lavc: G.723.1 encoder

Additional improvements by Michael Niedermayer <michaelni@gmx.at>.

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>

10 files changed, 1316 insertions, 11 deletions

diff --git a/Changelog b/Changelog
index d37d5d9189..a45bb9eccc 100644
--- a/Changelog
+++ b/Changelog
@@ -49,6 +49,7 @@ version <next>:
 - innoHeim/Rsupport Screen Capture Codec decoder
 - support encoding 16-bit RLE SGI images
 - support Apple AVFoundation video capture
+- G.723.1 encoder
 
 
 version 11:
diff --git a/doc/general.texi b/doc/general.texi
index bddc075488..15e4a660d6 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -874,7 +874,7 @@ following image formats are supported:
 @item DV audio               @tab     @tab  X
 @item Enhanced AC-3          @tab  X  @tab  X
 @item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
-@item G.723.1                @tab     @tab  X
+@item G.723.1                @tab  X  @tab  X
 @item GSM                    @tab  E  @tab  X
     @tab encoding supported through external library libgsm
 @item GSM Microsoft variant  @tab  E  @tab  X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 85738fa1f0..ee76315886 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -227,6 +227,8 @@ OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
 OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o elsdec.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1dec.o g723_1.o \
                                           acelp_vectors.o celp_filters.o
+OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1enc.o g723_1.o \
+                                          acelp_vectors.o celp_filters.o
 OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o
 OBJS-$(CONFIG_GIF_ENCODER)             += gif.o lzwenc.o
 OBJS-$(CONFIG_GSM_DECODER)             += gsmdec.o gsmdec_data.o msgsmdec.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 0ad102cf26..6eec4fe1e1 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -330,7 +330,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(DSS_SP,            dss_sp);
     REGISTER_ENCDEC (EAC3,              eac3);
     REGISTER_ENCDEC (FLAC,              flac);
-    REGISTER_DECODER(G723_1,            g723_1);
+    REGISTER_ENCDEC (G723_1,            g723_1);
     REGISTER_DECODER(GSM,               gsm);
     REGISTER_DECODER(GSM_MS,            gsm_ms);
     REGISTER_DECODER(IAC,               iac);
diff --git a/libavcodec/celp_math.c b/libavcodec/celp_math.c
index a9ebef697c..8a788f58ec 100644
--- a/libavcodec/celp_math.c
+++ b/libavcodec/celp_math.c
@@ -26,6 +26,8 @@
 
 #include "avcodec.h"
 #include "celp_math.h"
+#include "mathops.h"
+
 #include "libavutil/common.h"
 
 static const uint16_t exp2a[]=
@@ -86,3 +88,14 @@ int ff_log2_q15(uint32_t value)
 
     return (power_int << 15) + value;
 }
+
+int64_t ff_dot_product(const int16_t *a, const int16_t *b, int length)
+{
+    int i;
+    int64_t sum = 0;
+
+    for (i = 0; i < length; i++)
+        sum += MUL16(a[i], b[i]);
+
+    return sum;
+}
diff --git a/libavcodec/celp_math.h b/libavcodec/celp_math.h
index ed3f8c0fe2..9cebdfe695 100644
--- a/libavcodec/celp_math.h
+++ b/libavcodec/celp_math.h
@@ -43,6 +43,16 @@ int ff_exp2(uint16_t power);
 int ff_log2_q15(uint32_t value);
 
 /**
+ * Calculate the dot product of 2 int16_t vectors.
+ * @param a input data array
+ * @param b input data array
+ * @param length number of elements
+ *
+ * @return dot product = sum of elementwise products
+ */
+int64_t ff_dot_product(const int16_t *a, const int16_t *b, int length);
+
+/**
  * Shift value left or right depending on sign of offset parameter.
  * @param value value to shift
  * @param offset shift offset
diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c
index af4777cc35..3d45f9d1cf 100644
--- a/libavcodec/g723_1.c
+++ b/libavcodec/g723_1.c
@@ -53,13 +53,8 @@ int ff_g723_1_normalize_bits(int num, int width)
 
 int ff_g723_1_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);
 }
 
 void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation,
diff --git a/libavcodec/g723_1.h b/libavcodec/g723_1.h
index 391ca464a9..3fd52dfdee 100644
--- a/libavcodec/g723_1.h
+++ b/libavcodec/g723_1.h
@@ -33,6 +33,8 @@
 #define SUBFRAMES       4
 #define SUBFRAME_LEN    60
 #define FRAME_LEN       (SUBFRAME_LEN << 2)
+#define HALF_FRAME_LEN  (FRAME_LEN / 2)
+#define LPC_FRAME       (HALF_FRAME_LEN + SUBFRAME_LEN)
 #define LPC_ORDER       10
 #define LSP_BANDS       3
 #define LSP_CB_SIZE     256
@@ -92,6 +94,26 @@ typedef struct PPFParam {
     int16_t sc_gain;  ///< scaling gain
 } PPFParam;
 
+/**
+ * Harmonic filter parameters
+ */
+typedef struct HFParam {
+    int index;
+    int gain;
+} HFParam;
+
+/**
+ * Optimized fixed codebook excitation parameters
+ */
+typedef struct FCBParam {
+    int min_err;
+    int amp_index;
+    int grid_index;
+    int dirac_train;
+    int pulse_pos[PULSE_MAX];
+    int pulse_sign[PULSE_MAX];
+} FCBParam;
+
 typedef struct g723_1_context {
     AVClass *class;
 
@@ -122,6 +144,17 @@ typedef struct g723_1_context {
     int postfilter;
 
     int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
+
+    /* encoder */
+    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;
 
 
@@ -1329,6 +1362,55 @@ static const int16_t postfilter_tbl[2][LPC_ORDER] = {
     { 24576, 18432, 13824, 10368, 7776, 5832, 4374, 3281, 2460, 1845 }
 };
 
+
+/**
+ * Hamming window coefficients scaled by 2^15
+ */
+static const int16_t hamming_window[LPC_FRAME] = {
+     2621,  2631,  2659,  2705,  2770,  2853,  2955,  3074,  3212,  3367,
+     3541,  3731,  3939,  4164,  4405,  4663,  4937,  5226,  5531,  5851,
+     6186,  6534,  6897,  7273,  7661,  8062,  8475,  8899,  9334,  9780,
+    10235, 10699, 11172, 11653, 12141, 12636, 13138, 13645, 14157, 14673,
+    15193, 15716, 16242, 16769, 17298, 17827, 18356, 18884, 19411, 19935,
+    20457, 20975, 21489, 21999, 22503, 23002, 23494, 23978, 24455, 24924,
+    25384, 25834, 26274, 26704, 27122, 27529, 27924, 28306, 28675, 29031,
+    29373, 29700, 30012, 30310, 30592, 30857, 31107, 31340, 31557, 31756,
+    31938, 32102, 32249, 32377, 32488, 32580, 32654, 32710, 32747, 32766,
+    32766, 32747, 32710, 32654, 32580, 32488, 32377, 32249, 32102, 31938,
+    31756, 31557, 31340, 31107, 30857, 30592, 30310, 30012, 29700, 29373,
+    29031, 28675, 28306, 27924, 27529, 27122, 26704, 26274, 25834, 25384,
+    24924, 24455, 23978, 23494, 23002, 22503, 21999, 21489, 20975, 20457,
+    19935, 19411, 18884, 18356, 17827, 17298, 16769, 16242, 15716, 15193,
+    14673, 14157, 13645, 13138, 12636, 12141, 11653, 11172, 10699, 10235,
+     9780, 9334,   8899,  8475,  8062,  7661,  7273,  6897,  6534,  6186,
+     5851, 5531,   5226,  4937,  4663,  4405,  4164,  3939,  3731,  3541,
+     3367, 3212,   3074,  2955,  2853,  2770,  2705,  2659,  2631,  2621
+};
+
+/**
+ * Binomial window coefficients scaled by 2^15
+ */
+static const int16_t binomial_window[LPC_ORDER] = {
+    32749, 32695, 32604, 32477, 32315, 32118, 31887, 31622, 31324, 30995
+};
+
+/**
+ * 0.994^i scaled by 2^15
+ */
+static const int16_t bandwidth_expand[LPC_ORDER] = {
+    32571, 32376, 32182, 31989, 31797, 31606, 31416, 31228, 31040, 30854
+};
+
+/**
+ * 0.5^i scaled by 2^15
+ */
+static const int16_t percept_flt_tbl[2][LPC_ORDER] = {
+    /* Zero part */
+    {29491, 26542, 23888, 21499, 19349, 17414, 15673, 14106, 12695, 11425},
+    /* Pole part */
+    {16384,  8192,  4096,  2048,  1024,   512,   256,   128,    64,    32}
+};
+
 static const int cng_adaptive_cb_lag[4] = { 1, 0, 1, 3 };
 
 static const int cng_filt[4] = { 273, 998, 499, 333 };
diff --git a/libavcodec/g723_1enc.c b/libavcodec/g723_1enc.c
new file mode 100644
index 0000000000..1ebd465416
--- /dev/null
+++ b/libavcodec/g723_1enc.c
@@ -0,0 +1,1202 @@
+/*
+ * G.723.1 compatible encoder
+ * Copyright (c) Mohamed Naufal <naufal22@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * G.723.1 compatible encoder
+ */
+
+#include <stdint.h>
+#include <string.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/mem.h"
+#include "libavutil/opt.h"
+
+#include "avcodec.h"
+#include "celp_math.h"
+#include "g723_1.h"
+#include "internal.h"
+
+#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 AVERROR(EINVAL);
+    }
+
+    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 6300\n");
+        return AVERROR_PATCHWELCOME;
+    } else {
+        av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\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];
+
+    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       = ff_g723_1_normalize_bits(temp, 31);
+    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 = ff_g723_1_normalize_bits(max, 31);
+
+    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        = ff_g723_1_normalize_bits(sum, 31);
+            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  = 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 = ff_g723_1_normalize_bits(max, 15);
+    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);
+}
+
+/**
+ * Perform IIR filtering.
+ *
+ * @param fir_coef FIR coefficients
+ * @param iir_coef IIR coefficients
+ * @param src      source vector
+ * @param dest     destination vector
+ */
+static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
+                       int16_t *src, int16_t *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];
+        }
+
+        dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) +
+                                 (1 << 15)) >> 16;
+    }
+}
+
+/**
+ * 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);
+        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   = ff_g723_1_normalize_bits(ccr, 31);
+        ccr   = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16;
+        exp <<= 1;
+        ccr  *= ccr;
+        temp  = ff_g723_1_normalize_bits(ccr, 31);
+        ccr   = ccr << temp >> 16;
+        exp  += temp;
+
+        temp = ff_g723_1_normalize_bits(orig_eng, 31);
+        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 = ff_g723_1_normalize_bits(max, 31);
+    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++) {
+        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++) {
+            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 = ff_g723_1_normalize_bits(max, 31);
+
+    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;
+        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 = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
+
+    scale           = ff_g723_1_normalize_bits(temp, 31);
+    impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
+
+    for (i = 1; i < 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++) {
+        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)
+        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, AVPacket *avpkt)
+{
+    PutBitContext pb;
+    int info_bits = 0;
+    int i, temp;
+
+    init_put_bits(&pb, avpkt->data, avpkt->size);
+
+    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, i, j;
+    int16_t *in, *start;
+    HFParam hf[4];
+
+    /* duplicate input */
+    start = in = av_malloc(frame->nb_samples * sizeof(int16_t));
+    if (!in)
+        return AVERROR(ENOMEM);
+    memcpy(in, frame->data[0], frame->nb_samples * sizeof(int16_t));
+
+    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);
+
+    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);
+
+    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);
+        ff_g723_1_gen_acb_excitation(residual, p->prev_excitation,
+                                     p->pitch_lag[i >> 1], &p->subframe[i],
+                                     RATE_6300);
+        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_free(start);
+
+    ret = ff_alloc_packet(avpkt, 24);
+    if (ret < 0)
+        return ret;
+
+    *got_packet_ptr = 1;
+    return pack_bitstream(p, avpkt);
+}
+
+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
+    },
+};
diff --git a/libavcodec/version.h b/libavcodec/version.h
index f31a93d8f1..29bc4b7e54 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -29,8 +29,8 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 57
-#define LIBAVCODEC_VERSION_MINOR  9
-#define LIBAVCODEC_VERSION_MICRO  1
+#define LIBAVCODEC_VERSION_MINOR 10
+#define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \