g723_1: Move sharable functions to a separate file

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

4 files changed, 443 insertions, 373 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index dfefab66f8..85738fa1f0 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -225,8 +225,8 @@ OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
 OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o elsdec.o
-OBJS-$(CONFIG_G723_1_DECODER)          += g723_1dec.o acelp_vectors.o \
-                                          celp_filters.o
+OBJS-$(CONFIG_G723_1_DECODER)          += g723_1dec.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/g723_1.c b/libavcodec/g723_1.c
new file mode 100644
index 0000000000..af4777cc35
--- /dev/null
+++ b/libavcodec/g723_1.c
@@ -0,0 +1,267 @@
+/*
+ * G.723.1 compatible decoder
+ * Copyright (c) 2006 Benjamin Larsson
+ * Copyright (c) 2010 Mohamed Naufal Basheer
+ *
+ * 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
+ */
+
+#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]);
+
+    max  = FFMIN(max, 0x7FFF);
+    bits = ff_g723_1_normalize_bits(max, 15);
+
+    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)
+{
+    int i, sum = 0;
+
+    for (i = 0; i < length; i++) {
+        int prod = a[i] * b[i];
+        sum = av_sat_dadd32(sum, prod);
+    }
+    return 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 */
+    if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2)
+        cb_ptr = adaptive_cb_gain85;
+    else
+        cb_ptr = adaptive_cb_gain170;
+
+    /* Calculate adaptive vector */
+    cb_ptr += subfrm->ad_cb_gain * 20;
+    for (i = 0; i < SUBFRAME_LEN; i++) {
+        sum       = ff_g723_1_dot_product(residual + i, cb_ptr, PITCH_ORDER);
+        vector[i] = av_sat_dadd32(1 << 15, 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++) {
+        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;
+    }
+}
+
+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));
+}
diff --git a/libavcodec/g723_1.h b/libavcodec/g723_1.h
index 71e2df4ad3..391ca464a9 100644
--- a/libavcodec/g723_1.h
+++ b/libavcodec/g723_1.h
@@ -1,5 +1,5 @@
 /*
- * G.723.1 compatible decoder data tables.
+ * G.723.1 common header and data tables
  * Copyright (c) 2006 Benjamin Larsson
  * Copyright (c) 2010 Mohamed Naufal Basheer
  *
@@ -22,7 +22,7 @@
 
 /**
  * @file
- * G.723.1 compatible decoder data tables
+ * G.723.1 types, functions and data tables
  */
 
 #ifndef AVCODEC_G723_1_H
@@ -44,6 +44,143 @@
 #define GAIN_LEVELS     24
 #define COS_TBL_SIZE    512
 
+/**
+ * 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))
+
+/**
+ * G723.1 frame types
+ */
+enum FrameType {
+    ACTIVE_FRAME,        ///< Active speech
+    SID_FRAME,           ///< Silence Insertion Descriptor frame
+    UNTRANSMITTED_FRAME
+};
+
+/**
+ * G723.1 rate values
+ */
+enum Rate {
+    RATE_6300,
+    RATE_5300
+};
+
+/**
+ * G723.1 unpacked data subframe
+ */
+typedef struct G723_1_Subframe {
+    int ad_cb_lag;     ///< adaptive codebook lag
+    int ad_cb_gain;
+    int dirac_train;
+    int pulse_sign;
+    int grid_index;
+    int amp_index;
+    int pulse_pos;
+} G723_1_Subframe;
+
+/**
+ * Pitch postfilter parameters
+ */
+typedef struct PPFParam {
+    int     index;    ///< postfilter backward/forward lag
+    int16_t opt_gain; ///< optimal gain
+    int16_t sc_gain;  ///< scaling gain
+} PPFParam;
+
+typedef struct g723_1_context {
+    AVClass *class;
+
+    G723_1_Subframe subframe[4];
+    enum FrameType cur_frame_type;
+    enum FrameType past_frame_type;
+    enum Rate cur_rate;
+    uint8_t lsp_index[LSP_BANDS];
+    int pitch_lag[2];
+    int erased_frames;
+
+    int16_t prev_lsp[LPC_ORDER];
+    int16_t sid_lsp[LPC_ORDER];
+    int16_t prev_excitation[PITCH_MAX];
+    int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
+    int16_t synth_mem[LPC_ORDER];
+    int16_t fir_mem[LPC_ORDER];
+    int     iir_mem[LPC_ORDER];
+
+    int random_seed;
+    int cng_random_seed;
+    int interp_index;
+    int interp_gain;
+    int sid_gain;
+    int cur_gain;
+    int reflection_coef;
+    int pf_gain;
+    int postfilter;
+
+    int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
+} G723_1_Context;
+
+
+/**
+ * Scale vector contents based on the largest of their absolutes.
+ */
+int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length);
+
+/**
+ * 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)
+ */
+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);
+
+/**
+ * Get delayed contribution from the previous excitation vector.
+ */
+void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation,
+                            int lag);
+
+/**
+ * Generate a train of dirac functions with period as pitch lag.
+ */
+void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag);
+
+
+/**
+ * Generate adaptive codebook excitation.
+ */
+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);
+/**
+ * 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
+ */
+void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp,
+                               int16_t *prev_lsp);
+
+/**
+ * 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
+ */
+void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
+                             uint8_t *lsp_index, int bad_frame);
+
+
 static const uint8_t frame_size[4] = { 24, 20, 4, 1 };
 
 /* Postfilter gain weighting factors scaled by 2^15 */
diff --git a/libavcodec/g723_1dec.c b/libavcodec/g723_1dec.c
index dc05ed2121..99043169fe 100644
--- a/libavcodec/g723_1dec.c
+++ b/libavcodec/g723_1dec.c
@@ -38,74 +38,6 @@
 
 #define CNG_RANDOM_SEED 12345
 
-/**
- * G723.1 frame types
- */
-enum FrameType {
-    ACTIVE_FRAME,        ///< Active speech
-    SID_FRAME,           ///< Silence Insertion Descriptor frame
-    UNTRANSMITTED_FRAME
-};
-
-enum Rate {
-    RATE_6300,
-    RATE_5300
-};
-
-/**
- * G723.1 unpacked data subframe
- */
-typedef struct G723_1_Subframe {
-    int ad_cb_lag;     ///< adaptive codebook lag
-    int ad_cb_gain;
-    int dirac_train;
-    int pulse_sign;
-    int grid_index;
-    int amp_index;
-    int pulse_pos;
-} G723_1_Subframe;
-
-/**
- * Pitch postfilter parameters
- */
-typedef struct PPFParam {
-    int     index;    ///< postfilter backward/forward lag
-    int16_t opt_gain; ///< optimal gain
-    int16_t sc_gain;  ///< scaling gain
-} PPFParam;
-
-typedef struct g723_1_context {
-    AVClass *class;
-
-    G723_1_Subframe subframe[4];
-    enum FrameType cur_frame_type;
-    enum FrameType past_frame_type;
-    enum Rate cur_rate;
-    uint8_t lsp_index[LSP_BANDS];
-    int pitch_lag[2];
-    int erased_frames;
-
-    int16_t prev_lsp[LPC_ORDER];
-    int16_t sid_lsp[LPC_ORDER];
-    int16_t prev_excitation[PITCH_MAX];
-    int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
-    int16_t synth_mem[LPC_ORDER];
-    int16_t fir_mem[LPC_ORDER];
-    int     iir_mem[LPC_ORDER];
-
-    int random_seed;
-    int cng_random_seed;
-    int interp_index;
-    int interp_gain;
-    int sid_gain;
-    int cur_gain;
-    int reflection_coef;
-    int pf_gain;
-    int postfilter;
-
-    int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
-} G723_1_Context;
-
 static av_cold int g723_1_decode_init(AVCodecContext *avctx)
 {
     G723_1_Context *p = avctx->priv_data;
@@ -263,108 +195,6 @@ static int16_t square_root(int val)
 }
 
 /**
- * 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)
-{
-    return width - av_log2(num) - 1;
-}
-
-/**
- * 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]);
-
-    max   = FFMIN(max, 0x7FFF);
-    bits  = normalize_bits(max, 15);
-
-    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
@@ -374,116 +204,6 @@ static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
         ((((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) & 0x1FF;
-        int offset    = lpc[j] & 0x7f;
-        int temp1     = cos_tab[index] << 16;
-        int temp2     = (cos_tab[index + 1] - cos_tab[index]) *
-                          ((offset << 8) + 0x80) << 1;
-
-        lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
-    }
-
-    /*
-     * Compute sum and difference polynomial coefficients
-     * (bitexact alternative to lsp2poly() in lsp.c)
-     */
-    /* Initialize with values in Q28 */
-    f1[0] = 1 << 28;
-    f1[1] = (lpc[0] << 14) + (lpc[2] << 14);
-    f1[2] = lpc[0] * lpc[2] + (2 << 28);
-
-    f2[0] = 1 << 28;
-    f2[1] = (lpc[1] << 14) + (lpc[3] << 14);
-    f2[2] = lpc[1] * lpc[3] + (2 << 28);
-
-    /*
-     * Calculate and scale the coefficients by 1/2 in
-     * each iteration for a final scaling factor of Q25
-     */
-    for (i = 2; i < LPC_ORDER / 2; i++) {
-        f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);
-        f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);
-
-        for (j = i; j >= 2; j--) {
-            f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +
-                    (f1[j] >> 1) + (f1[j - 2] >> 1);
-            f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +
-                    (f2[j] >> 1) + (f2[j - 2] >> 1);
-        }
-
-        f1[0] >>= 1;
-        f2[0] >>= 1;
-        f1[1] = ((lpc[2 * i]     << 16 >> i) + f1[1]) >> 1;
-        f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
-    }
-
-    /* Convert polynomial coefficients to LPC coefficients */
-    for (i = 0; i < LPC_ORDER / 2; i++) {
-        int64_t ff1 = f1[i + 1] + f1[i];
-        int64_t ff2 = f2[i + 1] - f2[i];
-
-        lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;
-        lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +
-                                                (1 << 15)) >> 16;
-    }
-}
-
-/**
- * Quantize LSP frequencies by interpolation and convert them to
- * the corresponding LPC coefficients.
- *
- * @param lpc      buffer for LPC coefficients
- * @param cur_lsp  the current LSP vector
- * @param prev_lsp the previous LSP vector
- */
-static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
-{
-    int i;
-    int16_t *lpc_ptr = lpc;
-
-    /* cur_lsp * 0.25 + prev_lsp * 0.75 */
-    ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,
-                                 4096, 12288, 1 << 13, 14, LPC_ORDER);
-    ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,
-                                 8192, 8192, 1 << 13, 14, LPC_ORDER);
-    ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,
-                                 12288, 4096, 1 << 13, 14, LPC_ORDER);
-    memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc));
-
-    for (i = 0; i < SUBFRAMES; i++) {
-        lsp2lpc(lpc_ptr);
-        lpc_ptr += LPC_ORDER;
-    }
-}
-
-/**
- * Generate a train of dirac functions with period as pitch lag.
- */
-static void gen_dirac_train(int16_t *buf, int pitch_lag)
-{
-    int16_t vector[SUBFRAME_LEN];
-    int i, j;
-
-    memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
-    for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
-        for (j = 0; j < SUBFRAME_LEN - i; j++)
-            buf[i + j] += vector[j];
-    }
-}
-
-/**
  * Generate fixed codebook excitation vector.
  *
  * @param vector    decoded excitation vector
@@ -522,7 +242,7 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
                 break;
         }
         if (subfrm->dirac_train == 1)
-            gen_dirac_train(vector, pitch_lag);
+            ff_g723_1_gen_dirac_train(vector, pitch_lag);
     } else { /* 5300 bps */
         int cb_gain  = fixed_cb_gain[subfrm->amp_index];
         int cb_shift = subfrm->grid_index;
@@ -550,63 +270,6 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
 }
 
 /**
- * Get delayed contribution from the previous excitation vector.
- */
-static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
-{
-    int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
-    int i;
-
-    residual[0] = prev_excitation[offset];
-    residual[1] = prev_excitation[offset + 1];
-
-    offset += 2;
-    for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
-        residual[i] = prev_excitation[offset + (i - 2) % lag];
-}
-
-static int dot_product(const int16_t *a, const int16_t *b, int length)
-{
-    int i, sum = 0;
-
-    for (i = 0; i < length; i++) {
-        int prod = a[i] * b[i];
-        sum = av_sat_dadd32(sum, prod);
-    }
-    return sum;
-}
-
-/**
- * Generate adaptive codebook excitation.
- */
-static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
-                               int pitch_lag, G723_1_Subframe *subfrm,
-                               enum Rate cur_rate)
-{
-    int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
-    const int16_t *cb_ptr;
-    int lag = pitch_lag + subfrm->ad_cb_lag - 1;
-
-    int i;
-    int sum;
-
-    get_residual(residual, prev_excitation, lag);
-
-    /* Select quantization table */
-    if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2)
-        cb_ptr = adaptive_cb_gain85;
-    else
-        cb_ptr = adaptive_cb_gain170;
-
-    /* Calculate adaptive vector */
-    cb_ptr += subfrm->ad_cb_gain * 20;
-    for (i = 0; i < SUBFRAME_LEN; i++) {
-        sum = dot_product(residual + i, cb_ptr, PITCH_ORDER);
-        vector[i] = av_sat_dadd32(1 << 15, sum) >> 16;
-    }
-}
-
-/**
  * Estimate maximum auto-correlation around pitch lag.
  *
  * @param buf       buffer with offset applied
@@ -629,7 +292,7 @@ static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
         limit = pitch_lag + 3;
 
     for (i = pitch_lag - 3; i <= limit; i++) {
-        ccr = dot_product(buf, buf + dir * i, length);
+        ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
 
         if (ccr > *ccr_max) {
             *ccr_max = ccr;
@@ -728,22 +391,24 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
         return;
 
     /* Compute target energy */
-    energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
+    energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);
 
     /* Compute forward residual energy */
     if (fwd_lag)
-        energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
+        energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
+                                          SUBFRAME_LEN);
 
     /* Compute backward residual energy */
     if (back_lag)
-        energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);
+        energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
+                                          SUBFRAME_LEN);
 
     /* Normalize and shorten */
     temp1 = 0;
     for (i = 0; i < 5; i++)
         temp1 = FFMAX(energy[i], temp1);
 
-    scale = normalize_bits(temp1, 31);
+    scale = ff_g723_1_normalize_bits(temp1, 31);
     for (i = 0; i < 5; i++)
         energy[i] = (energy[i] << scale) >> 16;
 
@@ -789,7 +454,7 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag,
 
     int index, ccr, tgt_eng, best_eng, temp;
 
-    *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
+    *scale = ff_g723_1_scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
     buf   += offset;
 
     /* Compute maximum backward cross-correlation */
@@ -798,14 +463,15 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag,
     ccr   = av_sat_add32(ccr, 1 << 15) >> 16;
 
     /* Compute target energy */
-    tgt_eng  = dot_product(buf, buf, SUBFRAME_LEN * 2);
+    tgt_eng  = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);
     *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
 
     if (ccr <= 0)
         return 0;
 
     /* Compute best energy */
-    best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);
+    best_eng = ff_g723_1_dot_product(buf - index, buf - index,
+                                     SUBFRAME_LEN * 2);
     best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
 
     temp = best_eng * *exc_eng >> 3;
@@ -853,8 +519,8 @@ static void residual_interp(int16_t *buf, int16_t *out, int lag,
  * @param src      source vector
  * @param dest     destination vector
  */
-static inline void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
-                              int16_t *src, int *dest)
+static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
+                       int16_t *src, int *dest)
 {
     int m, n;
 
@@ -890,8 +556,8 @@ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
     }
 
     if (num && denom) {
-        bits1   = normalize_bits(num,   31);
-        bits2   = normalize_bits(denom, 31);
+        bits1   = ff_g723_1_normalize_bits(num,   31);
+        bits2   = ff_g723_1_normalize_bits(denom, 31);
         num     = num << bits1 >> 1;
         denom <<= bits2;
 
@@ -936,8 +602,7 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
             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);
+        iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i);
         lpc += LPC_ORDER;
     }
 
@@ -953,11 +618,11 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
         int scale, energy;
 
         /* Normalize */
-        scale = scale_vector(dst, buf, SUBFRAME_LEN);
+        scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);
 
         /* Compute auto correlation coefficients */
-        auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
-        auto_corr[1] = dot_product(dst, dst,     SUBFRAME_LEN);
+        auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
+        auto_corr[1] = ff_g723_1_dot_product(dst, dst,     SUBFRAME_LEN);
 
         /* Compute reflection coefficient */
         temp = auto_corr[1] >> 16;
@@ -1104,13 +769,13 @@ static void generate_noise(G723_1_Context *p)
     memcpy(vector_ptr, p->prev_excitation,
            PITCH_MAX * sizeof(*p->excitation));
     for (i = 0; i < SUBFRAMES; i += 2) {
-        gen_acb_excitation(vector_ptr, vector_ptr,
-                           p->pitch_lag[i >> 1], &p->subframe[i],
-                           p->cur_rate);
-        gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
-                           vector_ptr + SUBFRAME_LEN,
-                           p->pitch_lag[i >> 1], &p->subframe[i + 1],
-                           p->cur_rate);
+        ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
+                                     p->pitch_lag[i >> 1], &p->subframe[i],
+                                     p->cur_rate);
+        ff_g723_1_gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
+                                     vector_ptr + SUBFRAME_LEN,
+                                     p->pitch_lag[i >> 1], &p->subframe[i + 1],
+                                     p->cur_rate);
 
         t = 0;
         for (j = 0; j < SUBFRAME_LEN * 2; j++)
@@ -1231,8 +896,8 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
         else if (p->erased_frames != 3)
             p->erased_frames++;
 
-        inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
-        lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
+        ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
+        ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
 
         /* Save the lsp_vector for the next frame */
         memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
@@ -1249,9 +914,10 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
             for (i = 0; i < SUBFRAMES; i++) {
                 gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
                                    p->pitch_lag[i >> 1], i);
-                gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
-                                   p->pitch_lag[i >> 1], &p->subframe[i],
-                                   p->cur_rate);
+                ff_g723_1_gen_acb_excitation(acb_vector,
+                                             &p->excitation[SUBFRAME_LEN * i],
+                                             p->pitch_lag[i >> 1],
+                                             &p->subframe[i], p->cur_rate);
                 /* Get the total excitation */
                 for (j = 0; j < SUBFRAME_LEN; j++) {
                     int v = av_clip_int16(vector_ptr[j] << 1);
@@ -1312,7 +978,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
     } else {
         if (p->cur_frame_type == SID_FRAME) {
             p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
-            inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
+            ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
         } else if (p->past_frame_type == ACTIVE_FRAME) {
             p->sid_gain = estimate_sid_gain(p);
         }
@@ -1322,7 +988,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
         else
             p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
         generate_noise(p);
-        lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
+        ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
         /* Save the lsp_vector for the next frame */
         memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
     }