--- /dev/null
- * This file is part of Libav.
+ /*
+ * G.723.1 compatible decoder
+ * Copyright (c) 2006 Benjamin Larsson
+ * Copyright (c) 2010 Mohamed Naufal Basheer
+ *
- * Libav is free software; you can redistribute it and/or
++ * This file is part of FFmpeg.
+ *
- * Libav is distributed in the hope that it will be useful,
++ * 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.
+ *
- * License along with Libav; if not, write to the Free Software
++ * 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
- max = FFMIN(max, 0x7FFF);
- bits = ff_g723_1_normalize_bits(max, 15);
++ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+ #include <stdint.h>
+
+ #include "libavutil/common.h"
+
+ #include "acelp_vectors.h"
+ #include "avcodec.h"
+ #include "celp_math.h"
+ #include "g723_1.h"
+
+ int ff_g723_1_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]);
+
- int i, sum = 0;
-
- for (i = 0; i < length; i++) {
- int prod = a[i] * b[i];
- sum = av_sat_dadd32(sum, prod);
- }
- return sum;
++ 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;
+ }
+
+ int ff_g723_1_normalize_bits(int num, int width)
+ {
+ return width - av_log2(num) - 1;
+ }
+
+ int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length)
+ {
- if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2)
++ int sum = ff_dot_product(a,b,length);
++ return av_sat_add32(sum, sum);
+ }
+
+ void ff_g723_1_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];
+ }
+
+ void ff_g723_1_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];
+ }
+ }
+
+ void ff_g723_1_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;
+
+ ff_g723_1_get_residual(residual, prev_excitation, lag);
+
+ /* Select quantization table */
- else
++ if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {
+ cb_ptr = adaptive_cb_gain85;
- sum = ff_g723_1_dot_product(residual + i, cb_ptr, PITCH_ORDER);
- vector[i] = av_sat_dadd32(1 << 15, sum) >> 16;
++ } else
+ cb_ptr = adaptive_cb_gain170;
+
+ /* Calculate adaptive vector */
+ cb_ptr += subfrm->ad_cb_gain * 20;
+ for (i = 0; i < SUBFRAME_LEN; i++) {
- 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;
++ sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
++ vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
+ }
+ }
+
+ /**
+ * 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++) {
- f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1;
- f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
++ 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;
- lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) +
- (1 << 15)) >> 16;
++ 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];
+
- cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);
++ 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;
+ }
+ }
+
+ void ff_g723_1_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;
+ }
+ }
+
+ void ff_g723_1_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 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, 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];
- }
- }
- ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15))
++#define normalize_bits_int16(num) ff_g723_1_normalize_bits(num, 15)
++#define normalize_bits_int32(num) ff_g723_1_normalize_bits(num, 31)
/**
* Generate fixed codebook excitation vector.
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);
- iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i);
++ iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);
lpc += LPC_ORDER;
}
.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
.priv_class = &g723_1dec_class,
};
- scale_vector(vector, buf, LPC_FRAME);
+
+#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];
+
- error = dot_product(lsp + (offset), temp, size) << 1;\
- error -= dot_product(lsp_band##num[i], temp, size);\
++ ff_g723_1_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;\
+ }\
- get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
++ error = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1;\
++ error -= ff_g723_1_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++) {
- ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
++ ff_g723_1_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++) {
- gen_dirac_train(impulse_r, pitch_lag);
++ ccr_buf[count++] = ff_g723_1_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;
- temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
++ ff_g723_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 + i, temp_corr, SUBFRAME_LEN - i);
++ temp = ff_g723_1_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(buf + i, impulse_r, SUBFRAME_LEN - i);
++ temp = ff_g723_1_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++){
- gen_dirac_train(buf, p->pitch_lag[index >> 1]);
++ temp = ff_g723_1_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 = INT_MIN;
+ 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)
- scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
++ ff_g723_1_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);
+
- inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
- lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
++ ff_g723_1_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);
+
- gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
++ ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
++ ff_g723_1_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(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
++ ff_g723_1_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 */
++ ff_g723_1_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)) < 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
projects / ffmpeg / commitdiff