G.723.1 demuxer and decoder

Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>

1 files changed, 1175 insertions, 0 deletions

diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c
new file mode 100644
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+++ b/libavcodec/g723_1.c
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+/*
+ * 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
+ */
+
+/**
+ * @file
+ * G.723.1 compatible decoder
+ */
+
+#define BITSTREAM_READER_LE
+#include "libavutil/audioconvert.h"
+#include "libavutil/lzo.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "acelp_vectors.h"
+#include "celp_filters.h"
+#include "celp_math.h"
+#include "lsp.h"
+#include "g723_1_data.h"
+
+/**
+ * G723.1 frame types
+ */
+enum FrameType {
+    ACTIVE_FRAME,        ///< Active speech
+    SID_FRAME,           ///< Silence Insertion Descriptor frame
+    UNTRANSMITTED_FRAME
+};
+
+enum Rate {
+    RATE_6300,
+    RATE_5300
+};
+
+/**
+ * G723.1 unpacked data subframe
+ */
+typedef struct {
+    int ad_cb_lag;     ///< adaptive codebook lag
+    int ad_cb_gain;
+    int dirac_train;
+    int pulse_sign;
+    int grid_index;
+    int amp_index;
+    int pulse_pos;
+} G723_1_Subframe;
+
+/**
+ * Pitch postfilter parameters
+ */
+typedef struct {
+    int     index;    ///< postfilter backward/forward lag
+    int16_t opt_gain; ///< optimal gain
+    int16_t sc_gain;  ///< scaling gain
+} PPFParam;
+
+typedef struct g723_1_context {
+    AVClass *class;
+    AVFrame frame;
+
+    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 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;
+    int postfilter;
+
+    int16_t audio[FRAME_LEN + LPC_ORDER];
+} G723_1_Context;
+
+static av_cold int g723_1_decode_init(AVCodecContext *avctx)
+{
+    G723_1_Context *p = avctx->priv_data;
+
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;
+    avctx->channels       = 1;
+    avctx->sample_rate    = 8000;
+    p->pf_gain            = 1 << 12;
+
+    avcodec_get_frame_defaults(&p->frame);
+    avctx->coded_frame    = &p->frame;
+
+    memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
+
+    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 = UNTRANSMITTED_FRAME;
+        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 = SID_FRAME;
+        p->subframe[0].amp_index = get_bits(&gb, 6);
+        return 0;
+    }
+
+    /* Extract the info common to both rates */
+    p->cur_rate       = info_bits ? RATE_5300 : RATE_6300;
+    p->cur_frame_type = ACTIVE_FRAME;
+
+    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 == RATE_6300 && 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 == RATE_6300) {
+        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 { /* 5300 bps */
+        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)
+{
+    if (!num)
+        return 0;
+    if (num == -1)
+        return width;
+    if (num < 0)
+        num = ~num;
+
+    return width - av_log2(num);
+}
+
+/**
+ * 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;
+
+
+    for (i = 0; i < length; i++)
+        max = FFMAX(max, FFABS(vector[i]));
+
+    bits  = normalize_bits(max, 15);
+    scale = (bits == 15) ? 0x7FFF : (1 << bits);
+
+    for (i = 0; i < length; i++)
+        vector[i] = (vector[i] * scale) >> 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(*cur_lsp));
+}
+
+/**
+ * Bitexact implementation of 2ab scaled by 1/2^16.
+ *
+ * @param a 32 bit multiplicand
+ * @param b 16 bit multiplier
+ */
+#define MULL2(a, b) \
+        ((((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(*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
+ * @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,
+                               enum Rate cur_rate, int pitch_lag, int index)
+{
+    int temp, i, j;
+
+    memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
+
+    if (cur_rate == RATE_6300) {
+        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 { /* 5300 bps */
+        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];
+}
+
+static int dot_product(const int16_t *a, const int16_t *b, int length,
+                       int shift)
+{
+    int i, sum = 0;
+
+    for (i = 0; i < length; i++) {
+        int64_t prod = av_clipl_int32(MUL64(a[i], b[i]) << shift);
+        sum = av_clipl_int32(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;
+    int64_t 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, 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 = 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, enum 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, enum 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] = dot_product(buf, buf, SUBFRAME_LEN, 1);
+
+    /* Compute forward residual energy */
+    if (fwd_lag)
+        energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag,
+                                SUBFRAME_LEN, 1);
+
+    /* Compute backward residual energy */
+    if (back_lag)
+        energy[4] = 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, 31);
+    for (i = 0; i < 5; i++)
+        energy[i] = (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  = dot_product(buf, buf, SUBFRAME_LEN * 2, 1);
+    *exc_eng = av_clipl_int32((int64_t)tgt_eng + (1 << 15)) >> 16;
+
+    if (ccr <= 0)
+        return 0;
+
+    /* Compute best energy */
+    best_eng = 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(*vector_ptr),
+                          FRAME_LEN * sizeof(*vector_ptr));
+        memcpy(out, vector_ptr, FRAME_LEN * sizeof(*vector_ptr));
+    } 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(*buf));
+    }
+}
+
+/**
+ * Perform IIR filtering.
+ *
+ * @param fir_coef FIR coefficients
+ * @param iir_coef IIR coefficients
+ * @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)
+{
+    int m, n;
+
+    for (m = 0; m < SUBFRAME_LEN; m++) {
+        int64_t filter = 0;
+        for (n = 1; n <= LPC_ORDER; n++) {
+            filter -= fir_coef[n - 1] * src[m - n] -
+                      iir_coef[n - 1] * (dest[m - n] >> 16);
+        }
+
+        dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + (1 << 15));
+    }
+}
+
+/**
+ * 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,   31);
+        bits2   = normalize_bits(denom, 31);
+        num     = num << bits1 >> 1;
+        denom <<= bits2;
+
+        bits2 = 5 + bits1 - bits2;
+        bits2 = FFMAX(0, bits2);
+
+        gain = (num >> 1) / (denom >> 16);
+        gain = square_root(gain << 16 >> bits2);
+    } else {
+        gain = 1 << 12;
+    }
+
+    for (i = 0; i < SUBFRAME_LEN; i++) {
+        p->pf_gain = ((p->pf_gain << 4) - p->pf_gain + gain + (1 << 3)) >> 4;
+        buf[i]     = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
+                                   (1 << 10)) >> 11);
+    }
+}
+
+/**
+ * Perform formant filtering.
+ *
+ * @param p   the context
+ * @param lpc quantized lpc coefficients
+ * @param buf output buffer
+ */
+static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
+{
+    int16_t filter_coef[2][LPC_ORDER], *buf_ptr;
+    int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
+    int i, j, k;
+
+    memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
+    memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
+
+    for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
+        for (k = 0; k < LPC_ORDER; k++) {
+            filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
+                                 (1 << 14)) >> 15;
+            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);
+    }
+
+    memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(*p->fir_mem));
+    memcpy(p->iir_mem, filter_signal + FRAME_LEN,
+           LPC_ORDER * sizeof(*p->iir_mem));
+
+    buf_ptr    = buf + LPC_ORDER;
+    signal_ptr = filter_signal + LPC_ORDER;
+    for (i = 0; i < SUBFRAMES; i++) {
+        int16_t temp_vector[SUBFRAME_LEN];
+        int16_t temp;
+        int auto_corr[2];
+        int scale, energy;
+
+        /* Normalize */
+        memcpy(temp_vector, buf_ptr, SUBFRAME_LEN * sizeof(*temp_vector));
+        scale = scale_vector(temp_vector, SUBFRAME_LEN);
+
+        /* Compute auto correlation coefficients */
+        auto_corr[0] = dot_product(temp_vector, temp_vector + 1,
+                                   SUBFRAME_LEN - 1, 1);
+        auto_corr[1] = dot_product(temp_vector, temp_vector, SUBFRAME_LEN, 1);
+
+        /* Compute reflection coefficient */
+        temp = auto_corr[1] >> 16;
+        if (temp) {
+            temp = (auto_corr[0] >> 2) / temp;
+        }
+        p->reflection_coef = ((p->reflection_coef << 2) - p->reflection_coef +
+                              temp + 2) >> 2;
+        temp = (p->reflection_coef * 0xffffc >> 3) & 0xfffc;
+
+        /* Compensation filter */
+        for (j = 0; j < SUBFRAME_LEN; j++) {
+            buf_ptr[j] = av_clipl_int32(signal_ptr[j] +
+                                        ((signal_ptr[j - 1] >> 16) *
+                                         temp << 1)) >> 16;
+        }
+
+        /* Compute normalized signal energy */
+        temp = 2 * scale + 4;
+        if (temp < 0) {
+            energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
+        } else
+            energy = auto_corr[1] >> temp;
+
+        gain_scale(p, buf_ptr, energy);
+
+        buf_ptr    += SUBFRAME_LEN;
+        signal_ptr += SUBFRAME_LEN;
+    }
+}
+
+static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
+                               int *got_frame_ptr, AVPacket *avpkt)
+{
+    G723_1_Context *p  = avctx->priv_data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size       = avpkt->size;
+    int dec_mode       = buf[0] & 3;
+
+    PPFParam ppf[SUBFRAMES];
+    int16_t cur_lsp[LPC_ORDER];
+    int16_t lpc[SUBFRAMES * LPC_ORDER];
+    int16_t acb_vector[SUBFRAME_LEN];
+    int16_t *vector_ptr;
+    int bad_frame = 0, i, j, ret;
+
+    if (buf_size < frame_size[dec_mode]) {
+        if (buf_size)
+            av_log(avctx, AV_LOG_WARNING,
+                   "Expected %d bytes, got %d - skipping packet\n",
+                   frame_size[dec_mode], buf_size);
+        *got_frame_ptr = 0;
+        return buf_size;
+    }
+
+    if (unpack_bitstream(p, buf, buf_size) < 0) {
+        bad_frame = 1;
+        if (p->past_frame_type == ACTIVE_FRAME)
+            p->cur_frame_type = ACTIVE_FRAME;
+        else
+            p->cur_frame_type = UNTRANSMITTED_FRAME;
+    }
+
+    p->frame.nb_samples = FRAME_LEN;
+    if ((ret = avctx->get_buffer(avctx, &p->frame)) < 0) {
+         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+         return ret;
+    }
+
+    if (p->cur_frame_type == ACTIVE_FRAME) {
+        if (!bad_frame)
+            p->erased_frames = 0;
+        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);
+
+        /* Save the lsp_vector for the next frame */
+        memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
+
+        /* Generate the excitation for the frame */
+        memcpy(p->excitation, p->prev_excitation,
+               PITCH_MAX * sizeof(*p->excitation));
+        vector_ptr = p->excitation + PITCH_MAX;
+        if (!p->erased_frames) {
+            /* Update interpolation gain memory */
+            p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
+                                            p->subframe[3].amp_index) >> 1];
+            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);
+                /* Get the total excitation */
+                for (j = 0; j < SUBFRAME_LEN; j++) {
+                    vector_ptr[j] = av_clip_int16(vector_ptr[j] << 1);
+                    vector_ptr[j] = av_clip_int16(vector_ptr[j] +
+                                                  acb_vector[j]);
+                }
+                vector_ptr += SUBFRAME_LEN;
+            }
+
+            vector_ptr = p->excitation + PITCH_MAX;
+
+            /* Save the excitation */
+            memcpy(p->audio, vector_ptr, FRAME_LEN * sizeof(*p->audio));
+
+            p->interp_index = comp_interp_index(p, p->pitch_lag[1],
+                                                &p->sid_gain, &p->cur_gain);
+
+            if (p->postfilter) {
+                i = PITCH_MAX;
+                for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+                    comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
+                                   ppf + j, p->cur_rate);
+            }
+
+            /* Restore the original excitation */
+            memcpy(p->excitation, p->prev_excitation,
+                   PITCH_MAX * sizeof(*p->excitation));
+            memcpy(vector_ptr, p->audio, FRAME_LEN * sizeof(*vector_ptr));
+
+            /* Peform pitch postfiltering */
+            if (p->postfilter)
+                for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+                    ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
+                                                 vector_ptr + i,
+                                                 vector_ptr + i + ppf[j].index,
+                                                 ppf[j].sc_gain,
+                                                 ppf[j].opt_gain,
+                                                 1 << 14, 15, SUBFRAME_LEN);
+
+        } else {
+            p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
+            if (p->erased_frames == 3) {
+                /* Mute output */
+                memset(p->excitation, 0,
+                       (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
+                memset(p->frame.data[0], 0,
+                       (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
+            } else {
+                /* Regenerate frame */
+                residual_interp(p->excitation, p->audio + LPC_ORDER, p->interp_index,
+                                p->interp_gain, &p->random_seed);
+            }
+        }
+        /* Save the excitation for the next frame */
+        memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
+               PITCH_MAX * sizeof(*p->excitation));
+    } else {
+        memset(p->frame.data[0], 0, FRAME_LEN * 2);
+        av_log(avctx, AV_LOG_WARNING,
+               "G.723.1: Comfort noise generation not supported yet\n");
+
+        *got_frame_ptr   = 1;
+        *(AVFrame *)data = p->frame;
+        return frame_size[dec_mode];
+    }
+
+    p->past_frame_type = p->cur_frame_type;
+
+    memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
+    for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+        ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
+                                    p->audio + i, SUBFRAME_LEN, LPC_ORDER,
+                                    0, 1, 1 << 12);
+    memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
+
+    if (p->postfilter)
+        formant_postfilter(p, lpc, p->audio);
+
+    memcpy(p->frame.data[0], p->audio + LPC_ORDER, FRAME_LEN * 2);
+
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = p->frame;
+
+    return frame_size[dec_mode];
+}
+
+#define OFFSET(x) offsetof(G723_1_Context, x)
+#define AD     AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
+
+static const AVOption options[] = {
+    { "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,
+      { 1 }, 0, 1, AD },
+    { NULL }
+};
+
+
+static const AVClass g723_1dec_class = {
+    .class_name = "G.723.1 decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_g723_1_decoder = {
+    .name           = "g723_1",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_G723_1,
+    .priv_data_size = sizeof(G723_1_Context),
+    .init           = g723_1_decode_init,
+    .decode         = g723_1_decode_frame,
+    .long_name      = NULL_IF_CONFIG_SMALL("G.723.1"),
+    .capabilities   = CODEC_CAP_SUBFRAMES,
+    .priv_class     = &g723_1dec_class,
+};