/* * SIPR / ACELP.NET decoder * * Copyright (c) 2008 Vladimir Voroshilov * Copyright (c) 2009 Vitor Sessak * * 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 */ #include <math.h> #include <stdint.h> #include "libavutil/mathematics.h" #include "avcodec.h" #define ALT_BITSTREAM_READER_LE #include "get_bits.h" #include "dsputil.h" #include "lsp.h" #include "celp_math.h" #include "acelp_vectors.h" #include "acelp_pitch_delay.h" #include "acelp_filters.h" #include "celp_filters.h" #define MAX_SUBFRAME_COUNT 5 #include "sipr.h" #include "siprdata.h" typedef struct { const char *mode_name; uint16_t bits_per_frame; uint8_t subframe_count; uint8_t frames_per_packet; float pitch_sharp_factor; /* bitstream parameters */ uint8_t number_of_fc_indexes; uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor /** size in bits of the i-th stage vector of quantizer */ uint8_t vq_indexes_bits[5]; /** size in bits of the adaptive-codebook index for every subframe */ uint8_t pitch_delay_bits[5]; uint8_t gp_index_bits; uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes } SiprModeParam; static const SiprModeParam modes[MODE_COUNT] = { [MODE_16k] = { .mode_name = "16k", .bits_per_frame = 160, .subframe_count = SUBFRAME_COUNT_16k, .frames_per_packet = 1, .pitch_sharp_factor = 0.00, .number_of_fc_indexes = 10, .ma_predictor_bits = 1, .vq_indexes_bits = {7, 8, 7, 7, 7}, .pitch_delay_bits = {9, 6}, .gp_index_bits = 4, .fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5}, .gc_index_bits = 5 }, [MODE_8k5] = { .mode_name = "8k5", .bits_per_frame = 152, .subframe_count = 3, .frames_per_packet = 1, .pitch_sharp_factor = 0.8, .number_of_fc_indexes = 3, .ma_predictor_bits = 0, .vq_indexes_bits = {6, 7, 7, 7, 5}, .pitch_delay_bits = {8, 5, 5}, .gp_index_bits = 0, .fc_index_bits = {9, 9, 9}, .gc_index_bits = 7 }, [MODE_6k5] = { .mode_name = "6k5", .bits_per_frame = 232, .subframe_count = 3, .frames_per_packet = 2, .pitch_sharp_factor = 0.8, .number_of_fc_indexes = 3, .ma_predictor_bits = 0, .vq_indexes_bits = {6, 7, 7, 7, 5}, .pitch_delay_bits = {8, 5, 5}, .gp_index_bits = 0, .fc_index_bits = {5, 5, 5}, .gc_index_bits = 7 }, [MODE_5k0] = { .mode_name = "5k0", .bits_per_frame = 296, .subframe_count = 5, .frames_per_packet = 2, .pitch_sharp_factor = 0.85, .number_of_fc_indexes = 1, .ma_predictor_bits = 0, .vq_indexes_bits = {6, 7, 7, 7, 5}, .pitch_delay_bits = {8, 5, 8, 5, 5}, .gp_index_bits = 0, .fc_index_bits = {10}, .gc_index_bits = 7 } }; const float ff_pow_0_5[] = { 1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4), 1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8), 1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12), 1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16) }; static void dequant(float *out, const int *idx, const float *cbs[]) { int i; int stride = 2; int num_vec = 5; for (i = 0; i < num_vec; i++) memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float)); } static void lsf_decode_fp(float *lsfnew, float *lsf_history, const SiprParameters *parm) { int i; float lsf_tmp[LP_FILTER_ORDER]; dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks); for (i = 0; i < LP_FILTER_ORDER; i++) lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i]; ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1); /* Note that a minimum distance is not enforced between the last value and the previous one, contrary to what is done in ff_acelp_reorder_lsf() */ ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1); lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI); memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history)); for (i = 0; i < LP_FILTER_ORDER - 1; i++) lsfnew[i] = cos(lsfnew[i]); lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI; } /** Apply pitch lag to the fixed vector (AMR section 6.1.2). */ static void pitch_sharpening(int pitch_lag_int, float beta, float *fixed_vector) { int i; for (i = pitch_lag_int; i < SUBFR_SIZE; i++) fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int]; } /** * Extracts decoding parameters from the input bitstream. * @param parms parameters structure * @param pgb pointer to initialized GetBitContext structure */ static void decode_parameters(SiprParameters* parms, GetBitContext *pgb, const SiprModeParam *p) { int i, j; parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits); for (i = 0; i < 5; i++) parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]); for (i = 0; i < p->subframe_count; i++) { parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]); parms->gp_index[i] = get_bits(pgb, p->gp_index_bits); for (j = 0; j < p->number_of_fc_indexes; j++) parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]); parms->gc_index[i] = get_bits(pgb, p->gc_index_bits); } } static void lsp2lpc_sipr(const double *lsp, float *Az) { int lp_half_order = LP_FILTER_ORDER >> 1; double buf[(LP_FILTER_ORDER >> 1) + 1]; double pa[(LP_FILTER_ORDER >> 1) + 1]; double *qa = buf + 1; int i,j; qa[-1] = 0.0; ff_lsp2polyf(lsp , pa, lp_half_order ); ff_lsp2polyf(lsp + 1, qa, lp_half_order - 1); for (i = 1, j = LP_FILTER_ORDER - 1; i < lp_half_order; i++, j--) { double paf = pa[i] * (1 + lsp[LP_FILTER_ORDER - 1]); double qaf = (qa[i] - qa[i-2]) * (1 - lsp[LP_FILTER_ORDER - 1]); Az[i-1] = (paf + qaf) * 0.5; Az[j-1] = (paf - qaf) * 0.5; } Az[lp_half_order - 1] = (1.0 + lsp[LP_FILTER_ORDER - 1]) * pa[lp_half_order] * 0.5; Az[LP_FILTER_ORDER - 1] = lsp[LP_FILTER_ORDER - 1]; } static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, int num_subfr) { double lsfint[LP_FILTER_ORDER]; int i,j; float t, t0 = 1.0 / num_subfr; t = t0 * 0.5; for (i = 0; i < num_subfr; i++) { for (j = 0; j < LP_FILTER_ORDER; j++) lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j]; lsp2lpc_sipr(lsfint, Az); Az += LP_FILTER_ORDER; t += t0; } } /** * Evaluates the adaptative impulse response. */ static void eval_ir(const float *Az, int pitch_lag, float *freq, float pitch_sharp_factor) { float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1]; int i; tmp1[0] = 1.; for (i = 0; i < LP_FILTER_ORDER; i++) { tmp1[i+1] = Az[i] * ff_pow_0_55[i]; tmp2[i ] = Az[i] * ff_pow_0_7 [i]; } memset(tmp1 + 11, 0, 37 * sizeof(float)); ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE, LP_FILTER_ORDER); pitch_sharpening(pitch_lag, pitch_sharp_factor, freq); } /** * Evaluates the convolution of a vector with a sparse vector. */ static void convolute_with_sparse(float *out, const AMRFixed *pulses, const float *shape, int length) { int i, j; memset(out, 0, length*sizeof(float)); for (i = 0; i < pulses->n; i++) for (j = pulses->x[i]; j < length; j++) out[j] += pulses->y[i] * shape[j - pulses->x[i]]; } /** * Apply postfilter, very similar to AMR one. */ static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples) { float buf[SUBFR_SIZE + LP_FILTER_ORDER]; float *pole_out = buf + LP_FILTER_ORDER; float lpc_n[LP_FILTER_ORDER]; float lpc_d[LP_FILTER_ORDER]; int i; for (i = 0; i < LP_FILTER_ORDER; i++) { lpc_d[i] = lpc[i] * ff_pow_0_75[i]; lpc_n[i] = lpc[i] * ff_pow_0_5 [i]; }; memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem, LP_FILTER_ORDER*sizeof(float)); ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE, LP_FILTER_ORDER); memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER, LP_FILTER_ORDER*sizeof(float)); ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE); memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0, LP_FILTER_ORDER*sizeof(*pole_out)); memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER, LP_FILTER_ORDER*sizeof(*pole_out)); ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE, LP_FILTER_ORDER); } static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses, SiprMode mode, int low_gain) { int i; switch (mode) { case MODE_6k5: for (i = 0; i < 3; i++) { fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i; fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1; } fixed_sparse->n = 3; break; case MODE_8k5: for (i = 0; i < 3; i++) { fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i; fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i; fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0; fixed_sparse->y[2*i + 1] = (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ? -fixed_sparse->y[2*i ] : fixed_sparse->y[2*i]; } fixed_sparse->n = 6; break; case MODE_5k0: default: if (low_gain) { int offset = (pulses[0] & 0x200) ? 2 : 0; int val = pulses[0]; for (i = 0; i < 3; i++) { int index = (val & 0x7) * 6 + 4 - i*2; fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1; fixed_sparse->x[i] = index; val >>= 3; } fixed_sparse->n = 3; } else { int pulse_subset = (pulses[0] >> 8) & 1; fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset; fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1; fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1; fixed_sparse->y[1] = -fixed_sparse->y[0]; fixed_sparse->n = 2; } break; } } static void decode_frame(SiprContext *ctx, SiprParameters *params, float *out_data) { int i, j; int subframe_count = modes[ctx->mode].subframe_count; int frame_size = subframe_count * SUBFR_SIZE; float Az[LP_FILTER_ORDER * MAX_SUBFRAME_COUNT]; float *excitation; float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER]; float lsf_new[LP_FILTER_ORDER]; float *impulse_response = ir_buf + LP_FILTER_ORDER; float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for // memory alignment int t0_first = 0; AMRFixed fixed_cb; memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float)); lsf_decode_fp(lsf_new, ctx->lsf_history, params); sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count); memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float)); excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL; for (i = 0; i < subframe_count; i++) { float *pAz = Az + i*LP_FILTER_ORDER; float fixed_vector[SUBFR_SIZE]; int T0,T0_frac; float pitch_gain, gain_code, avg_energy; ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i, ctx->mode == MODE_5k0, 6); if (i == 0 || (i == 2 && ctx->mode == MODE_5k0)) t0_first = T0; ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0), ff_b60_sinc, 6, 2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER, SUBFR_SIZE); decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode, ctx->past_pitch_gain < 0.8); eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor); convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response, SUBFR_SIZE); avg_energy = (0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/ SUBFR_SIZE; ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0]; gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1], avg_energy, ctx->energy_history, 34 - 15.0/(0.05*M_LN10/M_LN2), pred); ff_weighted_vector_sumf(excitation, excitation, fixed_vector, pitch_gain, gain_code, SUBFR_SIZE); pitch_gain *= 0.5 * pitch_gain; pitch_gain = FFMIN(pitch_gain, 0.4); ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain; ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain); gain_code *= ctx->gain_mem; for (j = 0; j < SUBFR_SIZE; j++) fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j]; if (ctx->mode == MODE_5k0) { postfilter_5k0(ctx, pAz, fixed_vector); ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE, pAz, excitation, SUBFR_SIZE, LP_FILTER_ORDER); } ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector, SUBFR_SIZE, LP_FILTER_ORDER); excitation += SUBFR_SIZE; } memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER, LP_FILTER_ORDER * sizeof(float)); if (ctx->mode == MODE_5k0) { for (i = 0; i < subframe_count; i++) { float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE, ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE, SUBFR_SIZE); ff_adaptative_gain_control(&synth[i * SUBFR_SIZE], energy, SUBFR_SIZE, 0.9, &ctx->postfilter_agc); } memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size, LP_FILTER_ORDER*sizeof(float)); } memcpy(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL, (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float)); ff_acelp_apply_order_2_transfer_function(synth, (const float[2]) {-1.99997 , 1.000000000}, (const float[2]) {-1.93307352, 0.935891986}, 0.939805806, ctx->highpass_filt_mem, frame_size); ctx->dsp.vector_clipf(out_data, synth, -1, 32767./(1<<15), frame_size); } static av_cold int sipr_decoder_init(AVCodecContext * avctx) { SiprContext *ctx = avctx->priv_data; int i; if (avctx->bit_rate > 12200) ctx->mode = MODE_16k; else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5; else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5; else ctx->mode = MODE_5k0; av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name); if (ctx->mode == MODE_16k) ff_sipr_init_16k(ctx); for (i = 0; i < LP_FILTER_ORDER; i++) ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1)); for (i = 0; i < 4; i++) ctx->energy_history[i] = -14; avctx->sample_fmt = SAMPLE_FMT_FLT; dsputil_init(&ctx->dsp, avctx); return 0; } static int sipr_decode_frame(AVCodecContext *avctx, void *datap, int *data_size, AVPacket *avpkt) { SiprContext *ctx = avctx->priv_data; const uint8_t *buf=avpkt->data; SiprParameters parm; const SiprModeParam *mode_par = &modes[ctx->mode]; GetBitContext gb; float *data = datap; int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE; int i; ctx->avctx = avctx; if (avpkt->size < (mode_par->bits_per_frame >> 3)) { av_log(avctx, AV_LOG_ERROR, "Error processing packet: packet size (%d) too small\n", avpkt->size); *data_size = 0; return -1; } if (*data_size < subframe_size * mode_par->subframe_count * sizeof(float)) { av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer (%d) too small\n", *data_size); *data_size = 0; return -1; } init_get_bits(&gb, buf, mode_par->bits_per_frame); for (i = 0; i < mode_par->frames_per_packet; i++) { decode_parameters(&parm, &gb, mode_par); if (ctx->mode == MODE_16k) ff_sipr_decode_frame_16k(ctx, &parm, data); else decode_frame(ctx, &parm, data); data += subframe_size * mode_par->subframe_count; } *data_size = mode_par->frames_per_packet * subframe_size * mode_par->subframe_count * sizeof(float); return mode_par->bits_per_frame >> 3; }; AVCodec sipr_decoder = { "sipr", CODEC_TYPE_AUDIO, CODEC_ID_SIPR, sizeof(SiprContext), sipr_decoder_init, NULL, NULL, sipr_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"), };