame */ static float find_hb_gain(AMRWBContext *ctx, const float *synth, uint16_t hb_idx, uint8_t vad) { int wsp = (vad > 0); float tilt; float tmp; if (ctx->fr_cur_mode == MODE_23k85) return qua_hb_gain[hb_idx] * (1.0f / (1 << 14)); tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1); if (tmp > 0) { tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE); } else tilt = 0; /* return gain bounded by [0.1, 1.0] */ return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0); } /** * Generate the high-band excitation with the same energy from the lower * one and scaled by the given gain. * * @param[in] ctx The context * @param[out] hb_exc Buffer for the excitation * @param[in] synth_exc Low-band excitation used for synthesis * @param[in] hb_gain Wanted excitation gain */ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc, const float *synth_exc, float hb_gain) { int i; float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE); /* Generate a white-noise excitation */ for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng); ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc, energy * hb_gain * hb_gain, AMRWB_SFR_SIZE_16k); } /** * Calculate the auto-correlation for the ISF difference vector. */ static float auto_correlation(float *diff_isf, float mean, int lag) { int i; float sum = 0.0; for (i = 7; i < LP_ORDER - 2; i++) { float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean); sum += prod * prod; } return sum; } /** * Extrapolate a ISF vector to the 16kHz range (20th order LP) * used at mode 6k60 LP filter for the high frequency band. * * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER * values on input */ static void extrapolate_isf(float isf[LP_ORDER_16k]) { float diff_isf[LP_ORDER - 2], diff_mean; float corr_lag[3]; float est, scale; int i, j, i_max_corr; isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1]; /* Calculate the difference vector */ for (i = 0; i < LP_ORDER - 2; i++) diff_isf[i] = isf[i + 1] - isf[i]; diff_mean = 0.0; for (i = 2; i < LP_ORDER - 2; i++) diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4)); /* Find which is the maximum autocorrelation */ i_max_corr = 0; for (i = 0; i < 3; i++) { corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2); if (corr_lag[i] > corr_lag[i_max_corr]) i_max_corr = i; } i_max_corr++; for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++) isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr] - isf[i - 2 - i_max_corr]; /* Calculate an estimate for ISF(18) and scale ISF based on the error */ est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0; scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) / (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]); for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++) diff_isf[j] = scale * (isf[i] - isf[i - 1]); /* Stability insurance */ for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++) if (diff_isf[i] + diff_isf[i - 1] < 5.0) { if (diff_isf[i] > diff_isf[i - 1]) { diff_isf[i - 1] = 5.0 - diff_isf[i]; } else diff_isf[i] = 5.0 - diff_isf[i - 1]; } for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++) isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15)); /* Scale the ISF vector for 16000 Hz */ for (i = 0; i < LP_ORDER_16k - 1; i++) isf[i] *= 0.8; } /** * Spectral expand the LP coefficients using the equation: * y[i] = x[i] * (gamma ** i) * * @param[out] out Output buffer (may use input array) * @param[in] lpc LP coefficients array * @param[in] gamma Weighting factor * @param[in] size LP array size */ static void lpc_weighting(float *out, const float *lpc, float gamma, int size) { int i; float fac = gamma; for (i = 0; i < size; i++) { out[i] = lpc[i] * fac; fac *= gamma; } } /** * Conduct 20th order linear predictive coding synthesis for the high * frequency band excitation at 16kHz. * * @param[in] ctx The context * @param[in] subframe Current subframe index (0 to 3) * @param[in,out] samples Pointer to the output speech samples * @param[in] exc Generated white-noise scaled excitation * @param[in] isf Current frame isf vector * @param[in] isf_past Past frame final isf vector */ static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples, const float *exc, const float *isf, const float *isf_past) { float hb_lpc[LP_ORDER_16k]; enum Mode mode = ctx->fr_cur_mode; if (mode == MODE_6k60) { float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation double e_isp[LP_ORDER_16k]; ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe], 1.0 - isfp_inter[subframe], LP_ORDER); extrapolate_isf(e_isf); e_isf[LP_ORDER_16k - 1] *= 2.0; ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k); ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k); lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k); } else { lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER); } ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k, (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER); } /** * Apply a 15th order filter to high-band samples. * The filter characteristic depends on the given coefficients. * * @param[out] out Buffer for filtered output * @param[in] fir_coef Filter coefficients * @param[in,out] mem State from last filtering (updated) * @param[in] in Input speech data (high-band) * * @remark It is safe to pass the same array in in and out parameters */ #ifndef hb_fir_filter static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1], float mem[HB_FIR_SIZE], const float *in) { int i, j; float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples memcpy(data, mem, HB_FIR_SIZE * sizeof(float)); memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float)); for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) { out[i] = 0.0; for (j = 0; j <= HB_FIR_SIZE; j++) out[i] += data[i + j] * fir_coef[j]; } memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float)); } #endif /* hb_fir_filter */ /** * Update context state before the next subframe. */ static void update_sub_state(AMRWBContext *ctx) { memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE], (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float)); memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float)); memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float)); memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE], LP_ORDER * sizeof(float)); memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE], UPS_MEM_SIZE * sizeof(float)); memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k], LP_ORDER_16k * sizeof(float)); } static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { AMRWBContext *ctx = avctx->priv_data; AVFrame *frame = data; AMRWBFrame *cf = &ctx->frame; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; int expected_fr_size, header_size; float *buf_out; float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing float fixed_gain_factor; // fixed gain correction factor (gamma) float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use float synth_fixed_gain; // the fixed gain that synthesis should use float voice_fac, stab_fac; // parameters used for gain smoothing float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis float hb_gain; int sub, i, ret; /* get output buffer */ frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; buf_out = (float *)frame->data[0]; header_size = decode_mime_header(ctx, buf); if (ctx->fr_cur_mode > MODE_SID) { av_log(avctx, AV_LOG_ERROR, "Invalid mode %d\n", ctx->fr_cur_mode); return AVERROR_INVALIDDATA; } expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1; if (buf_size < expected_fr_size) { av_log(avctx, AV_LOG_ERROR, "Frame too small (%d bytes). Truncated file?\n", buf_size); *got_frame_ptr = 0; return AVERROR_INVALIDDATA; } if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID) av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n"); if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */ avpriv_request_sample(avctx, "SID mode"); return AVERROR_PATCHWELCOME; } ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame), buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]); /* Decode the quantized ISF vector */ if (ctx->fr_cur_mode == MODE_6k60) { decode_isf_indices_36b(cf->isp_id, ctx->isf_cur); } else { decode_isf_indices_46b(cf->isp_id, ctx->isf_cur); } isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past); ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1); stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final); ctx->isf_cur[LP_ORDER - 1] *= 2.0; ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER); /* Generate a ISP vector for each subframe */ if (ctx->first_frame) { ctx->first_frame = 0; memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double)); } interpolate_isp(ctx->isp, ctx->isp_sub4_past); for (sub = 0; sub < 4; sub++) ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER); for (sub = 0; sub < 4; sub++) { const AMRWBSubFrame *cur_subframe = &cf->subframe[sub]; float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k; /* Decode adaptive codebook (pitch vector) */ decode_pitch_vector(ctx, cur_subframe, sub); /* Decode innovative codebook (fixed vector) */ decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih, cur_subframe->pul_il, ctx->fr_cur_mode); pitch_sharpening(ctx, ctx->fixed_vector); decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode, &fixed_gain_factor, &ctx->pitch_gain[0]); ctx->fixed_gain[0] = ff_amr_set_fixed_gain(fixed_gain_factor, ctx->celpm_ctx.dot_productf(ctx->fixed_vector, ctx->fixed_vector, AMRWB_SFR_SIZE) / AMRWB_SFR_SIZE, ctx->prediction_error, ENERGY_MEAN, energy_pred_fac); /* Calculate voice factor and store tilt for next subframe */ voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0], ctx->fixed_vector, ctx->fixed_gain[0], &ctx->celpm_ctx); ctx->tilt_coef = voice_fac * 0.25 + 0.25; /* Construct current excitation */ for (i = 0; i < AMRWB_SFR_SIZE; i++) { ctx->excitation[i] *= ctx->pitch_gain[0]; ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i]; ctx->excitation[i] = truncf(ctx->excitation[i]); } /* Post-processing of excitation elements */ synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain, voice_fac, stab_fac); synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector, spare_vector); pitch_enhancer(synth_fixed_vector, voice_fac); synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain, synth_fixed_vector, &ctx->samples_az[LP_ORDER]); /* Synthesis speech post-processing */ de_emphasis(&ctx->samples_up[UPS_MEM_SIZE], &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem); ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE], &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles, hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE); upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE], AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx); /* High frequency band (6.4 - 7.0 kHz) generation part */ ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples, &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles, hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE); hb_gain = find_hb_gain(ctx, hb_samples, cur_subframe->hb_gain, cf->vad); scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain); hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k], hb_exc, ctx->isf_cur, ctx->isf_past_final); /* High-band post-processing filters */ hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem, &ctx->samples_hb[LP_ORDER_16k]); if (ctx->fr_cur_mode == MODE_23k85) hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem, hb_samples); /* Add the low and high frequency bands */ for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15)); /* Update buffers and history */ update_sub_state(ctx); } /* update state for next frame */ memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0])); memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float)); *got_frame_ptr = 1; return expected_fr_size; } AVCodec ff_amrwb_decoder = { .name = "amrwb", .long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_AMR_WB, .priv_data_size = sizeof(AMRWBContext), .init = amrwb_decode_init, .decode = amrwb_decode_frame, .capabilities = AV_CODEC_CAP_DR1, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_NONE }, };