Merge commit '165cc6fb'

* commit '165cc6fb':
  g723_1: Move sharable functions to a separate file
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

libavcodec/Makefile

@@ -276,9 +276,9 @@ 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-$(CONFIG_G723_1_DECODER)          += g723_1dec.o acelp_vectors.o \
+OBJS-$(CONFIG_G723_1_DECODER)          += g723_1dec.o g723_1.o acelp_vectors.o \
                                           celp_filters.o celp_math.o
-OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1dec.o acelp_vectors.o celp_math.o
+OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1dec.o g723_1.o acelp_vectors.o celp_math.o
 OBJS-$(CONFIG_G729_DECODER)            += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
 OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o
 OBJS-$(CONFIG_GIF_ENCODER)             += gif.o lzwenc.o

libavcodec/g723_1.c

+/*
+ * G.723.1 compatible decoder
+ * Copyright (c) 2006 Benjamin Larsson
+ * Copyright (c) 2010 Mohamed Naufal Basheer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#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]);
+
+    bits= 14 - av_log2_16bit(max);
+    bits= FFMAX(bits, 0);
+
+    for (i = 0; i < length; i++)
+        dst[i] = vector[i] << bits >> 3;
+
+    return bits - 3;
+}
+
+int ff_g723_1_normalize_bits(int num, int width)
+{
+    return width - av_log2(num) - 1;
+}
+
+int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length)
+{
+    int sum = ff_dot_product(a,b,length);
+    return av_sat_add32(sum, sum);
+}
+
+void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation,
+                            int lag)
+{
+    int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
+    int i;
+
+    residual[0] = prev_excitation[offset];
+    residual[1] = prev_excitation[offset + 1];
+
+    offset += 2;
+    for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
+        residual[i] = prev_excitation[offset + (i - 2) % lag];
+}
+
+void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag)
+{
+    int16_t vector[SUBFRAME_LEN];
+    int i, j;
+
+    memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
+    for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
+        for (j = 0; j < SUBFRAME_LEN - i; j++)
+            buf[i + j] += vector[j];
+    }
+}
+
+void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
+                                  int pitch_lag, G723_1_Subframe *subfrm,
+                                  enum Rate cur_rate)
+{
+    int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
+    const int16_t *cb_ptr;
+    int lag = pitch_lag + subfrm->ad_cb_lag - 1;
+
+    int i;
+    int sum;
+
+    ff_g723_1_get_residual(residual, prev_excitation, lag);
+
+    /* Select quantization table */
+    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_dot_product(residual + i, cb_ptr, PITCH_ORDER);
+        vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
+    }
+}
+
+/**
+ * Convert LSP frequencies to LPC coefficients.
+ *
+ * @param lpc buffer for LPC coefficients
+ */
+static void lsp2lpc(int16_t *lpc)
+{
+    int f1[LPC_ORDER / 2 + 1];
+    int f2[LPC_ORDER / 2 + 1];
+    int i, j;
+
+    /* Calculate negative cosine */
+    for (j = 0; j < LPC_ORDER; j++) {
+        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));
+}

libavcodec/g723_1.h

 /*
- * G723.1 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
- * G723.1 compatible decoder data tables
+ * G.723.1 types, functions and data tables
  */
 
 #ifndef AVCODEC_G723_1_H
@@ -46,21 +46,31 @@
 #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
  */
-typedef enum FrameType {
+enum FrameType {
     ACTIVE_FRAME,        ///< Active speech
     SID_FRAME,           ///< Silence Insertion Descriptor frame
     UNTRANSMITTED_FRAME
-} FrameType;
-
-static const uint8_t frame_size[4] = { 24, 20, 4, 1 };
+};
 
-typedef enum Rate {
+/**
+ * G723.1 rate values
+ */
+enum Rate {
     RATE_6300,
     RATE_5300
-} Rate;
+};
 
 /**
  * G723.1 unpacked data subframe
@@ -104,6 +114,107 @@ typedef struct FCBParam {
     int pulse_sign[PULSE_MAX];
 } FCBParam;
 
+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;                 ///< formant postfilter
+                                 ///< gain scaling unit memory
+    int postfilter;
+
+    int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
+    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;
+
+
+/**
+ * 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
  */