From aac996cc01042194bf621d845bbe684549b5882e Mon Sep 17 00:00:00 2001 From: Vittorio Giovara Date: Mon, 23 Nov 2015 17:10:52 -0500 Subject: g723_1: Rename files to better reflect their purpose Signed-off-by: Vittorio Giovara --- libavcodec/g723_1.c | 1378 --------------------------------------------------- 1 file changed, 1378 deletions(-) delete mode 100644 libavcodec/g723_1.c (limited to 'libavcodec/g723_1.c') diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c deleted file mode 100644 index 80c7d1f491..0000000000 --- a/libavcodec/g723_1.c +++ /dev/null @@ -1,1378 +0,0 @@ -/* - * G.723.1 compatible decoder - * Copyright (c) 2006 Benjamin Larsson - * Copyright (c) 2010 Mohamed Naufal Basheer - * - * This file is part of Libav. - * - * 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. - * - * 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 Libav; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/** - * @file - * G.723.1 compatible decoder - */ - -#define BITSTREAM_READER_LE -#include "libavutil/channel_layout.h" -#include "libavutil/mem.h" -#include "libavutil/opt.h" -#include "avcodec.h" -#include "get_bits.h" -#include "acelp_vectors.h" -#include "celp_filters.h" -#include "g723_1_data.h" -#include "internal.h" - -#define CNG_RANDOM_SEED 12345 - -/** - * G723.1 frame types - */ -enum FrameType { - ACTIVE_FRAME, ///< Active speech - SID_FRAME, ///< Silence Insertion Descriptor frame - UNTRANSMITTED_FRAME -}; - -enum Rate { - RATE_6300, - RATE_5300 -}; - -/** - * G723.1 unpacked data subframe - */ -typedef struct G723_1_Subframe { - int ad_cb_lag; ///< adaptive codebook lag - int ad_cb_gain; - int dirac_train; - int pulse_sign; - int grid_index; - int amp_index; - int pulse_pos; -} G723_1_Subframe; - -/** - * Pitch postfilter parameters - */ -typedef struct PPFParam { - int index; ///< postfilter backward/forward lag - int16_t opt_gain; ///< optimal gain - int16_t sc_gain; ///< scaling gain -} PPFParam; - -typedef struct g723_1_context { - AVClass *class; - - G723_1_Subframe subframe[4]; - enum FrameType cur_frame_type; - enum FrameType past_frame_type; - enum Rate cur_rate; - uint8_t lsp_index[LSP_BANDS]; - int pitch_lag[2]; - int erased_frames; - - int16_t prev_lsp[LPC_ORDER]; - int16_t sid_lsp[LPC_ORDER]; - int16_t prev_excitation[PITCH_MAX]; - int16_t excitation[PITCH_MAX + FRAME_LEN + 4]; - int16_t synth_mem[LPC_ORDER]; - int16_t fir_mem[LPC_ORDER]; - int iir_mem[LPC_ORDER]; - - int random_seed; - int cng_random_seed; - int interp_index; - int interp_gain; - int sid_gain; - int cur_gain; - int reflection_coef; - int pf_gain; - int postfilter; - - int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4]; -} G723_1_Context; - -static av_cold int g723_1_decode_init(AVCodecContext *avctx) -{ - G723_1_Context *p = avctx->priv_data; - - avctx->channel_layout = AV_CH_LAYOUT_MONO; - avctx->sample_fmt = AV_SAMPLE_FMT_S16; - avctx->channels = 1; - avctx->sample_rate = 8000; - p->pf_gain = 1 << 12; - - memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); - memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp)); - - p->cng_random_seed = CNG_RANDOM_SEED; - p->past_frame_type = SID_FRAME; - - return 0; -} - -/** - * Unpack the frame into parameters. - * - * @param p the context - * @param buf pointer to the input buffer - * @param buf_size size of the input buffer - */ -static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, - int buf_size) -{ - GetBitContext gb; - int ad_cb_len; - int temp, info_bits, i; - - init_get_bits(&gb, buf, buf_size * 8); - - /* Extract frame type and rate info */ - info_bits = get_bits(&gb, 2); - - if (info_bits == 3) { - p->cur_frame_type = UNTRANSMITTED_FRAME; - return 0; - } - - /* Extract 24 bit lsp indices, 8 bit for each band */ - p->lsp_index[2] = get_bits(&gb, 8); - p->lsp_index[1] = get_bits(&gb, 8); - p->lsp_index[0] = get_bits(&gb, 8); - - if (info_bits == 2) { - p->cur_frame_type = SID_FRAME; - p->subframe[0].amp_index = get_bits(&gb, 6); - return 0; - } - - /* Extract the info common to both rates */ - p->cur_rate = info_bits ? RATE_5300 : RATE_6300; - p->cur_frame_type = ACTIVE_FRAME; - - p->pitch_lag[0] = get_bits(&gb, 7); - if (p->pitch_lag[0] > 123) /* test if forbidden code */ - return -1; - p->pitch_lag[0] += PITCH_MIN; - p->subframe[1].ad_cb_lag = get_bits(&gb, 2); - - p->pitch_lag[1] = get_bits(&gb, 7); - if (p->pitch_lag[1] > 123) - return -1; - p->pitch_lag[1] += PITCH_MIN; - p->subframe[3].ad_cb_lag = get_bits(&gb, 2); - p->subframe[0].ad_cb_lag = 1; - p->subframe[2].ad_cb_lag = 1; - - for (i = 0; i < SUBFRAMES; i++) { - /* Extract combined gain */ - temp = get_bits(&gb, 12); - ad_cb_len = 170; - p->subframe[i].dirac_train = 0; - if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) { - p->subframe[i].dirac_train = temp >> 11; - temp &= 0x7FF; - ad_cb_len = 85; - } - p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS); - if (p->subframe[i].ad_cb_gain < ad_cb_len) { - p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain * - GAIN_LEVELS; - } else { - return -1; - } - } - - p->subframe[0].grid_index = get_bits(&gb, 1); - p->subframe[1].grid_index = get_bits(&gb, 1); - p->subframe[2].grid_index = get_bits(&gb, 1); - p->subframe[3].grid_index = get_bits(&gb, 1); - - if (p->cur_rate == RATE_6300) { - skip_bits(&gb, 1); /* skip reserved bit */ - - /* Compute pulse_pos index using the 13-bit combined position index */ - temp = get_bits(&gb, 13); - p->subframe[0].pulse_pos = temp / 810; - - temp -= p->subframe[0].pulse_pos * 810; - p->subframe[1].pulse_pos = FASTDIV(temp, 90); - - temp -= p->subframe[1].pulse_pos * 90; - p->subframe[2].pulse_pos = FASTDIV(temp, 9); - p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9; - - p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) + - get_bits(&gb, 16); - p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) + - get_bits(&gb, 14); - p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) + - get_bits(&gb, 16); - p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) + - get_bits(&gb, 14); - - p->subframe[0].pulse_sign = get_bits(&gb, 6); - p->subframe[1].pulse_sign = get_bits(&gb, 5); - p->subframe[2].pulse_sign = get_bits(&gb, 6); - p->subframe[3].pulse_sign = get_bits(&gb, 5); - } else { /* 5300 bps */ - p->subframe[0].pulse_pos = get_bits(&gb, 12); - p->subframe[1].pulse_pos = get_bits(&gb, 12); - p->subframe[2].pulse_pos = get_bits(&gb, 12); - p->subframe[3].pulse_pos = get_bits(&gb, 12); - - p->subframe[0].pulse_sign = get_bits(&gb, 4); - p->subframe[1].pulse_sign = get_bits(&gb, 4); - p->subframe[2].pulse_sign = get_bits(&gb, 4); - p->subframe[3].pulse_sign = get_bits(&gb, 4); - } - - return 0; -} - -/** - * Bitexact implementation of sqrt(val/2). - */ -static int16_t square_root(int val) -{ - int16_t res = 0; - int16_t exp = 0x4000; - int i; - - for (i = 0; i < 14; i ++) { - int res_exp = res + exp; - if (val >= res_exp * res_exp << 1) - res += exp; - exp >>= 1; - } - return res; -} - -/** - * Calculate the number of left-shifts required for normalizing the input. - * - * @param num input number - * @param width width of the input, 16 bits(0) / 32 bits(1) - */ -static int normalize_bits(int num, int width) -{ - return width - av_log2(num) - 1; -} - -/** - * Scale vector contents based on the largest of their absolutes. - */ -static int scale_vector(int16_t *dst, const int16_t *vector, int length) -{ - int bits, max = 0; - int i; - - - for (i = 0; i < length; i++) - max |= FFABS(vector[i]); - - max = FFMIN(max, 0x7FFF); - bits = normalize_bits(max, 15); - - for (i = 0; i < length; i++) - dst[i] = vector[i] << bits >> 3; - - return bits - 3; -} - -/** - * Perform inverse quantization of LSP frequencies. - * - * @param cur_lsp the current LSP vector - * @param prev_lsp the previous LSP vector - * @param lsp_index VQ indices - * @param bad_frame bad frame flag - */ -static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, - uint8_t *lsp_index, int bad_frame) -{ - int min_dist, pred; - int i, j, temp, stable; - - /* Check for frame erasure */ - if (!bad_frame) { - min_dist = 0x100; - pred = 12288; - } else { - min_dist = 0x200; - pred = 23552; - lsp_index[0] = lsp_index[1] = lsp_index[2] = 0; - } - - /* Get the VQ table entry corresponding to the transmitted index */ - cur_lsp[0] = lsp_band0[lsp_index[0]][0]; - cur_lsp[1] = lsp_band0[lsp_index[0]][1]; - cur_lsp[2] = lsp_band0[lsp_index[0]][2]; - cur_lsp[3] = lsp_band1[lsp_index[1]][0]; - cur_lsp[4] = lsp_band1[lsp_index[1]][1]; - cur_lsp[5] = lsp_band1[lsp_index[1]][2]; - cur_lsp[6] = lsp_band2[lsp_index[2]][0]; - cur_lsp[7] = lsp_band2[lsp_index[2]][1]; - cur_lsp[8] = lsp_band2[lsp_index[2]][2]; - cur_lsp[9] = lsp_band2[lsp_index[2]][3]; - - /* Add predicted vector & DC component to the previously quantized vector */ - for (i = 0; i < LPC_ORDER; i++) { - temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15; - cur_lsp[i] += dc_lsp[i] + temp; - } - - for (i = 0; i < LPC_ORDER; i++) { - cur_lsp[0] = FFMAX(cur_lsp[0], 0x180); - cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00); - - /* Stability check */ - for (j = 1; j < LPC_ORDER; j++) { - temp = min_dist + cur_lsp[j - 1] - cur_lsp[j]; - if (temp > 0) { - temp >>= 1; - cur_lsp[j - 1] -= temp; - cur_lsp[j] += temp; - } - } - stable = 1; - for (j = 1; j < LPC_ORDER; j++) { - temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4; - if (temp > 0) { - stable = 0; - break; - } - } - if (stable) - break; - } - if (!stable) - memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp)); -} - -/** - * Bitexact implementation of 2ab scaled by 1/2^16. - * - * @param a 32 bit multiplicand - * @param b 16 bit multiplier - */ -#define MULL2(a, b) \ - ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) - -/** - * Convert LSP frequencies to LPC coefficients. - * - * @param lpc buffer for LPC coefficients - */ -static void lsp2lpc(int16_t *lpc) -{ - int f1[LPC_ORDER / 2 + 1]; - int f2[LPC_ORDER / 2 + 1]; - int i, j; - - /* Calculate negative cosine */ - for (j = 0; j < LPC_ORDER; j++) { - int index = (lpc[j] >> 7) & 0x1FF; - int offset = lpc[j] & 0x7f; - int temp1 = cos_tab[index] << 16; - int temp2 = (cos_tab[index + 1] - cos_tab[index]) * - ((offset << 8) + 0x80) << 1; - - lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16); - } - - /* - * Compute sum and difference polynomial coefficients - * (bitexact alternative to lsp2poly() in lsp.c) - */ - /* Initialize with values in Q28 */ - f1[0] = 1 << 28; - f1[1] = (lpc[0] << 14) + (lpc[2] << 14); - f1[2] = lpc[0] * lpc[2] + (2 << 28); - - f2[0] = 1 << 28; - f2[1] = (lpc[1] << 14) + (lpc[3] << 14); - f2[2] = lpc[1] * lpc[3] + (2 << 28); - - /* - * Calculate and scale the coefficients by 1/2 in - * each iteration for a final scaling factor of Q25 - */ - for (i = 2; i < LPC_ORDER / 2; i++) { - f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]); - f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]); - - for (j = i; j >= 2; j--) { - f1[j] = MULL2(f1[j - 1], lpc[2 * i]) + - (f1[j] >> 1) + (f1[j - 2] >> 1); - f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) + - (f2[j] >> 1) + (f2[j - 2] >> 1); - } - - f1[0] >>= 1; - f2[0] >>= 1; - f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1; - f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1; - } - - /* Convert polynomial coefficients to LPC coefficients */ - for (i = 0; i < LPC_ORDER / 2; i++) { - int64_t ff1 = f1[i + 1] + f1[i]; - int64_t ff2 = f2[i + 1] - f2[i]; - - lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16; - lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) + - (1 << 15)) >> 16; - } -} - -/** - * Quantize LSP frequencies by interpolation and convert them to - * the corresponding LPC coefficients. - * - * @param lpc buffer for LPC coefficients - * @param cur_lsp the current LSP vector - * @param prev_lsp the previous LSP vector - */ -static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp) -{ - int i; - int16_t *lpc_ptr = lpc; - - /* cur_lsp * 0.25 + prev_lsp * 0.75 */ - ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp, - 4096, 12288, 1 << 13, 14, LPC_ORDER); - ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp, - 8192, 8192, 1 << 13, 14, LPC_ORDER); - ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp, - 12288, 4096, 1 << 13, 14, LPC_ORDER); - memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc)); - - for (i = 0; i < SUBFRAMES; i++) { - lsp2lpc(lpc_ptr); - lpc_ptr += LPC_ORDER; - } -} - -/** - * Generate a train of dirac functions with period as pitch lag. - */ -static void gen_dirac_train(int16_t *buf, int pitch_lag) -{ - int16_t vector[SUBFRAME_LEN]; - int i, j; - - memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector)); - for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) { - for (j = 0; j < SUBFRAME_LEN - i; j++) - buf[i + j] += vector[j]; - } -} - -/** - * Generate fixed codebook excitation vector. - * - * @param vector decoded excitation vector - * @param subfrm current subframe - * @param cur_rate current bitrate - * @param pitch_lag closed loop pitch lag - * @param index current subframe index - */ -static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm, - enum Rate cur_rate, int pitch_lag, int index) -{ - int temp, i, j; - - memset(vector, 0, SUBFRAME_LEN * sizeof(*vector)); - - if (cur_rate == RATE_6300) { - if (subfrm->pulse_pos >= max_pos[index]) - return; - - /* Decode amplitudes and positions */ - j = PULSE_MAX - pulses[index]; - temp = subfrm->pulse_pos; - for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) { - temp -= combinatorial_table[j][i]; - if (temp >= 0) - continue; - temp += combinatorial_table[j++][i]; - if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) { - vector[subfrm->grid_index + GRID_SIZE * i] = - -fixed_cb_gain[subfrm->amp_index]; - } else { - vector[subfrm->grid_index + GRID_SIZE * i] = - fixed_cb_gain[subfrm->amp_index]; - } - if (j == PULSE_MAX) - break; - } - if (subfrm->dirac_train == 1) - gen_dirac_train(vector, pitch_lag); - } else { /* 5300 bps */ - int cb_gain = fixed_cb_gain[subfrm->amp_index]; - int cb_shift = subfrm->grid_index; - int cb_sign = subfrm->pulse_sign; - int cb_pos = subfrm->pulse_pos; - int offset, beta, lag; - - for (i = 0; i < 8; i += 2) { - offset = ((cb_pos & 7) << 3) + cb_shift + i; - vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain; - cb_pos >>= 3; - cb_sign >>= 1; - } - - /* Enhance harmonic components */ - lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag + - subfrm->ad_cb_lag - 1; - beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1]; - - if (lag < SUBFRAME_LEN - 2) { - for (i = lag; i < SUBFRAME_LEN; i++) - vector[i] += beta * vector[i - lag] >> 15; - } - } -} - -/** - * Get delayed contribution from the previous excitation vector. - */ -static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag) -{ - int offset = PITCH_MAX - PITCH_ORDER / 2 - lag; - int i; - - residual[0] = prev_excitation[offset]; - residual[1] = prev_excitation[offset + 1]; - - offset += 2; - for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++) - residual[i] = prev_excitation[offset + (i - 2) % lag]; -} - -static int dot_product(const int16_t *a, const int16_t *b, int length) -{ - int i, sum = 0; - - for (i = 0; i < length; i++) { - int prod = a[i] * b[i]; - sum = av_sat_dadd32(sum, prod); - } - return sum; -} - -/** - * Generate adaptive codebook excitation. - */ -static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, - int pitch_lag, G723_1_Subframe *subfrm, - enum Rate cur_rate) -{ - int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1]; - const int16_t *cb_ptr; - int lag = pitch_lag + subfrm->ad_cb_lag - 1; - - int i; - int sum; - - get_residual(residual, prev_excitation, lag); - - /* Select quantization table */ - if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) - cb_ptr = adaptive_cb_gain85; - else - cb_ptr = adaptive_cb_gain170; - - /* Calculate adaptive vector */ - cb_ptr += subfrm->ad_cb_gain * 20; - for (i = 0; i < SUBFRAME_LEN; i++) { - sum = dot_product(residual + i, cb_ptr, PITCH_ORDER); - vector[i] = av_sat_dadd32(1 << 15, sum) >> 16; - } -} - -/** - * Estimate maximum auto-correlation around pitch lag. - * - * @param buf buffer with offset applied - * @param offset offset of the excitation vector - * @param ccr_max pointer to the maximum auto-correlation - * @param pitch_lag decoded pitch lag - * @param length length of autocorrelation - * @param dir forward lag(1) / backward lag(-1) - */ -static int autocorr_max(const int16_t *buf, int offset, int *ccr_max, - int pitch_lag, int length, int dir) -{ - int limit, ccr, lag = 0; - int i; - - pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag); - if (dir > 0) - limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3); - else - limit = pitch_lag + 3; - - for (i = pitch_lag - 3; i <= limit; i++) { - ccr = dot_product(buf, buf + dir * i, length); - - if (ccr > *ccr_max) { - *ccr_max = ccr; - lag = i; - } - } - return lag; -} - -/** - * Calculate pitch postfilter optimal and scaling gains. - * - * @param lag pitch postfilter forward/backward lag - * @param ppf pitch postfilter parameters - * @param cur_rate current bitrate - * @param tgt_eng target energy - * @param ccr cross-correlation - * @param res_eng residual energy - */ -static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate, - int tgt_eng, int ccr, int res_eng) -{ - int pf_residual; /* square of postfiltered residual */ - int temp1, temp2; - - ppf->index = lag; - - temp1 = tgt_eng * res_eng >> 1; - temp2 = ccr * ccr << 1; - - if (temp2 > temp1) { - if (ccr >= res_eng) { - ppf->opt_gain = ppf_gain_weight[cur_rate]; - } else { - ppf->opt_gain = (ccr << 15) / res_eng * - ppf_gain_weight[cur_rate] >> 15; - } - /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */ - temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1); - temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng; - pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16; - - if (tgt_eng >= pf_residual << 1) { - temp1 = 0x7fff; - } else { - temp1 = (tgt_eng << 14) / pf_residual; - } - - /* scaling_gain = sqrt(tgt_eng/pf_res^2) */ - ppf->sc_gain = square_root(temp1 << 16); - } else { - ppf->opt_gain = 0; - ppf->sc_gain = 0x7fff; - } - - ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15); -} - -/** - * Calculate pitch postfilter parameters. - * - * @param p the context - * @param offset offset of the excitation vector - * @param pitch_lag decoded pitch lag - * @param ppf pitch postfilter parameters - * @param cur_rate current bitrate - */ -static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag, - PPFParam *ppf, enum Rate cur_rate) -{ - - int16_t scale; - int i; - int temp1, temp2; - - /* - * 0 - target energy - * 1 - forward cross-correlation - * 2 - forward residual energy - * 3 - backward cross-correlation - * 4 - backward residual energy - */ - int energy[5] = {0, 0, 0, 0, 0}; - int16_t *buf = p->audio + LPC_ORDER + offset; - int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag, - SUBFRAME_LEN, 1); - int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag, - SUBFRAME_LEN, -1); - - ppf->index = 0; - ppf->opt_gain = 0; - ppf->sc_gain = 0x7fff; - - /* Case 0, Section 3.6 */ - if (!back_lag && !fwd_lag) - return; - - /* Compute target energy */ - energy[0] = dot_product(buf, buf, SUBFRAME_LEN); - - /* Compute forward residual energy */ - if (fwd_lag) - energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN); - - /* Compute backward residual energy */ - if (back_lag) - energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN); - - /* Normalize and shorten */ - temp1 = 0; - for (i = 0; i < 5; i++) - temp1 = FFMAX(energy[i], temp1); - - scale = normalize_bits(temp1, 31); - for (i = 0; i < 5; i++) - energy[i] = (energy[i] << scale) >> 16; - - if (fwd_lag && !back_lag) { /* Case 1 */ - comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1], - energy[2]); - } else if (!fwd_lag) { /* Case 2 */ - comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3], - energy[4]); - } else { /* Case 3 */ - - /* - * Select the largest of energy[1]^2/energy[2] - * and energy[3]^2/energy[4] - */ - temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15); - temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15); - if (temp1 >= temp2) { - comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1], - energy[2]); - } else { - comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3], - energy[4]); - } - } -} - -/** - * Classify frames as voiced/unvoiced. - * - * @param p the context - * @param pitch_lag decoded pitch_lag - * @param exc_eng excitation energy estimation - * @param scale scaling factor of exc_eng - * - * @return residual interpolation index if voiced, 0 otherwise - */ -static int comp_interp_index(G723_1_Context *p, int pitch_lag, - int *exc_eng, int *scale) -{ - int offset = PITCH_MAX + 2 * SUBFRAME_LEN; - int16_t *buf = p->audio + LPC_ORDER; - - int index, ccr, tgt_eng, best_eng, temp; - - *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX); - buf += offset; - - /* Compute maximum backward cross-correlation */ - ccr = 0; - index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1); - ccr = av_sat_add32(ccr, 1 << 15) >> 16; - - /* Compute target energy */ - tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2); - *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16; - - if (ccr <= 0) - return 0; - - /* Compute best energy */ - best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2); - best_eng = av_sat_add32(best_eng, 1 << 15) >> 16; - - temp = best_eng * *exc_eng >> 3; - - if (temp < ccr * ccr) - return index; - else - return 0; -} - -/** - * Peform residual interpolation based on frame classification. - * - * @param buf decoded excitation vector - * @param out output vector - * @param lag decoded pitch lag - * @param gain interpolated gain - * @param rseed seed for random number generator - */ -static void residual_interp(int16_t *buf, int16_t *out, int lag, - int gain, int *rseed) -{ - int i; - if (lag) { /* Voiced */ - int16_t *vector_ptr = buf + PITCH_MAX; - /* Attenuate */ - for (i = 0; i < lag; i++) - out[i] = vector_ptr[i - lag] * 3 >> 2; - av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out), - (FRAME_LEN - lag) * sizeof(*out)); - } else { /* Unvoiced */ - for (i = 0; i < FRAME_LEN; i++) { - *rseed = *rseed * 521 + 259; - out[i] = gain * *rseed >> 15; - } - memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf)); - } -} - -/** - * Perform IIR filtering. - * - * @param fir_coef FIR coefficients - * @param iir_coef IIR coefficients - * @param src source vector - * @param dest destination vector - */ -static inline void iir_filter(int16_t *fir_coef, int16_t *iir_coef, - int16_t *src, int *dest) -{ - int m, n; - - for (m = 0; m < SUBFRAME_LEN; m++) { - int64_t filter = 0; - for (n = 1; n <= LPC_ORDER; n++) { - filter -= fir_coef[n - 1] * src[m - n] - - iir_coef[n - 1] * (dest[m - n] >> 16); - } - - dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + (1 << 15)); - } -} - -/** - * Adjust gain of postfiltered signal. - * - * @param p the context - * @param buf postfiltered output vector - * @param energy input energy coefficient - */ -static void gain_scale(G723_1_Context *p, int16_t * buf, int energy) -{ - int num, denom, gain, bits1, bits2; - int i; - - num = energy; - denom = 0; - for (i = 0; i < SUBFRAME_LEN; i++) { - int temp = buf[i] >> 2; - temp *= temp; - denom = av_sat_dadd32(denom, temp); - } - - if (num && denom) { - bits1 = normalize_bits(num, 31); - bits2 = normalize_bits(denom, 31); - num = num << bits1 >> 1; - denom <<= bits2; - - bits2 = 5 + bits1 - bits2; - bits2 = FFMAX(0, bits2); - - gain = (num >> 1) / (denom >> 16); - gain = square_root(gain << 16 >> bits2); - } else { - gain = 1 << 12; - } - - for (i = 0; i < SUBFRAME_LEN; i++) { - p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4; - buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) + - (1 << 10)) >> 11); - } -} - -/** - * Perform formant filtering. - * - * @param p the context - * @param lpc quantized lpc coefficients - * @param buf input buffer - * @param dst output buffer - */ -static void formant_postfilter(G723_1_Context *p, int16_t *lpc, - int16_t *buf, int16_t *dst) -{ - int16_t filter_coef[2][LPC_ORDER]; - int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr; - int i, j, k; - - memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf)); - memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal)); - - for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { - for (k = 0; k < LPC_ORDER; k++) { - filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] + - (1 << 14)) >> 15; - filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] + - (1 << 14)) >> 15; - } - iir_filter(filter_coef[0], filter_coef[1], buf + i, - filter_signal + i); - lpc += LPC_ORDER; - } - - memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(*p->fir_mem)); - memcpy(p->iir_mem, filter_signal + FRAME_LEN, - LPC_ORDER * sizeof(*p->iir_mem)); - - buf += LPC_ORDER; - signal_ptr = filter_signal + LPC_ORDER; - for (i = 0; i < SUBFRAMES; i++) { - int temp; - int auto_corr[2]; - int scale, energy; - - /* Normalize */ - scale = scale_vector(dst, buf, SUBFRAME_LEN); - - /* Compute auto correlation coefficients */ - auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1); - auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN); - - /* Compute reflection coefficient */ - temp = auto_corr[1] >> 16; - if (temp) { - temp = (auto_corr[0] >> 2) / temp; - } - p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2; - temp = -p->reflection_coef >> 1 & ~3; - - /* Compensation filter */ - for (j = 0; j < SUBFRAME_LEN; j++) { - dst[j] = av_sat_dadd32(signal_ptr[j], - (signal_ptr[j - 1] >> 16) * temp) >> 16; - } - - /* Compute normalized signal energy */ - temp = 2 * scale + 4; - if (temp < 0) { - energy = av_clipl_int32((int64_t)auto_corr[1] << -temp); - } else - energy = auto_corr[1] >> temp; - - gain_scale(p, dst, energy); - - buf += SUBFRAME_LEN; - signal_ptr += SUBFRAME_LEN; - dst += SUBFRAME_LEN; - } -} - -static int sid_gain_to_lsp_index(int gain) -{ - if (gain < 0x10) - return gain << 6; - else if (gain < 0x20) - return gain - 8 << 7; - else - return gain - 20 << 8; -} - -static inline int cng_rand(int *state, int base) -{ - *state = (*state * 521 + 259) & 0xFFFF; - return (*state & 0x7FFF) * base >> 15; -} - -static int estimate_sid_gain(G723_1_Context *p) -{ - int i, shift, seg, seg2, t, val, val_add, x, y; - - shift = 16 - p->cur_gain * 2; - if (shift > 0) - t = p->sid_gain << shift; - else - t = p->sid_gain >> -shift; - x = t * cng_filt[0] >> 16; - - if (x >= cng_bseg[2]) - return 0x3F; - - if (x >= cng_bseg[1]) { - shift = 4; - seg = 3; - } else { - shift = 3; - seg = (x >= cng_bseg[0]); - } - seg2 = FFMIN(seg, 3); - - val = 1 << shift; - val_add = val >> 1; - for (i = 0; i < shift; i++) { - t = seg * 32 + (val << seg2); - t *= t; - if (x >= t) - val += val_add; - else - val -= val_add; - val_add >>= 1; - } - - t = seg * 32 + (val << seg2); - y = t * t - x; - if (y <= 0) { - t = seg * 32 + (val + 1 << seg2); - t = t * t - x; - val = (seg2 - 1 << 4) + val; - if (t >= y) - val++; - } else { - t = seg * 32 + (val - 1 << seg2); - t = t * t - x; - val = (seg2 - 1 << 4) + val; - if (t >= y) - val--; - } - - return val; -} - -static void generate_noise(G723_1_Context *p) -{ - int i, j, idx, t; - int off[SUBFRAMES]; - int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11]; - int tmp[SUBFRAME_LEN * 2]; - int16_t *vector_ptr; - int64_t sum; - int b0, c, delta, x, shift; - - p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123; - p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123; - - for (i = 0; i < SUBFRAMES; i++) { - p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1; - p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i]; - } - - for (i = 0; i < SUBFRAMES / 2; i++) { - t = cng_rand(&p->cng_random_seed, 1 << 13); - off[i * 2] = t & 1; - off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN; - t >>= 2; - for (j = 0; j < 11; j++) { - signs[i * 11 + j] = (t & 1) * 2 - 1 << 14; - t >>= 1; - } - } - - idx = 0; - for (i = 0; i < SUBFRAMES; i++) { - for (j = 0; j < SUBFRAME_LEN / 2; j++) - tmp[j] = j; - t = SUBFRAME_LEN / 2; - for (j = 0; j < pulses[i]; j++, idx++) { - int idx2 = cng_rand(&p->cng_random_seed, t); - - pos[idx] = tmp[idx2] * 2 + off[i]; - tmp[idx2] = tmp[--t]; - } - } - - vector_ptr = p->audio + LPC_ORDER; - memcpy(vector_ptr, p->prev_excitation, - PITCH_MAX * sizeof(*p->excitation)); - for (i = 0; i < SUBFRAMES; i += 2) { - gen_acb_excitation(vector_ptr, vector_ptr, - p->pitch_lag[i >> 1], &p->subframe[i], - p->cur_rate); - gen_acb_excitation(vector_ptr + SUBFRAME_LEN, - vector_ptr + SUBFRAME_LEN, - p->pitch_lag[i >> 1], &p->subframe[i + 1], - p->cur_rate); - - t = 0; - for (j = 0; j < SUBFRAME_LEN * 2; j++) - t |= FFABS(vector_ptr[j]); - t = FFMIN(t, 0x7FFF); - if (!t) { - shift = 0; - } else { - shift = -10 + av_log2(t); - if (shift < -2) - shift = -2; - } - sum = 0; - if (shift < 0) { - for (j = 0; j < SUBFRAME_LEN * 2; j++) { - t = vector_ptr[j] << -shift; - sum += t * t; - tmp[j] = t; - } - } else { - for (j = 0; j < SUBFRAME_LEN * 2; j++) { - t = vector_ptr[j] >> shift; - sum += t * t; - tmp[j] = t; - } - } - - b0 = 0; - for (j = 0; j < 11; j++) - b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j]; - b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11 - - c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5); - if (shift * 2 + 3 >= 0) - c >>= shift * 2 + 3; - else - c <<= -(shift * 2 + 3); - c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15; - - delta = b0 * b0 * 2 - c; - if (delta <= 0) { - x = -b0; - } else { - delta = square_root(delta); - x = delta - b0; - t = delta + b0; - if (FFABS(t) < FFABS(x)) - x = -t; - } - shift++; - if (shift < 0) - x >>= -shift; - else - x <<= shift; - x = av_clip(x, -10000, 10000); - - for (j = 0; j < 11; j++) { - idx = (i / 2) * 11 + j; - vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] + - (x * signs[idx] >> 15)); - } - - /* copy decoded data to serve as a history for the next decoded subframes */ - memcpy(vector_ptr + PITCH_MAX, vector_ptr, - sizeof(*vector_ptr) * SUBFRAME_LEN * 2); - vector_ptr += SUBFRAME_LEN * 2; - } - /* Save the excitation for the next frame */ - memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN, - PITCH_MAX * sizeof(*p->excitation)); -} - -static int g723_1_decode_frame(AVCodecContext *avctx, void *data, - int *got_frame_ptr, AVPacket *avpkt) -{ - G723_1_Context *p = avctx->priv_data; - AVFrame *frame = data; - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - int dec_mode = buf[0] & 3; - - PPFParam ppf[SUBFRAMES]; - int16_t cur_lsp[LPC_ORDER]; - int16_t lpc[SUBFRAMES * LPC_ORDER]; - int16_t acb_vector[SUBFRAME_LEN]; - int16_t *out; - int bad_frame = 0, i, j, ret; - int16_t *audio = p->audio; - - if (buf_size < frame_size[dec_mode]) { - if (buf_size) - av_log(avctx, AV_LOG_WARNING, - "Expected %d bytes, got %d - skipping packet\n", - frame_size[dec_mode], buf_size); - *got_frame_ptr = 0; - return buf_size; - } - - if (unpack_bitstream(p, buf, buf_size) < 0) { - bad_frame = 1; - if (p->past_frame_type == ACTIVE_FRAME) - p->cur_frame_type = ACTIVE_FRAME; - else - p->cur_frame_type = UNTRANSMITTED_FRAME; - } - - frame->nb_samples = FRAME_LEN; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - return ret; - } - - out = (int16_t *)frame->data[0]; - - if (p->cur_frame_type == ACTIVE_FRAME) { - if (!bad_frame) - p->erased_frames = 0; - else if (p->erased_frames != 3) - p->erased_frames++; - - inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame); - lsp_interpolate(lpc, cur_lsp, p->prev_lsp); - - /* Save the lsp_vector for the next frame */ - memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); - - /* Generate the excitation for the frame */ - memcpy(p->excitation, p->prev_excitation, - PITCH_MAX * sizeof(*p->excitation)); - if (!p->erased_frames) { - int16_t *vector_ptr = p->excitation + PITCH_MAX; - - /* Update interpolation gain memory */ - p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index + - p->subframe[3].amp_index) >> 1]; - for (i = 0; i < SUBFRAMES; i++) { - gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate, - p->pitch_lag[i >> 1], i); - gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i], - p->pitch_lag[i >> 1], &p->subframe[i], - p->cur_rate); - /* Get the total excitation */ - for (j = 0; j < SUBFRAME_LEN; j++) { - int v = av_clip_int16(vector_ptr[j] << 1); - vector_ptr[j] = av_clip_int16(v + acb_vector[j]); - } - vector_ptr += SUBFRAME_LEN; - } - - vector_ptr = p->excitation + PITCH_MAX; - - p->interp_index = comp_interp_index(p, p->pitch_lag[1], - &p->sid_gain, &p->cur_gain); - - /* Peform pitch postfiltering */ - if (p->postfilter) { - i = PITCH_MAX; - for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) - comp_ppf_coeff(p, i, p->pitch_lag[j >> 1], - ppf + j, p->cur_rate); - - for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) - ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i, - vector_ptr + i, - vector_ptr + i + ppf[j].index, - ppf[j].sc_gain, - ppf[j].opt_gain, - 1 << 14, 15, SUBFRAME_LEN); - } else { - audio = vector_ptr - LPC_ORDER; - } - - /* Save the excitation for the next frame */ - memcpy(p->prev_excitation, p->excitation + FRAME_LEN, - PITCH_MAX * sizeof(*p->excitation)); - } else { - p->interp_gain = (p->interp_gain * 3 + 2) >> 2; - if (p->erased_frames == 3) { - /* Mute output */ - memset(p->excitation, 0, - (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation)); - memset(p->prev_excitation, 0, - PITCH_MAX * sizeof(*p->excitation)); - memset(frame->data[0], 0, - (FRAME_LEN + LPC_ORDER) * sizeof(int16_t)); - } else { - int16_t *buf = p->audio + LPC_ORDER; - - /* Regenerate frame */ - residual_interp(p->excitation, buf, p->interp_index, - p->interp_gain, &p->random_seed); - - /* Save the excitation for the next frame */ - memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX), - PITCH_MAX * sizeof(*p->excitation)); - } - } - p->cng_random_seed = CNG_RANDOM_SEED; - } else { - if (p->cur_frame_type == SID_FRAME) { - p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index); - inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0); - } else if (p->past_frame_type == ACTIVE_FRAME) { - p->sid_gain = estimate_sid_gain(p); - } - - if (p->past_frame_type == ACTIVE_FRAME) - p->cur_gain = p->sid_gain; - else - p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3; - generate_noise(p); - lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp); - /* Save the lsp_vector for the next frame */ - memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); - } - - p->past_frame_type = p->cur_frame_type; - - memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio)); - for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) - ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER], - audio + i, SUBFRAME_LEN, LPC_ORDER, - 0, 1, 1 << 12); - memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio)); - - if (p->postfilter) { - formant_postfilter(p, lpc, p->audio, out); - } else { // if output is not postfiltered it should be scaled by 2 - for (i = 0; i < FRAME_LEN; i++) - out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1); - } - - *got_frame_ptr = 1; - - return frame_size[dec_mode]; -} - -#define OFFSET(x) offsetof(G723_1_Context, x) -#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM - -static const AVOption options[] = { - { "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT, - { .i64 = 1 }, 0, 1, AD }, - { NULL } -}; - - -static const AVClass g723_1dec_class = { - .class_name = "G.723.1 decoder", - .item_name = av_default_item_name, - .option = options, - .version = LIBAVUTIL_VERSION_INT, -}; - -AVCodec ff_g723_1_decoder = { - .name = "g723_1", - .long_name = NULL_IF_CONFIG_SMALL("G.723.1"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_G723_1, - .priv_data_size = sizeof(G723_1_Context), - .init = g723_1_decode_init, - .decode = g723_1_decode_frame, - .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, - .priv_class = &g723_1dec_class, -}; -- cgit v1.2.3