diff options
Diffstat (limited to 'libavcodec/g723_1.c')
| -rw-r--r-- | libavcodec/g723_1.c | 1296 |
1 files changed, 1199 insertions, 97 deletions
diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c index 377a718ffb..d2cde8a864 100644 --- a/libavcodec/g723_1.c +++ b/libavcodec/g723_1.c @@ -3,20 +3,20 @@ * Copyright (c) 2006 Benjamin Larsson * Copyright (c) 2010 Mohamed Naufal Basheer * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -30,50 +30,16 @@ #include "libavutil/mem.h" #include "libavutil/opt.h" #include "avcodec.h" +#include "internal.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 -/** - * 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 { - 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 { - 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; @@ -100,10 +66,21 @@ typedef struct g723_1_context { int sid_gain; int cur_gain; int reflection_coef; - int pf_gain; + 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) @@ -113,7 +90,6 @@ static av_cold int g723_1_decode_init(AVCodecContext *avctx) 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)); @@ -197,13 +173,13 @@ static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, } } - 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); + 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_bits(&gb, 1); /* skip reserved bit */ + skip_bits1(&gb); /* skip reserved bit */ /* Compute pulse_pos index using the 13-bit combined position index */ temp = get_bits(&gb, 13); @@ -247,32 +223,27 @@ static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, /** * Bitexact implementation of sqrt(val/2). */ -static int16_t square_root(int val) +static int16_t square_root(unsigned val) { - int16_t res = 0; - int16_t exp = 0x4000; - int i; + av_assert2(!(val & 0x80000000)); - for (i = 0; i < 14; i ++) { - int res_exp = res + exp; - if (val >= res_exp * res_exp << 1) - res += exp; - exp >>= 1; - } - return res; + 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, 16 bits(0) / 32 bits(1) + * @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. */ @@ -281,12 +252,11 @@ 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); + bits= 14 - av_log2_16bit(max); + bits= FFMAX(bits, 0); for (i = 0; i < length; i++) dst[i] = vector[i] << bits >> 3; @@ -371,7 +341,7 @@ static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, * @param b 16 bit multiplier */ #define MULL2(a, b) \ - ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) + MULL(a,b,15) /** * Convert LSP frequencies to LPC coefficients. @@ -386,7 +356,7 @@ static void lsp2lpc(int16_t *lpc) /* Calculate negative cosine */ for (j = 0; j < LPC_ORDER; j++) { - int index = lpc[j] >> 7; + 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]) * @@ -567,13 +537,8 @@ static void get_residual(int16_t *residual, int16_t *prev_excitation, int 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; + int sum = ff_dot_product(a,b,length); + return av_sat_add32(sum, sum); } /** @@ -593,16 +558,16 @@ static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, get_residual(residual, prev_excitation, lag); /* Select quantization table */ - if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) + if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) { cb_ptr = adaptive_cb_gain85; - else + } 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; + sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER); + vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16; } } @@ -810,9 +775,9 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag, temp = best_eng * *exc_eng >> 3; - if (temp < ccr * ccr) + if (temp < ccr * ccr) { return index; - else + } else return 0; } @@ -852,21 +817,24 @@ static void residual_interp(int16_t *buf, int16_t *out, int lag, * @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) */ -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)); - } +#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;\ + }\ } /** @@ -937,13 +905,12 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, (1 << 14)) >> 15; } iir_filter(filter_coef[0], filter_coef[1], buf + i, - filter_signal + i); + filter_signal + i, 1); 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)); + 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; @@ -1219,8 +1186,8 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data, frame->nb_samples = FRAME_LEN; if ((ret = ff_get_buffer(avctx, frame)) < 0) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - return ret; + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } out = (int16_t *)frame->data[0]; @@ -1376,3 +1343,1138 @@ AVCodec ff_g723_1_decoder = { .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); + } + + 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 = (const int16_t *)frame->data[0]; + + HFParam hf[4]; + int i, j; + + 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; + } + + 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", + .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, + .long_name = NULL_IF_CONFIG_SMALL("G.723.1"), + .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE}, +}; +#endif |