@@ -159,6 +159,8 @@ OBJS-$(CONFIG_FLIC_DECODER)            += flicvideo.o

 OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o

 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o

 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o

+OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \

+                                          celp_filters.o celp_math.o

 OBJS-$(CONFIG_G729_DECODER)            += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o

 OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o

 OBJS-$(CONFIG_GIF_ENCODER)             += gif.o lzwenc.o

@@ -258,6 +258,7 @@ void avcodec_register_all(void)

     REGISTER_DECODER (DSICINAUDIO, dsicinaudio);

     REGISTER_ENCDEC  (EAC3, eac3);

     REGISTER_ENCDEC  (FLAC, flac);

+    REGISTER_DECODER (G723_1, g723_1);

     REGISTER_DECODER (G729, g729);

     REGISTER_DECODER (GSM, gsm);

     REGISTER_DECODER (GSM_MS, gsm_ms);

@@ -345,6 +345,7 @@ enum CodecID {

     CODEC_ID_QDMC,

     CODEC_ID_CELT,

     CODEC_ID_G729 = 0x15800,

+    CODEC_ID_G723_1= 0x15801,

     /* subtitle codecs */

     CODEC_ID_FIRST_SUBTITLE = 0x17000,          ///< A dummy ID pointing at the start of subtitle codecs.

new file mode 100644

@@ -0,0 +1,1081 @@

+/*

+ * 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

+ */

+

+#include "avcodec.h"

+#define ALT_BITSTREAM_READER_LE

+#include "get_bits.h"

+#include "acelp_vectors.h"

+#include "celp_filters.h"

+#include "celp_math.h"

+#include "lsp.h"

+#include "libavutil/lzo.h"

+#include "g723_1_data.h"

+

+typedef struct g723_1_context {

+    G723_1_Subframe subframe[4];

+    FrameType cur_frame_type;

+    FrameType past_frame_type;

+    Rate cur_rate;

+    uint8_t lsp_index[LSP_BANDS];

+    int pitch_lag[2];

+    int erased_frames;

+

+    int16_t prev_lsp[LPC_ORDER];

+    int16_t prev_excitation[PITCH_MAX];

+    int16_t excitation[PITCH_MAX + FRAME_LEN];

+    int16_t synth_mem[LPC_ORDER];

+    int16_t fir_mem[LPC_ORDER];

+    int     iir_mem[LPC_ORDER];

+

+    int 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

+} G723_1_Context;

+

+static av_cold int g723_1_decode_init(AVCodecContext *avctx)

+{

+    G723_1_Context *p  = avctx->priv_data;

+

+    avctx->sample_fmt  = SAMPLE_FMT_S16;

+    p->pf_gain         = 1 << 12;

+    memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));

+

+    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 = UntransmittedFrame;

+        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 = SIDFrame;

+        p->subframe[0].amp_index = get_bits(&gb, 6);

+        return 0;

+    }

+

+    /* Extract the info common to both rates */

+    p->cur_rate       = info_bits ? Rate5k3 : Rate6k3;

+    p->cur_frame_type = ActiveFrame;

+

+    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 == Rate6k3 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {

+            p->subframe[i].dirac_train = temp >> 11;

+            temp &= 0x7ff;

+            ad_cb_len = 85;

+        }

+        p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);

+        if (p->subframe[i].ad_cb_gain < ad_cb_len) {

+            p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *

+                                       GAIN_LEVELS;

+        } else {

+            return -1;

+        }

+    }

+

+    p->subframe[0].grid_index = get_bits(&gb, 1);

+    p->subframe[1].grid_index = get_bits(&gb, 1);

+    p->subframe[2].grid_index = get_bits(&gb, 1);

+    p->subframe[3].grid_index = get_bits(&gb, 1);

+

+    if (p->cur_rate == Rate6k3) {

+        skip_bits(&gb, 1);  /* skip reserved bit */

+

+        /* Compute pulse_pos index using the 13-bit combined position index */

+        temp = get_bits(&gb, 13);

+        p->subframe[0].pulse_pos = temp / 810;

+

+        temp -= p->subframe[0].pulse_pos * 810;

+        p->subframe[1].pulse_pos = FASTDIV(temp, 90);

+

+        temp -= p->subframe[1].pulse_pos * 90;

+        p->subframe[2].pulse_pos = FASTDIV(temp, 9);

+        p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;

+

+        p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +

+                                   get_bits(&gb, 16);

+        p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +

+                                   get_bits(&gb, 14);

+        p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +

+                                   get_bits(&gb, 16);

+        p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +

+                                   get_bits(&gb, 14);

+

+        p->subframe[0].pulse_sign = get_bits(&gb, 6);

+        p->subframe[1].pulse_sign = get_bits(&gb, 5);

+        p->subframe[2].pulse_sign = get_bits(&gb, 6);

+        p->subframe[3].pulse_sign = get_bits(&gb, 5);

+    } else { /* Rate5k3 */

+        p->subframe[0].pulse_pos  = get_bits(&gb, 12);

+        p->subframe[1].pulse_pos  = get_bits(&gb, 12);

+        p->subframe[2].pulse_pos  = get_bits(&gb, 12);

+        p->subframe[3].pulse_pos  = get_bits(&gb, 12);

+

+        p->subframe[0].pulse_sign = get_bits(&gb, 4);

+        p->subframe[1].pulse_sign = get_bits(&gb, 4);

+        p->subframe[2].pulse_sign = get_bits(&gb, 4);

+        p->subframe[3].pulse_sign = get_bits(&gb, 4);

+    }

+

+    return 0;

+}

+

+/**

+ * Bitexact implementation of sqrt(val/2).

+ */

+static int16_t square_root(int val)

+{

+    int16_t res = 0;

+    int16_t exp = 0x4000;

+    int i;

+

+    for (i = 0; i < 14; i ++) {

+        int res_exp = res + exp;

+        if (val >= res_exp * res_exp << 1)

+            res += exp;

+        exp >>= 1;

+    }

+    return res;

+}

+

+/**

+ * Calculate the number of left-shifts required for normalizing the input.

+ *

+ * @param num   input number

+ * @param width width of the input, 16 bits(0) / 32 bits(1)

+ */

+static int normalize_bits(int num, int width)

+{

+    int i = 0;

+    int bits = (width) ? 31 : 15;

+    int limit = 1 << (bits - 1);

+

+    if (num) {

+        if (num == -1)

+            return bits;

+        if (num < 0)

+            num = ~num;

+        for (i = 0; num < limit; i++)

+            num <<= 1;

+    }

+    return i;

+}

+

+/**

+ * Scale vector contents based on the largest of their absolutes.

+ */

+static int scale_vector(int16_t *vector, int length)

+{

+    int bits, scale, max = 0;

+    int i;

+

+    const int16_t shift_table[16] = {

+        0x0001, 0x0002, 0x0004, 0x0008, 0x0010, 0x0020, 0x0040, 0x0080,

+        0x0100, 0x0200, 0x0400, 0x0800, 0x1000, 0x2000, 0x4000, 0x7fff

+    };

+

+    for (i = 0; i < length; i++)

+        max = FFMAX(max, FFABS(vector[i]));

+

+    bits  = normalize_bits(max, 0);

+    scale = shift_table[bits];

+

+    for (i = 0; i < length; i++)

+        vector[i] = (int16_t)(av_clipl_int32(vector[i] * scale << 1) >> 4);

+

+    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(int16_t));

+}

+

+/**

+ * Bitexact implementation of 2ab scaled by 1/2^16.

+ *

+ * @param a 32 bit multiplicand

+ * @param b 16 bit multiplier

+ */

+#define MULL2(a, b) \

+        ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15))

+

+/**

+ * Convert LSP frequencies to LPC coefficients.

+ *

+ * @param lpc buffer for LPC coefficients

+ */

+static void lsp2lpc(int16_t *lpc)

+{

+    int f1[LPC_ORDER / 2 + 1];

+    int f2[LPC_ORDER / 2 + 1];

+    int i, j;

+

+    /* Calculate negative cosine */

+    for (j = 0; j < LPC_ORDER; j++) {

+        int index     = lpc[j] >> 7;

+        int offset    = lpc[j] & 0x7f;

+        int64_t temp1 = cos_tab[index] << 16;

+        int temp2     = (cos_tab[index + 1] - cos_tab[index]) *

+                          ((offset << 8) + 0x80) << 1;

+

+        lpc[j] = -(av_clipl_int32(((temp1 + temp2) << 1) + (1 << 15)) >> 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(int16_t));

+

+    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(int16_t));

+    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,

+                               Rate cur_rate, int pitch_lag, int index)

+{

+    int temp, i, j;

+

+    memset(vector, 0, SUBFRAME_LEN * sizeof(int16_t));

+

+    if (cur_rate == Rate6k3) {

+        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 { /* Rate5k3 */

+        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];

+}

+

+/**

+ * Generate adaptive codebook excitation.

+ */

+static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,

+                               int pitch_lag, G723_1_Subframe subfrm,

+                               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;

+    int64_t sum;

+

+    get_residual(residual, prev_excitation, lag);

+

+    /* Select quantization table */

+    if (cur_rate == Rate6k3 && 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, 1);

+        vector[i] = av_clipl_int32((sum << 1) + (1 << 15)) >> 16;

+    }

+}

+

+/**

+ * Estimate maximum auto-correlation around pitch lag.

+ *

+ * @param p         the context

+ * @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(G723_1_Context *p, int offset, int *ccr_max,

+                        int pitch_lag, int length, int dir)

+{

+    int limit, ccr, lag = 0;

+    int16_t *buf = p->excitation + offset;

+    int i;

+

+    pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);

+    limit     = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);

+

+    for (i = pitch_lag - 3; i <= limit; i++) {

+        ccr = ff_dot_product(buf, buf + dir * i, length, 1);

+

+        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, Rate cur_rate,

+                           int tgt_eng, int ccr, int res_eng)

+{

+    int pf_residual;     /* square of postfiltered residual */

+    int64_t 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_clipl_int32(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, Rate cur_rate)

+{

+

+    int16_t scale;

+    int i;

+    int64_t 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->excitation + offset;

+    int fwd_lag   = autocorr_max(p, offset, &energy[1], pitch_lag,

+                                 SUBFRAME_LEN, 1);

+    int back_lag  = autocorr_max(p, 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] = ff_dot_product(buf, buf, SUBFRAME_LEN, 1);

+

+    /* Compute forward residual energy */

+    if (fwd_lag)

+        energy[2] = ff_dot_product(buf + fwd_lag, buf + fwd_lag,

+                                   SUBFRAME_LEN, 1);

+

+    /* Compute backward residual energy */

+    if (back_lag)

+        energy[4] = ff_dot_product(buf - back_lag, buf - back_lag,

+                                   SUBFRAME_LEN, 1);

+

+    /* Normalize and shorten */

+    temp1 = 0;

+    for (i = 0; i < 5; i++)

+        temp1 = FFMAX(energy[i], temp1);

+

+    scale = normalize_bits(temp1, 1);

+    for (i = 0; i < 5; i++)

+        energy[i] = av_clipl_int32(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->excitation + offset;

+

+    int index, ccr, tgt_eng, best_eng, temp;

+

+    *scale = scale_vector(p->excitation, FRAME_LEN + PITCH_MAX);

+

+    /* Compute maximum backward cross-correlation */

+    ccr   = 0;

+    index = autocorr_max(p, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);

+    ccr   = av_clipl_int32((int64_t)ccr + (1 << 15)) >> 16;

+

+    /* Compute target energy */

+    tgt_eng  = ff_dot_product(buf, buf, SUBFRAME_LEN * 2, 1);

+    *exc_eng = av_clipl_int32(tgt_eng + (1 << 15)) >> 16;

+

+    if (ccr <= 0)

+        return 0;

+

+    /* Compute best energy */

+    best_eng = ff_dot_product(buf - index, buf - index,

+                              SUBFRAME_LEN * 2, 1);

+    best_eng = av_clipl_int32((int64_t)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++)

+            vector_ptr[i - lag] = vector_ptr[i - lag] * 3 >> 2;

+        av_memcpy_backptr((uint8_t*)vector_ptr, lag * sizeof(int16_t),

+                          FRAME_LEN * sizeof(int16_t));

+        memcpy(out, vector_ptr, FRAME_LEN * sizeof(int16_t));

+    } 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(int16_t));

+    }

+}

+

+/**

+ * 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++) {

+        int64_t temp = buf[i] >> 2;

+        temp  = av_clipl_int32(MUL64(temp, temp) << 1);

+        denom = av_clipl_int32(denom + temp);

+    }

+

+    if (num && denom) {

+        bits1   = normalize_bits(num, 1);

+        bits2   = normalize_bits(denom, 1);