| /* |
| * G.723.1 compatible decoder |
| * Copyright (c) 2006 Benjamin Larsson |
| * Copyright (c) 2010 Mohamed Naufal Basheer |
| * |
| * 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 |
| * 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 "celp_math.h" |
| #include "g723_1_data.h" |
| #include "internal.h" |
| |
| #define CNG_RANDOM_SEED 12345 |
| |
| 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; ///< formant postfilter |
| ///< gain scaling unit memory |
| int postfilter; |
| |
| int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4]; |
| int16_t prev_data[HALF_FRAME_LEN]; |
| int16_t prev_weight_sig[PITCH_MAX]; |
| |
| |
| int16_t hpf_fir_mem; ///< highpass filter fir |
| int hpf_iir_mem; ///< and iir memories |
| int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir |
| int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories |
| |
| int16_t harmonic_mem[PITCH_MAX]; |
| } 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; |
| 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_bits1(&gb); |
| p->subframe[1].grid_index = get_bits1(&gb); |
| p->subframe[2].grid_index = get_bits1(&gb); |
| p->subframe[3].grid_index = get_bits1(&gb); |
| |
| if (p->cur_rate == RATE_6300) { |
| skip_bits1(&gb); /* 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(unsigned val) |
| { |
| av_assert2(!(val & 0x80000000)); |
| |
| return (ff_sqrt(val << 1) >> 1) & (~1); |
| } |
| |
| /** |
| * Calculate the number of left-shifts required for normalizing the input. |
| * |
| * @param num input number |
| * @param width width of the input, 15 or 31 bits |
| */ |
| static int normalize_bits(int num, int width) |
| { |
| return width - av_log2(num) - 1; |
| } |
| |
| #define normalize_bits_int16(num) normalize_bits(num, 15) |
| #define normalize_bits_int32(num) normalize_bits(num, 31) |
| |
| /** |
| * 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]); |
| |
| bits= 14 - av_log2_16bit(max); |
| bits= FFMAX(bits, 0); |
| |
| 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) \ |
| MULL(a,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 sum = ff_dot_product(a,b,length); |
| return av_sat_add32(sum, 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 = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER); |
| vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, 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 |
| * @param width width of the output, 16 bits(0) / 32 bits(1) |
| */ |
| #define iir_filter(fir_coef, iir_coef, src, dest, width)\ |
| {\ |
| int m, n;\ |
| int res_shift = 16 & ~-(width);\ |
| int in_shift = 16 - res_shift;\ |
| \ |
| 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] >> in_shift);\ |
| }\ |
| \ |
| (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\ |
| (1 << 15)) >> res_shift;\ |
| }\ |
| } |
| |
| /** |
| * 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, 1); |
| lpc += LPC_ORDER; |
| } |
| |
| memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t)); |
| memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int)); |
| |
| 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) |
| 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 = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, |
| .priv_class = &g723_1dec_class, |
| }; |
| |
| #if CONFIG_G723_1_ENCODER |
| #define BITSTREAM_WRITER_LE |
| #include "put_bits.h" |
| |
| static av_cold int g723_1_encode_init(AVCodecContext *avctx) |
| { |
| G723_1_Context *p = avctx->priv_data; |
| |
| if (avctx->sample_rate != 8000) { |
| av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n"); |
| return -1; |
| } |
| |
| if (avctx->channels != 1) { |
| av_log(avctx, AV_LOG_ERROR, "Only mono supported\n"); |
| return AVERROR(EINVAL); |
| } |
| |
| if (avctx->bit_rate == 6300) { |
| p->cur_rate = RATE_6300; |
| } else if (avctx->bit_rate == 5300) { |
| av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n"); |
| return AVERROR_PATCHWELCOME; |
| } else { |
| av_log(avctx, AV_LOG_ERROR, |
| "Bitrate not supported, use 6.3k\n"); |
| return AVERROR(EINVAL); |
| } |
| avctx->frame_size = 240; |
| memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t)); |
| |
| return 0; |
| } |
| |
| /** |
| * Remove DC component from the input signal. |
| * |
| * @param buf input signal |
| * @param fir zero memory |
| * @param iir pole memory |
| */ |
| static void highpass_filter(int16_t *buf, int16_t *fir, int *iir) |
| { |
| int i; |
| for (i = 0; i < FRAME_LEN; i++) { |
| *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00); |
| *fir = buf[i]; |
| buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16; |
| } |
| } |
| |
| /** |
| * Estimate autocorrelation of the input vector. |
| * |
| * @param buf input buffer |
| * @param autocorr autocorrelation coefficients vector |
| */ |
| static void comp_autocorr(int16_t *buf, int16_t *autocorr) |
| { |
| int i, scale, temp; |
| int16_t vector[LPC_FRAME]; |
| |
| scale_vector(vector, buf, LPC_FRAME); |
| |
| /* Apply the Hamming window */ |
| for (i = 0; i < LPC_FRAME; i++) |
| vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15; |
| |
| /* Compute the first autocorrelation coefficient */ |
| temp = ff_dot_product(vector, vector, LPC_FRAME); |
| |
| /* Apply a white noise correlation factor of (1025/1024) */ |
| temp += temp >> 10; |
| |
| /* Normalize */ |
| scale = normalize_bits_int32(temp); |
| autocorr[0] = av_clipl_int32((int64_t)(temp << scale) + |
| (1 << 15)) >> 16; |
| |
| /* Compute the remaining coefficients */ |
| if (!autocorr[0]) { |
| memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t)); |
| } else { |
| for (i = 1; i <= LPC_ORDER; i++) { |
| temp = ff_dot_product(vector, vector + i, LPC_FRAME - i); |
| temp = MULL2((temp << scale), binomial_window[i - 1]); |
| autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16; |
| } |
| } |
| } |
| |
| /** |
| * Use Levinson-Durbin recursion to compute LPC coefficients from |
| * autocorrelation values. |
| * |
| * @param lpc LPC coefficients vector |
| * @param autocorr autocorrelation coefficients vector |
| * @param error prediction error |
| */ |
| static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error) |
| { |
| int16_t vector[LPC_ORDER]; |
| int16_t partial_corr; |
| int i, j, temp; |
| |
| memset(lpc, 0, LPC_ORDER * sizeof(int16_t)); |
| |
| for (i = 0; i < LPC_ORDER; i++) { |
| /* Compute the partial correlation coefficient */ |
| temp = 0; |
| for (j = 0; j < i; j++) |
| temp -= lpc[j] * autocorr[i - j - 1]; |
| temp = ((autocorr[i] << 13) + temp) << 3; |
| |
| if (FFABS(temp) >= (error << 16)) |
| break; |
| |
| partial_corr = temp / (error << 1); |
| |
| lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) + |
| (1 << 15)) >> 16; |
| |
| /* Update the prediction error */ |
| temp = MULL2(temp, partial_corr); |
| error = av_clipl_int32((int64_t)(error << 16) - temp + |
| (1 << 15)) >> 16; |
| |
| memcpy(vector, lpc, i * sizeof(int16_t)); |
| for (j = 0; j < i; j++) { |
| temp = partial_corr * vector[i - j - 1] << 1; |
| lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp + |
| (1 << 15)) >> 16; |
| } |
| } |
| } |
| |
| /** |
| * Calculate LPC coefficients for the current frame. |
| * |
| * @param buf current frame |
| * @param prev_data 2 trailing subframes of the previous frame |
| * @param lpc LPC coefficients vector |
| */ |
| static void comp_lpc_coeff(int16_t *buf, int16_t *lpc) |
| { |
| int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES]; |
| int16_t *autocorr_ptr = autocorr; |
| int16_t *lpc_ptr = lpc; |
| int i, j; |
| |
| for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { |
| comp_autocorr(buf + i, autocorr_ptr); |
| levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]); |
| |
| lpc_ptr += LPC_ORDER; |
| autocorr_ptr += LPC_ORDER + 1; |
| } |
| } |
| |
| static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp) |
| { |
| int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference |
| ///< polynomials (F1, F2) ordered as |
| ///< f1[0], f2[0], ...., f1[5], f2[5] |
| |
| int max, shift, cur_val, prev_val, count, p; |
| int i, j; |
| int64_t temp; |
| |
| /* Initialize f1[0] and f2[0] to 1 in Q25 */ |
| for (i = 0; i < LPC_ORDER; i++) |
| lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15; |
| |
| /* Apply bandwidth expansion on the LPC coefficients */ |
| f[0] = f[1] = 1 << 25; |
| |
| /* Compute the remaining coefficients */ |
| for (i = 0; i < LPC_ORDER / 2; i++) { |
| /* f1 */ |
| f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12); |
| /* f2 */ |
| f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12); |
| } |
| |
| /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */ |
| f[LPC_ORDER] >>= 1; |
| f[LPC_ORDER + 1] >>= 1; |
| |
| /* Normalize and shorten */ |
| max = FFABS(f[0]); |
| for (i = 1; i < LPC_ORDER + 2; i++) |
| max = FFMAX(max, FFABS(f[i])); |
| |
| shift = normalize_bits_int32(max); |
| |
| for (i = 0; i < LPC_ORDER + 2; i++) |
| f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16; |
| |
| /** |
| * Evaluate F1 and F2 at uniform intervals of pi/256 along the |
| * unit circle and check for zero crossings. |
| */ |
| p = 0; |
| temp = 0; |
| for (i = 0; i <= LPC_ORDER / 2; i++) |
| temp += f[2 * i] * cos_tab[0]; |
| prev_val = av_clipl_int32(temp << 1); |
| count = 0; |
| for ( i = 1; i < COS_TBL_SIZE / 2; i++) { |
| /* Evaluate */ |
| temp = 0; |
| for (j = 0; j <= LPC_ORDER / 2; j++) |
| temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE]; |
| cur_val = av_clipl_int32(temp << 1); |
| |
| /* Check for sign change, indicating a zero crossing */ |
| if ((cur_val ^ prev_val) < 0) { |
| int abs_cur = FFABS(cur_val); |
| int abs_prev = FFABS(prev_val); |
| int sum = abs_cur + abs_prev; |
| |
| shift = normalize_bits_int32(sum); |
| sum <<= shift; |
| abs_prev = abs_prev << shift >> 8; |
| lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16); |
| |
| if (count == LPC_ORDER) |
| break; |
| |
| /* Switch between sum and difference polynomials */ |
| p ^= 1; |
| |
| /* Evaluate */ |
| temp = 0; |
| for (j = 0; j <= LPC_ORDER / 2; j++){ |
| temp += f[LPC_ORDER - 2 * j + p] * |
| cos_tab[i * j % COS_TBL_SIZE]; |
| } |
| cur_val = av_clipl_int32(temp<<1); |
| } |
| prev_val = cur_val; |
| } |
| |
| if (count != LPC_ORDER) |
| memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t)); |
| } |
| |
| /** |
| * Quantize the current LSP subvector. |
| * |
| * @param num band number |
| * @param offset offset of the current subvector in an LPC_ORDER vector |
| * @param size size of the current subvector |
| */ |
| #define get_index(num, offset, size) \ |
| {\ |
| int error, max = -1;\ |
| int16_t temp[4];\ |
| int i, j;\ |
| for (i = 0; i < LSP_CB_SIZE; i++) {\ |
| for (j = 0; j < size; j++){\ |
| temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\ |
| (1 << 14)) >> 15;\ |
| }\ |
| error = dot_product(lsp + (offset), temp, size) << 1;\ |
| error -= dot_product(lsp_band##num[i], temp, size);\ |
| if (error > max) {\ |
| max = error;\ |
| lsp_index[num] = i;\ |
| }\ |
| }\ |
| } |
| |
| /** |
| * Vector quantize the LSP frequencies. |
| * |
| * @param lsp the current lsp vector |
| * @param prev_lsp the previous lsp vector |
| */ |
| static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp) |
| { |
| int16_t weight[LPC_ORDER]; |
| int16_t min, max; |
| int shift, i; |
| |
| /* Calculate the VQ weighting vector */ |
| weight[0] = (1 << 20) / (lsp[1] - lsp[0]); |
| weight[LPC_ORDER - 1] = (1 << 20) / |
| (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]); |
| |
| for (i = 1; i < LPC_ORDER - 1; i++) { |
| min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]); |
| if (min > 0x20) |
| weight[i] = (1 << 20) / min; |
| else |
| weight[i] = INT16_MAX; |
| } |
| |
| /* Normalize */ |
| max = 0; |
| for (i = 0; i < LPC_ORDER; i++) |
| max = FFMAX(weight[i], max); |
| |
| shift = normalize_bits_int16(max); |
| for (i = 0; i < LPC_ORDER; i++) { |
| weight[i] <<= shift; |
| } |
| |
| /* Compute the VQ target vector */ |
| for (i = 0; i < LPC_ORDER; i++) { |
| lsp[i] -= dc_lsp[i] + |
| (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15); |
| } |
| |
| get_index(0, 0, 3); |
| get_index(1, 3, 3); |
| get_index(2, 6, 4); |
| } |
| |
| /** |
| * Apply the formant perceptual weighting filter. |
| * |
| * @param flt_coef filter coefficients |
| * @param unq_lpc unquantized lpc vector |
| */ |
| static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef, |
| int16_t *unq_lpc, int16_t *buf) |
| { |
| int16_t vector[FRAME_LEN + LPC_ORDER]; |
| int i, j, k, l = 0; |
| |
| memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER); |
| memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER); |
| memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN); |
| |
| for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { |
| for (k = 0; k < LPC_ORDER; k++) { |
| flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] + |
| (1 << 14)) >> 15; |
| flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] * |
| percept_flt_tbl[1][k] + |
| (1 << 14)) >> 15; |
| } |
| iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i, |
| buf + i, 0); |
| l += LPC_ORDER; |
| } |
| memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER); |
| memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER); |
| } |
| |
| /** |
| * Estimate the open loop pitch period. |
| * |
| * @param buf perceptually weighted speech |
| * @param start estimation is carried out from this position |
| */ |
| static int estimate_pitch(int16_t *buf, int start) |
| { |
| int max_exp = 32; |
| int max_ccr = 0x4000; |
| int max_eng = 0x7fff; |
| int index = PITCH_MIN; |
| int offset = start - PITCH_MIN + 1; |
| |
| int ccr, eng, orig_eng, ccr_eng, exp; |
| int diff, temp; |
| |
| int i; |
| |
| orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN); |
| |
| for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) { |
| offset--; |
| |
| /* Update energy and compute correlation */ |
| orig_eng += buf[offset] * buf[offset] - |
| buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN]; |
| ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN); |
| if (ccr <= 0) |
| continue; |
| |
| /* Split into mantissa and exponent to maintain precision */ |
| exp = normalize_bits_int32(ccr); |
| ccr = av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16; |
| exp <<= 1; |
| ccr *= ccr; |
| temp = normalize_bits_int32(ccr); |
| ccr = ccr << temp >> 16; |
| exp += temp; |
| |
| temp = normalize_bits_int32(orig_eng); |
| eng = av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16; |
| exp -= temp; |
| |
| if (ccr >= eng) { |
| exp--; |
| ccr >>= 1; |
| } |
| if (exp > max_exp) |
| continue; |
| |
| if (exp + 1 < max_exp) |
| goto update; |
| |
| /* Equalize exponents before comparison */ |
| if (exp + 1 == max_exp) |
| temp = max_ccr >> 1; |
| else |
| temp = max_ccr; |
| ccr_eng = ccr * max_eng; |
| diff = ccr_eng - eng * temp; |
| if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) { |
| update: |
| index = i; |
| max_exp = exp; |
| max_ccr = ccr; |
| max_eng = eng; |
| } |
| } |
| return index; |
| } |
| |
| /** |
| * Compute harmonic noise filter parameters. |
| * |
| * @param buf perceptually weighted speech |
| * @param pitch_lag open loop pitch period |
| * @param hf harmonic filter parameters |
| */ |
| static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf) |
| { |
| int ccr, eng, max_ccr, max_eng; |
| int exp, max, diff; |
| int energy[15]; |
| int i, j; |
| |
| for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) { |
| /* Compute residual energy */ |
| energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN); |
| /* Compute correlation */ |
| energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN); |
| } |
| |
| /* Compute target energy */ |
| energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN); |
| |
| /* Normalize */ |
| max = 0; |
| for (i = 0; i < 15; i++) |
| max = FFMAX(max, FFABS(energy[i])); |
| |
| exp = normalize_bits_int32(max); |
| for (i = 0; i < 15; i++) { |
| energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) + |
| (1 << 15)) >> 16; |
| } |
| |
| hf->index = -1; |
| hf->gain = 0; |
| max_ccr = 1; |
| max_eng = 0x7fff; |
| |
| for (i = 0; i <= 6; i++) { |
| eng = energy[i << 1]; |
| ccr = energy[(i << 1) + 1]; |
| |
| if (ccr <= 0) |
| continue; |
| |
| ccr = (ccr * ccr + (1 << 14)) >> 15; |
| diff = ccr * max_eng - eng * max_ccr; |
| if (diff > 0) { |
| max_ccr = ccr; |
| max_eng = eng; |
| hf->index = i; |
| } |
| } |
| |
| if (hf->index == -1) { |
| hf->index = pitch_lag; |
| return; |
| } |
| |
| eng = energy[14] * max_eng; |
| eng = (eng >> 2) + (eng >> 3); |
| ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1]; |
| if (eng < ccr) { |
| eng = energy[(hf->index << 1) + 1]; |
| |
| if (eng >= max_eng) |
| hf->gain = 0x2800; |
| else |
| hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15; |
| } |
| hf->index += pitch_lag - 3; |
| } |
| |
| /** |
| * Apply the harmonic noise shaping filter. |
| * |
| * @param hf filter parameters |
| */ |
| static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest) |
| { |
| int i; |
| |
| for (i = 0; i < SUBFRAME_LEN; i++) { |
| int64_t temp = hf->gain * src[i - hf->index] << 1; |
| dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16; |
| } |
| } |
| |
| static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest) |
| { |
| int i; |
| for (i = 0; i < SUBFRAME_LEN; i++) { |
| int64_t temp = hf->gain * src[i - hf->index] << 1; |
| dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp + |
| (1 << 15)) >> 16; |
| |
| } |
| } |
| |
| /** |
| * Combined synthesis and formant perceptual weighting filer. |
| * |
| * @param qnt_lpc quantized lpc coefficients |
| * @param perf_lpc perceptual filter coefficients |
| * @param perf_fir perceptual filter fir memory |
| * @param perf_iir perceptual filter iir memory |
| * @param scale the filter output will be scaled by 2^scale |
| */ |
| static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc, |
| int16_t *perf_fir, int16_t *perf_iir, |
| const int16_t *src, int16_t *dest, int scale) |
| { |
| int i, j; |
| int16_t buf_16[SUBFRAME_LEN + LPC_ORDER]; |
| int64_t buf[SUBFRAME_LEN]; |
| |
| int16_t *bptr_16 = buf_16 + LPC_ORDER; |
| |
| memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER); |
| memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER); |
| |
| for (i = 0; i < SUBFRAME_LEN; i++) { |
| int64_t temp = 0; |
| for (j = 1; j <= LPC_ORDER; j++) |
| temp -= qnt_lpc[j - 1] * bptr_16[i - j]; |
| |
| buf[i] = (src[i] << 15) + (temp << 3); |
| bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16; |
| } |
| |
| for (i = 0; i < SUBFRAME_LEN; i++) { |
| int64_t fir = 0, iir = 0; |
| for (j = 1; j <= LPC_ORDER; j++) { |
| fir -= perf_lpc[j - 1] * bptr_16[i - j]; |
| iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j]; |
| } |
| dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) + |
| (1 << 15)) >> 16; |
| } |
| memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER); |
| memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER, |
| sizeof(int16_t) * LPC_ORDER); |
| } |
| |
| /** |
| * Compute the adaptive codebook contribution. |
| * |
| * @param buf input signal |
| * @param index the current subframe index |
| */ |
| static void acb_search(G723_1_Context *p, int16_t *residual, |
| int16_t *impulse_resp, const int16_t *buf, |
| int index) |
| { |
| |
| int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN]; |
| |
| const int16_t *cb_tbl = adaptive_cb_gain85; |
| |
| int ccr_buf[PITCH_ORDER * SUBFRAMES << 2]; |
| |
| int pitch_lag = p->pitch_lag[index >> 1]; |
| int acb_lag = 1; |
| int acb_gain = 0; |
| int odd_frame = index & 1; |
| int iter = 3 + odd_frame; |
| int count = 0; |
| int tbl_size = 85; |
| |
| int i, j, k, l, max; |
| int64_t temp; |
| |
| if (!odd_frame) { |
| if (pitch_lag == PITCH_MIN) |
| pitch_lag++; |
| else |
| pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5); |
| } |
| |
| for (i = 0; i < iter; i++) { |
| get_residual(residual, p->prev_excitation, pitch_lag + i - 1); |
| |
| for (j = 0; j < SUBFRAME_LEN; j++) { |
| temp = 0; |
| for (k = 0; k <= j; k++) |
| temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k]; |
| flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) + |
| (1 << 15)) >> 16; |
| } |
| |
| for (j = PITCH_ORDER - 2; j >= 0; j--) { |
| flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15; |
| for (k = 1; k < SUBFRAME_LEN; k++) { |
| temp = (flt_buf[j + 1][k - 1] << 15) + |
| residual[j] * impulse_resp[k]; |
| flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16; |
| } |
| } |
| |
| /* Compute crosscorrelation with the signal */ |
| for (j = 0; j < PITCH_ORDER; j++) { |
| temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN); |
| ccr_buf[count++] = av_clipl_int32(temp << 1); |
| } |
| |
| /* Compute energies */ |
| for (j = 0; j < PITCH_ORDER; j++) { |
| ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j], |
| SUBFRAME_LEN); |
| } |
| |
| for (j = 1; j < PITCH_ORDER; j++) { |
| for (k = 0; k < j; k++) { |
| temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN); |
| ccr_buf[count++] = av_clipl_int32(temp<<2); |
| } |
| } |
| } |
| |
| /* Normalize and shorten */ |
| max = 0; |
| for (i = 0; i < 20 * iter; i++) |
| max = FFMAX(max, FFABS(ccr_buf[i])); |
| |
| temp = normalize_bits_int32(max); |
| |
| for (i = 0; i < 20 * iter; i++){ |
| ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) + |
| (1 << 15)) >> 16; |
| } |
| |
| max = 0; |
| for (i = 0; i < iter; i++) { |
| /* Select quantization table */ |
| if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 || |
| odd_frame && pitch_lag >= SUBFRAME_LEN - 2) { |
| cb_tbl = adaptive_cb_gain170; |
| tbl_size = 170; |
| } |
| |
| for (j = 0, k = 0; j < tbl_size; j++, k += 20) { |
| temp = 0; |
| for (l = 0; l < 20; l++) |
| temp += ccr_buf[20 * i + l] * cb_tbl[k + l]; |
| temp = av_clipl_int32(temp); |
| |
| if (temp > max) { |
| max = temp; |
| acb_gain = j; |
| acb_lag = i; |
| } |
| } |
| } |
| |
| if (!odd_frame) { |
| pitch_lag += acb_lag - 1; |
| acb_lag = 1; |
| } |
| |
| p->pitch_lag[index >> 1] = pitch_lag; |
| p->subframe[index].ad_cb_lag = acb_lag; |
| p->subframe[index].ad_cb_gain = acb_gain; |
| } |
| |
| /** |
| * Subtract the adaptive codebook contribution from the input |
| * to obtain the residual. |
| * |
| * @param buf target vector |
| */ |
| static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp, |
| int16_t *buf) |
| { |
| int i, j; |
| /* Subtract adaptive CB contribution to obtain the residual */ |
| for (i = 0; i < SUBFRAME_LEN; i++) { |
| int64_t temp = buf[i] << 14; |
| for (j = 0; j <= i; j++) |
| temp -= residual[j] * impulse_resp[i - j]; |
| |
| buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16; |
| } |
| } |
| |
| /** |
| * Quantize the residual signal using the fixed codebook (MP-MLQ). |
| * |
| * @param optim optimized fixed codebook parameters |
| * @param buf excitation vector |
| */ |
| static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp, |
| int16_t *buf, int pulse_cnt, int pitch_lag) |
| { |
| FCBParam param; |
| int16_t impulse_r[SUBFRAME_LEN]; |
| int16_t temp_corr[SUBFRAME_LEN]; |
| int16_t impulse_corr[SUBFRAME_LEN]; |
| |
| int ccr1[SUBFRAME_LEN]; |
| int ccr2[SUBFRAME_LEN]; |
| int amp, err, max, max_amp_index, min, scale, i, j, k, l; |
| |
| int64_t temp; |
| |
| /* Update impulse response */ |
| memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN); |
| param.dirac_train = 0; |
| if (pitch_lag < SUBFRAME_LEN - 2) { |
| param.dirac_train = 1; |
| gen_dirac_train(impulse_r, pitch_lag); |
| } |
| |
| for (i = 0; i < SUBFRAME_LEN; i++) |
| temp_corr[i] = impulse_r[i] >> 1; |
| |
| /* Compute impulse response autocorrelation */ |
| temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN); |
| |
| scale = normalize_bits_int32(temp); |
| impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16; |
| |
| for (i = 1; i < SUBFRAME_LEN; i++) { |
| temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i); |
| impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16; |
| } |
| |
| /* Compute crosscorrelation of impulse response with residual signal */ |
| scale -= 4; |
| for (i = 0; i < SUBFRAME_LEN; i++){ |
| temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i); |
| if (scale < 0) |
| ccr1[i] = temp >> -scale; |
| else |
| ccr1[i] = av_clipl_int32(temp << scale); |
| } |
| |
| /* Search loop */ |
| for (i = 0; i < GRID_SIZE; i++) { |
| /* Maximize the crosscorrelation */ |
| max = 0; |
| for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) { |
| temp = FFABS(ccr1[j]); |
| if (temp >= max) { |
| max = temp; |
| param.pulse_pos[0] = j; |
| } |
| } |
| |
| /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */ |
| amp = max; |
| min = 1 << 30; |
| max_amp_index = GAIN_LEVELS - 2; |
| for (j = max_amp_index; j >= 2; j--) { |
| temp = av_clipl_int32((int64_t)fixed_cb_gain[j] * |
| impulse_corr[0] << 1); |
| temp = FFABS(temp - amp); |
| if (temp < min) { |
| min = temp; |
| max_amp_index = j; |
| } |
| } |
| |
| max_amp_index--; |
| /* Select additional gain values */ |
| for (j = 1; j < 5; j++) { |
| for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) { |
| temp_corr[k] = 0; |
| ccr2[k] = ccr1[k]; |
| } |
| param.amp_index = max_amp_index + j - 2; |
| amp = fixed_cb_gain[param.amp_index]; |
| |
| param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp; |
| temp_corr[param.pulse_pos[0]] = 1; |
| |
| for (k = 1; k < pulse_cnt; k++) { |
| max = -1 << 30; |
| for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) { |
| if (temp_corr[l]) |
| continue; |
| temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])]; |
| temp = av_clipl_int32((int64_t)temp * |
| param.pulse_sign[k - 1] << 1); |
| ccr2[l] -= temp; |
| temp = FFABS(ccr2[l]); |
| if (temp > max) { |
| max = temp; |
| param.pulse_pos[k] = l; |
| } |
| } |
| |
| param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ? |
| -amp : amp; |
| temp_corr[param.pulse_pos[k]] = 1; |
| } |
| |
| /* Create the error vector */ |
| memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN); |
| |
| for (k = 0; k < pulse_cnt; k++) |
| temp_corr[param.pulse_pos[k]] = param.pulse_sign[k]; |
| |
| for (k = SUBFRAME_LEN - 1; k >= 0; k--) { |
| temp = 0; |
| for (l = 0; l <= k; l++) { |
| int prod = av_clipl_int32((int64_t)temp_corr[l] * |
| impulse_r[k - l] << 1); |
| temp = av_clipl_int32(temp + prod); |
| } |
| temp_corr[k] = temp << 2 >> 16; |
| } |
| |
| /* Compute square of error */ |
| err = 0; |
| for (k = 0; k < SUBFRAME_LEN; k++) { |
| int64_t prod; |
| prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1); |
| err = av_clipl_int32(err - prod); |
| prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]); |
| err = av_clipl_int32(err + prod); |
| } |
| |
| /* Minimize */ |
| if (err < optim->min_err) { |
| optim->min_err = err; |
| optim->grid_index = i; |
| optim->amp_index = param.amp_index; |
| optim->dirac_train = param.dirac_train; |
| |
| for (k = 0; k < pulse_cnt; k++) { |
| optim->pulse_sign[k] = param.pulse_sign[k]; |
| optim->pulse_pos[k] = param.pulse_pos[k]; |
| } |
| } |
| } |
| } |
| } |
| |
| /** |
| * Encode the pulse position and gain of the current subframe. |
| * |
| * @param optim optimized fixed CB parameters |
| * @param buf excitation vector |
| */ |
| static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim, |
| int16_t *buf, int pulse_cnt) |
| { |
| int i, j; |
| |
| j = PULSE_MAX - pulse_cnt; |
| |
| subfrm->pulse_sign = 0; |
| subfrm->pulse_pos = 0; |
| |
| for (i = 0; i < SUBFRAME_LEN >> 1; i++) { |
| int val = buf[optim->grid_index + (i << 1)]; |
| if (!val) { |
| subfrm->pulse_pos += combinatorial_table[j][i]; |
| } else { |
| subfrm->pulse_sign <<= 1; |
| if (val < 0) subfrm->pulse_sign++; |
| j++; |
| |
| if (j == PULSE_MAX) break; |
| } |
| } |
| subfrm->amp_index = optim->amp_index; |
| subfrm->grid_index = optim->grid_index; |
| subfrm->dirac_train = optim->dirac_train; |
| } |
| |
| /** |
| * Compute the fixed codebook excitation. |
| * |
| * @param buf target vector |
| * @param impulse_resp impulse response of the combined filter |
| */ |
| static void fcb_search(G723_1_Context *p, int16_t *impulse_resp, |
| int16_t *buf, int index) |
| { |
| FCBParam optim; |
| int pulse_cnt = pulses[index]; |
| int i; |
| |
| optim.min_err = 1 << 30; |
| get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN); |
| |
| if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) { |
| get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, |
| p->pitch_lag[index >> 1]); |
| } |
| |
| /* Reconstruct the excitation */ |
| memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN); |
| for (i = 0; i < pulse_cnt; i++) |
| buf[optim.pulse_pos[i]] = optim.pulse_sign[i]; |
| |
| pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt); |
| |
| if (optim.dirac_train) |
| gen_dirac_train(buf, p->pitch_lag[index >> 1]); |
| } |
| |
| /** |
| * Pack the frame parameters into output bitstream. |
| * |
| * @param frame output buffer |
| * @param size size of the buffer |
| */ |
| static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size) |
| { |
| PutBitContext pb; |
| int info_bits, i, temp; |
| |
| init_put_bits(&pb, frame, size); |
| |
| if (p->cur_rate == RATE_6300) { |
| info_bits = 0; |
| put_bits(&pb, 2, info_bits); |
| }else |
| av_assert0(0); |
| |
| put_bits(&pb, 8, p->lsp_index[2]); |
| put_bits(&pb, 8, p->lsp_index[1]); |
| put_bits(&pb, 8, p->lsp_index[0]); |
| |
| put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN); |
| put_bits(&pb, 2, p->subframe[1].ad_cb_lag); |
| put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN); |
| put_bits(&pb, 2, p->subframe[3].ad_cb_lag); |
| |
| /* Write 12 bit combined gain */ |
| for (i = 0; i < SUBFRAMES; i++) { |
| temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS + |
| p->subframe[i].amp_index; |
| if (p->cur_rate == RATE_6300) |
| temp += p->subframe[i].dirac_train << 11; |
| put_bits(&pb, 12, temp); |
| } |
| |
| put_bits(&pb, 1, p->subframe[0].grid_index); |
| put_bits(&pb, 1, p->subframe[1].grid_index); |
| put_bits(&pb, 1, p->subframe[2].grid_index); |
| put_bits(&pb, 1, p->subframe[3].grid_index); |
| |
| if (p->cur_rate == RATE_6300) { |
| skip_put_bits(&pb, 1); /* reserved bit */ |
| |
| /* Write 13 bit combined position index */ |
| temp = (p->subframe[0].pulse_pos >> 16) * 810 + |
| (p->subframe[1].pulse_pos >> 14) * 90 + |
| (p->subframe[2].pulse_pos >> 16) * 9 + |
| (p->subframe[3].pulse_pos >> 14); |
| put_bits(&pb, 13, temp); |
| |
| put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff); |
| put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff); |
| put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff); |
| put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff); |
| |
| put_bits(&pb, 6, p->subframe[0].pulse_sign); |
| put_bits(&pb, 5, p->subframe[1].pulse_sign); |
| put_bits(&pb, 6, p->subframe[2].pulse_sign); |
| put_bits(&pb, 5, p->subframe[3].pulse_sign); |
| } |
| |
| flush_put_bits(&pb); |
| return frame_size[info_bits]; |
| } |
| |
| static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, |
| const AVFrame *frame, int *got_packet_ptr) |
| { |
| G723_1_Context *p = avctx->priv_data; |
| int16_t unq_lpc[LPC_ORDER * SUBFRAMES]; |
| int16_t qnt_lpc[LPC_ORDER * SUBFRAMES]; |
| int16_t cur_lsp[LPC_ORDER]; |
| int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1]; |
| int16_t vector[FRAME_LEN + PITCH_MAX]; |
| int offset, ret; |
| int16_t *in_orig = av_memdup(frame->data[0], frame->nb_samples * sizeof(int16_t)); |
| int16_t *in = in_orig; |
| |
| HFParam hf[4]; |
| int i, j; |
| |
| if (!in) |
| return AVERROR(ENOMEM); |
| |
| highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem); |
| |
| memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t)); |
| memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t)); |
| |
| comp_lpc_coeff(vector, unq_lpc); |
| lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp); |
| lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp); |
| |
| /* Update memory */ |
| memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN, |
| sizeof(int16_t) * SUBFRAME_LEN); |
| memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in, |
| sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN)); |
| memcpy(p->prev_data, in + HALF_FRAME_LEN, |
| sizeof(int16_t) * HALF_FRAME_LEN); |
| memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN); |
| |
| perceptual_filter(p, weighted_lpc, unq_lpc, vector); |
| |
| memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN); |
| memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX); |
| memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN); |
| |
| scale_vector(vector, vector, FRAME_LEN + PITCH_MAX); |
| |
| p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX); |
| p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN); |
| |
| for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) |
| comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j); |
| |
| memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX); |
| memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN); |
| memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX); |
| |
| for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) |
| harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i); |
| |
| inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0); |
| lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp); |
| |
| memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER); |
| |
| offset = 0; |
| for (i = 0; i < SUBFRAMES; i++) { |
| int16_t impulse_resp[SUBFRAME_LEN]; |
| int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1]; |
| int16_t flt_in[SUBFRAME_LEN]; |
| int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER]; |
| |
| /** |
| * Compute the combined impulse response of the synthesis filter, |
| * formant perceptual weighting filter and harmonic noise shaping filter |
| */ |
| memset(zero, 0, sizeof(int16_t) * LPC_ORDER); |
| memset(vector, 0, sizeof(int16_t) * PITCH_MAX); |
| memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN); |
| |
| flt_in[0] = 1 << 13; /* Unit impulse */ |
| synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1), |
| zero, zero, flt_in, vector + PITCH_MAX, 1); |
| harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp); |
| |
| /* Compute the combined zero input response */ |
| flt_in[0] = 0; |
| memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER); |
| memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER); |
| |
| synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1), |
| fir, iir, flt_in, vector + PITCH_MAX, 0); |
| memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX); |
| harmonic_noise_sub(hf + i, vector + PITCH_MAX, in); |
| |
| acb_search(p, residual, impulse_resp, in, i); |
| gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1], |
| &p->subframe[i], p->cur_rate); |
| sub_acb_contrib(residual, impulse_resp, in); |
| |
| fcb_search(p, impulse_resp, in, i); |
| |
| /* Reconstruct the excitation */ |
| gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1], |
| &p->subframe[i], RATE_6300); |
| |
| memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN, |
| sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN)); |
| for (j = 0; j < SUBFRAME_LEN; j++) |
| in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]); |
| memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in, |
| sizeof(int16_t) * SUBFRAME_LEN); |
| |
| /* Update filter memories */ |
| synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1), |
| p->perf_fir_mem, p->perf_iir_mem, |
| in, vector + PITCH_MAX, 0); |
| memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN, |
| sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN)); |
| memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX, |
| sizeof(int16_t) * SUBFRAME_LEN); |
| |
| in += SUBFRAME_LEN; |
| offset += LPC_ORDER; |
| } |
| |
| av_freep(&in_orig); in = NULL; |
| |
| if ((ret = ff_alloc_packet2(avctx, avpkt, 24)) < 0) |
| return ret; |
| |
| *got_packet_ptr = 1; |
| avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size); |
| return 0; |
| } |
| |
| AVCodec ff_g723_1_encoder = { |
| .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_encode_init, |
| .encode2 = g723_1_encode_frame, |
| .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16, |
| AV_SAMPLE_FMT_NONE}, |
| }; |
| #endif |