ffmpeg roll of source to mar 9 version...

patched-ffmpeg-mt/libavcodec/amrnbdec.c

Index: patched-ffmpeg-mt/libavcodec/amrnbdec.c
===================================================================
--- patched-ffmpeg-mt/libavcodec/amrnbdec.c	(revision 0)
+++ patched-ffmpeg-mt/libavcodec/amrnbdec.c	(revision 0)
@@ -0,0 +1,1081 @@
+/*
+ * AMR narrowband decoder
+ * Copyright (c) 2006-2007 Robert Swain
+ * Copyright (c) 2009 Colin McQuillan
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+
+/**
+ * @file libavcodec/amrnbdec.c
+ * AMR narrowband decoder
+ *
+ * This decoder uses floats for simplicity and so is not bit-exact. One
+ * difference is that differences in phase can accumulate. The test sequences
+ * in 3GPP TS 26.074 can still be useful.
+ *
+ * - Comparing this file's output to the output of the ref decoder gives a
+ *   PSNR of 30 to 80. Plotting the output samples shows a difference in
+ *   phase in some areas.
+ *
+ * - Comparing both decoders against their input, this decoder gives a similar
+ *   PSNR. If the test sequence homing frames are removed (this decoder does
+ *   not detect them), the PSNR is at least as good as the reference on 140
+ *   out of 169 tests.
+ */
+
+
+#include <string.h>
+#include <math.h>
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "libavutil/common.h"
+#include "celp_math.h"
+#include "celp_filters.h"
+#include "acelp_filters.h"
+#include "acelp_vectors.h"
+#include "acelp_pitch_delay.h"
+#include "lsp.h"
+
+#include "amrnbdata.h"
+
+#define AMR_BLOCK_SIZE              160   ///< samples per frame
+#define AMR_SAMPLE_BOUND        32768.0   ///< threshold for synthesis overflow
+
+/**
+ * Scale from constructed speech to [-1,1]
+ *
+ * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
+ * upscales by two (section 6.2.2).
+ *
+ * Fundamentally, this scale is determined by energy_mean through
+ * the fixed vector contribution to the excitation vector.
+ */
+#define AMR_SAMPLE_SCALE  (2.0 / 32768.0)
+
+/** Prediction factor for 12.2kbit/s mode */
+#define PRED_FAC_MODE_12k2             0.65
+
+#define LSF_R_FAC          (8000.0 / 32768.0) ///< LSF residual tables to Hertz
+#define MIN_LSF_SPACING    (50.0488 / 8000.0) ///< Ensures stability of LPC filter
+#define PITCH_LAG_MIN_MODE_12k2          18   ///< Lower bound on decoded lag search in 12.2kbit/s mode
+
+/** Initial energy in dB. Also used for bad frames (unimplemented). */
+#define MIN_ENERGY -14.0
+
+/** Maximum sharpening factor
+ *
+ * The specification says 0.8, which should be 13107, but the reference C code
+ * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
+ */
+#define SHARP_MAX 0.79449462890625
+
+/** Number of impulse response coefficients used for tilt factor */
+#define AMR_TILT_RESPONSE   22
+/** Tilt factor = 1st reflection coefficient * gamma_t */
+#define AMR_TILT_GAMMA_T   0.8
+/** Adaptive gain control factor used in post-filter */
+#define AMR_AGC_ALPHA      0.9
+
+typedef struct AMRContext {
+    AMRNBFrame                        frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
+    uint8_t             bad_frame_indicator; ///< bad frame ? 1 : 0
+    enum Mode                cur_frame_mode;
+
+    int16_t     prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
+    double          lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
+    double   prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
+
+    float         lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
+    float          lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
+
+    float           lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
+
+    uint8_t                   pitch_lag_int; ///< integer part of pitch lag from current subframe
+
+    float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
+    float                       *excitation; ///< pointer to the current excitation vector in excitation_buf
+
+    float   pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
+    float   fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
+
+    float               prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
+    float                     pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
+    float                     fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
+
+    float                              beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
+    uint8_t                      diff_count; ///< the number of subframes for which diff has been above 0.65
+    uint8_t                      hang_count; ///< the number of subframes since a hangover period started
+
+    float            prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
+    uint8_t               prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
+    uint8_t                 ir_filter_onset; ///< flag for impulse response filter strength
+
+    float                postfilter_mem[10]; ///< previous intermediate values in the formant filter
+    float                          tilt_mem; ///< previous input to tilt compensation filter
+    float                    postfilter_agc; ///< previous factor used for adaptive gain control
+    float                  high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
+
+    float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
+
+} AMRContext;
+
+/** Double version of ff_weighted_vector_sumf() */
+static void weighted_vector_sumd(double *out, const double *in_a,
+                                 const double *in_b, double weight_coeff_a,
+                                 double weight_coeff_b, int length)
+{
+    int i;
+
+    for (i = 0; i < length; i++)
+        out[i] = weight_coeff_a * in_a[i]
+               + weight_coeff_b * in_b[i];
+}
+
+static av_cold int amrnb_decode_init(AVCodecContext *avctx)
+{
+    AMRContext *p = avctx->priv_data;
+    int i;
+
+    avctx->sample_fmt = SAMPLE_FMT_FLT;
+
+    // p->excitation always points to the same position in p->excitation_buf
+    p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
+
+    for (i = 0; i < LP_FILTER_ORDER; i++) {
+        p->prev_lsp_sub4[i] =    lsp_sub4_init[i] * 1000 / (float)(1 << 15);
+        p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
+    }
+
+    for (i = 0; i < 4; i++)
+        p->prediction_error[i] = MIN_ENERGY;
+
+    return 0;
+}
+
+
+/**
+ * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
+ *
+ * The order of speech bits is specified by 3GPP TS 26.101.
+ *
+ * @param p the context
+ * @param buf               pointer to the input buffer
+ * @param buf_size          size of the input buffer
+ *
+ * @return the frame mode
+ */
+static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
+                                  int buf_size)
+{
+    GetBitContext gb;
+    enum Mode mode;
+
+    init_get_bits(&gb, buf, buf_size * 8);
+
+    // Decode the first octet.
+    skip_bits(&gb, 1);                        // padding bit
+    mode = get_bits(&gb, 4);                  // frame type
+    p->bad_frame_indicator = !get_bits1(&gb); // quality bit
+    skip_bits(&gb, 2);                        // two padding bits
+
+    if (mode <= MODE_DTX) {
+        uint16_t *data = (uint16_t *)&p->frame;
+        const uint8_t *order = amr_unpacking_bitmaps_per_mode[mode];
+        int field_size;
+
+        memset(&p->frame, 0, sizeof(AMRNBFrame));
+        buf++;
+        while ((field_size = *order++)) {
+            int field = 0;
+            int field_offset = *order++;
+            while (field_size--) {
+               int bit = *order++;
+               field <<= 1;
+               field |= buf[bit >> 3] >> (bit & 7) & 1;
+            }
+            data[field_offset] = field;
+        }
+    }
+
+    return mode;
+}
+
+
+/// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
+/// @{
+
+/**
+ * Convert an lsf vector into an lsp vector.
+ *
+ * @param lsf               input lsf vector
+ * @param lsp               output lsp vector
+ */
+static void lsf2lsp(const float *lsf, double *lsp)
+{
+    int i;
+
+    for (i = 0; i < LP_FILTER_ORDER; i++)
+        lsp[i] = cos(2.0 * M_PI * lsf[i]);
+}
+
+/**
+ * Interpolate the LSF vector (used for fixed gain smoothing).
+ * The interpolation is done over all four subframes even in MODE_12k2.
+ *
+ * @param[in,out] lsf_q     LSFs in [0,1] for each subframe
+ * @param[in]     lsf_new   New LSFs in [0,1] for subframe 4
+ */
+static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
+{
+    int i;
+
+    for (i = 0; i < 4; i++)
+        ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
+                                0.25 * (3 - i), 0.25 * (i + 1),
+                                LP_FILTER_ORDER);
+}
+
+/**
+ * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
+ *
+ * @param p the context
+ * @param lsp output LSP vector
+ * @param lsf_no_r LSF vector without the residual vector added
+ * @param lsf_quantizer pointers to LSF dictionary tables
+ * @param quantizer_offset offset in tables
+ * @param sign for the 3 dictionary table
+ * @param update store data for computing the next frame's LSFs
+ */
+static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
+                                 const float lsf_no_r[LP_FILTER_ORDER],
+                                 const int16_t *lsf_quantizer[5],
+                                 const int quantizer_offset,
+                                 const int sign, const int update)
+{
+    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
+    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
+    int i;
+
+    for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
+        memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
+               2 * sizeof(*lsf_r));
+
+    if (sign) {
+        lsf_r[4] *= -1;
+        lsf_r[5] *= -1;
+    }
+
+    if (update)
+        memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float));
+
+    for (i = 0; i < LP_FILTER_ORDER; i++)
+        lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
+
+    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
+
+    if (update)
+        interpolate_lsf(p->lsf_q, lsf_q);
+
+    lsf2lsp(lsf_q, lsp);
+}
+
+/**
+ * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
+ *
+ * @param p                 pointer to the AMRContext
+ */
+static void lsf2lsp_5(AMRContext *p)
+{
+    const uint16_t *lsf_param = p->frame.lsf;
+    float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
+    const int16_t *lsf_quantizer[5];
+    int i;
+
+    lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
+    lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
+    lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
+    lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
+    lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
+
+    for (i = 0; i < LP_FILTER_ORDER; i++)
+        lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
+
+    lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
+    lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
+
+    // interpolate LSP vectors at subframes 1 and 3
+    weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
+    weighted_vector_sumd(p->lsp[2], p->lsp[1]       , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
+}
+
+/**
+ * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
+ *
+ * @param p                 pointer to the AMRContext
+ */
+static void lsf2lsp_3(AMRContext *p)
+{
+    const uint16_t *lsf_param = p->frame.lsf;
+    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
+    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
+    const int16_t *lsf_quantizer;
+    int i, j;
+
+    lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
+    memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
+
+    lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
+    memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
+
+    lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
+    memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
+
+    // calculate mean-removed LSF vector and add mean
+    for (i = 0; i < LP_FILTER_ORDER; i++)
+        lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);
+
+    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
+
+    // store data for computing the next frame's LSFs
+    interpolate_lsf(p->lsf_q, lsf_q);
+    memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
+
+    lsf2lsp(lsf_q, p->lsp[3]);
+
+    // interpolate LSP vectors at subframes 1, 2 and 3
+    for (i = 1; i <= 3; i++)
+        for(j = 0; j < LP_FILTER_ORDER; j++)
+            p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
+                (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
+}
+
+/// @}
+
+
+/// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions
+/// @{
+
+/**
+ * Like ff_decode_pitch_lag(), but with 1/6 resolution
+ */
+static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
+                                 const int prev_lag_int, const int subframe)
+{
+    if (subframe == 0 || subframe == 2) {
+        if (pitch_index < 463) {
+            *lag_int  = (pitch_index + 107) * 10923 >> 16;
+            *lag_frac = pitch_index - *lag_int * 6 + 105;
+        } else {
+            *lag_int  = pitch_index - 368;
+            *lag_frac = 0;
+        }
+    } else {
+        *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
+        *lag_frac = pitch_index - *lag_int * 6 - 3;
+        *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
+                            PITCH_DELAY_MAX - 9);
+    }
+}
+
+static void decode_pitch_vector(AMRContext *p,
+                                const AMRNBSubframe *amr_subframe,
+                                const int subframe)
+{
+    int pitch_lag_int, pitch_lag_frac;
+    enum Mode mode = p->cur_frame_mode;
+
+    if (p->cur_frame_mode == MODE_12k2) {
+        decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
+                             amr_subframe->p_lag, p->pitch_lag_int,
+                             subframe);
+    } else
+        ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
+                            amr_subframe->p_lag,
+                            p->pitch_lag_int, subframe,
+                            mode != MODE_4k75 && mode != MODE_5k15,
+                            mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
+
+    p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
+
+    pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
+
+    pitch_lag_int += pitch_lag_frac > 0;
+
+    /* Calculate the pitch vector by interpolating the past excitation at the
+       pitch lag using a b60 hamming windowed sinc function.   */
+    ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
+                          ff_b60_sinc, 6,
+                          pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
+                          10, AMR_SUBFRAME_SIZE);
+
+    memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
+}
+
+/// @}
+
+
+/// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions
+/// @{
+
+/**
+ * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
+ */
+static void decode_10bit_pulse(int code, int pulse_position[8],
+                               int i1, int i2, int i3)
+{
+    // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
+    // the 3 pulses and the upper 7 bits being coded in base 5
+    const uint8_t *positions = base_five_table[code >> 3];
+    pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
+    pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
+    pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
+}
+
+/**
+ * Decode the algebraic codebook index to pulse positions and signs and
+ * construct the algebraic codebook vector for MODE_10k2.
+ *
+ * @param fixed_index          positions of the eight pulses
+ * @param fixed_sparse         pointer to the algebraic codebook vector
+ */
+static void decode_8_pulses_31bits(const int16_t *fixed_index,
+                                   AMRFixed *fixed_sparse)
+{
+    int pulse_position[8];
+    int i, temp;
+
+    decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
+    decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
+
+    // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
+    // the 2 pulses and the upper 5 bits being coded in base 5
+    temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
+    pulse_position[3] = temp % 5;
+    pulse_position[7] = temp / 5;
+    if (pulse_position[7] & 1)
+        pulse_position[3] = 4 - pulse_position[3];
+    pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
+    pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
+
+    fixed_sparse->n = 8;
+    for (i = 0; i < 4; i++) {
+        const int pos1   = (pulse_position[i]     << 2) + i;
+        const int pos2   = (pulse_position[i + 4] << 2) + i;
+        const float sign = fixed_index[i] ? -1.0 : 1.0;
+        fixed_sparse->x[i    ] = pos1;
+        fixed_sparse->x[i + 4] = pos2;
+        fixed_sparse->y[i    ] = sign;
+        fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
+    }
+}
+
+/**
+ * Decode the algebraic codebook index to pulse positions and signs,
+ * then construct the algebraic codebook vector.
+ *
+ *                              nb of pulses | bits encoding pulses
+ * For MODE_4k75 or MODE_5k15,             2 | 1-3, 4-6, 7
+ *                  MODE_5k9,              2 | 1,   2-4, 5-6, 7-9
+ *                  MODE_6k7,              3 | 1-3, 4,   5-7, 8,  9-11
+ *      MODE_7k4 or MODE_7k95,             4 | 1-3, 4-6, 7-9, 10, 11-13
+ *
+ * @param fixed_sparse pointer to the algebraic codebook vector
+ * @param pulses       algebraic codebook indexes
+ * @param mode         mode of the current frame
+ * @param subframe     current subframe number
+ */
+static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
+                                const enum Mode mode, const int subframe)
+{
+    assert(MODE_4k75 <= mode && mode <= MODE_12k2);
+
+    if (mode == MODE_12k2) {
+        ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
+    } else if (mode == MODE_10k2) {
+        decode_8_pulses_31bits(pulses, fixed_sparse);
+    } else {
+        int *pulse_position = fixed_sparse->x;
+        int i, pulse_subset;
+        const int fixed_index = pulses[0];
+
+        if (mode <= MODE_5k15) {
+            pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
+            pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
+            pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
+            fixed_sparse->n = 2;
+        } else if (mode == MODE_5k9) {
+            pulse_subset      = ((fixed_index & 1) << 1) + 1;
+            pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
+            pulse_subset      = (fixed_index  >> 4) & 3;
+            pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
+            fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
+        } else if (mode == MODE_6k7) {
+            pulse_position[0] = (fixed_index        & 7) * 5;
+            pulse_subset      = (fixed_index  >> 2) & 2;
+            pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
+            pulse_subset      = (fixed_index  >> 6) & 2;
+            pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
+            fixed_sparse->n = 3;
+        } else { // mode <= MODE_7k95
+            pulse_position[0] = gray_decode[ fixed_index        & 7];
+            pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
+            pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
+            pulse_subset      = (fixed_index >> 9) & 1;
+            pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
+            fixed_sparse->n = 4;
+        }
+        for (i = 0; i < fixed_sparse->n; i++)
+            fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
+    }
+}
+
+/**
+ * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
+ *
+ * @param p the context
+ * @param subframe unpacked amr subframe
+ * @param mode mode of the current frame
+ * @param fixed_sparse sparse respresentation of the fixed vector
+ */
+static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
+                             AMRFixed *fixed_sparse)
+{
+    // The spec suggests the current pitch gain is always used, but in other
+    // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
+    // so the codebook gain cannot depend on the quantized pitch gain.
+    if (mode == MODE_12k2)
+        p->beta = FFMIN(p->pitch_gain[4], 1.0);
+
+    fixed_sparse->pitch_lag  = p->pitch_lag_int;
+    fixed_sparse->pitch_fac  = p->beta;
+
+    // Save pitch sharpening factor for the next subframe
+    // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
+    // the fact that the gains for two subframes are jointly quantized.
+    if (mode != MODE_4k75 || subframe & 1)
+        p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
+}
+/// @}
+
+
+/// @defgroup amr_gain_decoding AMR gain decoding functions
+/// @{
+
+/**
+ * fixed gain smoothing
+ * Note that where the spec specifies the "spectrum in the q domain"
+ * in section 6.1.4, in fact frequencies should be used.
+ *
+ * @param p the context
+ * @param lsf LSFs for the current subframe, in the range [0,1]
+ * @param lsf_avg averaged LSFs
+ * @param mode mode of the current frame
+ *
+ * @return fixed gain smoothed
+ */
+static float fixed_gain_smooth(AMRContext *p , const float *lsf,
+                               const float *lsf_avg, const enum Mode mode)
+{
+    float diff = 0.0;
+    int i;
+
+    for (i = 0; i < LP_FILTER_ORDER; i++)
+        diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
+
+    // If diff is large for ten subframes, disable smoothing for a 40-subframe
+    // hangover period.
+    p->diff_count++;
+    if (diff <= 0.65)
+        p->diff_count = 0;
+
+    if (p->diff_count > 10) {
+        p->hang_count = 0;
+        p->diff_count--; // don't let diff_count overflow
+    }
+
+    if (p->hang_count < 40) {
+        p->hang_count++;
+    } else if (mode < MODE_7k4 || mode == MODE_10k2) {
+        const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
+        const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
+                                       p->fixed_gain[2] + p->fixed_gain[3] +
+                                       p->fixed_gain[4]) * 0.2;
+        return smoothing_factor * p->fixed_gain[4] +
+               (1.0 - smoothing_factor) * fixed_gain_mean;
+    }
+    return p->fixed_gain[4];
+}
+
+/**
+ * Decode pitch gain and fixed gain factor (part of section 6.1.3).
+ *
+ * @param p the context
+ * @param amr_subframe unpacked amr subframe
+ * @param mode mode of the current frame
+ * @param subframe current subframe number
+ * @param fixed_gain_factor decoded gain correction factor
+ */
+static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
+                         const enum Mode mode, const int subframe,
+                         float *fixed_gain_factor)
+{
+    if (mode == MODE_12k2 || mode == MODE_7k95) {
+        p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
+            * (1.0 / 16384.0);
+        *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
+            * (1.0 /  2048.0);
+    } else {
+        const uint16_t *gains;
+
+        if (mode >= MODE_6k7) {
+            gains = gains_high[amr_subframe->p_gain];
+        } else if (mode >= MODE_5k15) {
+            gains = gains_low [amr_subframe->p_gain];
+        } else {
+            // gain index is only coded in subframes 0,2 for MODE_4k75
+            gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
+        }
+
+        p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
+        *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
+    }
+}
+
+/// @}
+
+
+/// @defgroup amr_pre_processing AMR pre-processing functions
+/// @{
+
+/**
+ * Circularly convolve a sparse fixed vector with a phase dispersion impulse
+ * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
+ *
+ * @param out vector with filter applied
+ * @param in source vector
+ * @param filter phase filter coefficients
+ *
+ *  out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
+ */
+static void apply_ir_filter(float *out, const AMRFixed *in,
+                            const float *filter)
+{
+    float filter1[AMR_SUBFRAME_SIZE],     //!< filters at pitch lag*1 and *2
+          filter2[AMR_SUBFRAME_SIZE];
+    int   lag = in->pitch_lag;
+    float fac = in->pitch_fac;
+    int i;
+
+    if (lag < AMR_SUBFRAME_SIZE) {
+        ff_celp_circ_addf(filter1, filter, filter, lag, fac,
+                          AMR_SUBFRAME_SIZE);
+
+        if (lag < AMR_SUBFRAME_SIZE >> 1)
+            ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
+                              AMR_SUBFRAME_SIZE);
+    }
+
+    memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
+    for (i = 0; i < in->n; i++) {
+        int   x = in->x[i];
+        float y = in->y[i];
+        const float *filterp;
+
+        if (x >= AMR_SUBFRAME_SIZE - lag) {
+            filterp = filter;
+        } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
+            filterp = filter1;
+        } else
+            filterp = filter2;
+
+        ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
+    }
+}
+
+/**
+ * Reduce fixed vector sparseness by smoothing with one of three IR filters.
+ * Also know as "adaptive phase dispersion".
+ *
+ * This implements 3GPP TS 26.090 section 6.1(5).
+ *
+ * @param p the context
+ * @param fixed_sparse algebraic codebook vector
+ * @param fixed_vector unfiltered fixed vector
+ * @param fixed_gain smoothed gain
+ * @param out space for modified vector if necessary
+ */
+static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
+                                    const float *fixed_vector,
+                                    float fixed_gain, float *out)
+{
+    int ir_filter_nr;
+
+    if (p->pitch_gain[4] < 0.6) {
+        ir_filter_nr = 0;      // strong filtering
+    } else if (p->pitch_gain[4] < 0.9) {
+        ir_filter_nr = 1;      // medium filtering
+    } else
+        ir_filter_nr = 2;      // no filtering
+
+    // detect 'onset'
+    if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
+        p->ir_filter_onset = 2;
+    } else if (p->ir_filter_onset)
+        p->ir_filter_onset--;
+
+    if (!p->ir_filter_onset) {
+        int i, count = 0;
+
+        for (i = 0; i < 5; i++)
+            if (p->pitch_gain[i] < 0.6)
+                count++;
+        if (count > 2)
+            ir_filter_nr = 0;
+
+        if (ir_filter_nr > p->prev_ir_filter_nr + 1)
+            ir_filter_nr--;
+    } else if (ir_filter_nr < 2)
+        ir_filter_nr++;
+
+    // Disable filtering for very low level of fixed_gain.
+    // Note this step is not specified in the technical description but is in
+    // the reference source in the function Ph_disp.
+    if (fixed_gain < 5.0)
+        ir_filter_nr = 2;
+
+    if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
+         && ir_filter_nr < 2) {
+        apply_ir_filter(out, fixed_sparse,
+                        (p->cur_frame_mode == MODE_7k95 ?
+                             ir_filters_lookup_MODE_7k95 :
+                             ir_filters_lookup)[ir_filter_nr]);
+        fixed_vector = out;
+    }
+
+    // update ir filter strength history
+    p->prev_ir_filter_nr       = ir_filter_nr;
+    p->prev_sparse_fixed_gain  = fixed_gain;
+
+    return fixed_vector;
+}
+
+/// @}
+
+
+/// @defgroup amr_synthesis AMR synthesis functions
+/// @{
+
+/**
+ * Conduct 10th order linear predictive coding synthesis.
+ *
+ * @param p             pointer to the AMRContext
+ * @param lpc           pointer to the LPC coefficients
+ * @param fixed_gain    fixed codebook gain for synthesis
+ * @param fixed_vector  algebraic codebook vector
+ * @param samples       pointer to the output speech samples
+ * @param overflow      16-bit overflow flag
+ */
+static int synthesis(AMRContext *p, float *lpc,
+                     float fixed_gain, const float *fixed_vector,
+                     float *samples, uint8_t overflow)
+{
+    int i, overflow_temp = 0;
+    float excitation[AMR_SUBFRAME_SIZE];
+
+    // if an overflow has been detected, the pitch vector is scaled down by a
+    // factor of 4
+    if (overflow)
+        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
+            p->pitch_vector[i] *= 0.25;
+
+    ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
+                            p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
+
+    // emphasize pitch vector contribution
+    if (p->pitch_gain[4] > 0.5 && !overflow) {
+        float energy = ff_dot_productf(excitation, excitation,
+                                       AMR_SUBFRAME_SIZE);
+        float pitch_factor =
+            p->pitch_gain[4] *
+            (p->cur_frame_mode == MODE_12k2 ?
+                0.25 * FFMIN(p->pitch_gain[4], 1.0) :
+                0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));
+
+        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
+            excitation[i] += pitch_factor * p->pitch_vector[i];
+
+        ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
+                                                AMR_SUBFRAME_SIZE);
+    }
+
+    ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
+                                 LP_FILTER_ORDER);
+
+    // detect overflow
+    for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
+        if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
+            overflow_temp = 1;
+            samples[i] = av_clipf(samples[i], -AMR_SAMPLE_BOUND,
+                                               AMR_SAMPLE_BOUND);
+        }
+
+    return overflow_temp;
+}
+
+/// @}
+
+
+/// @defgroup amr_update AMR update functions
+/// @{
+
+/**
+ * Update buffers and history at the end of decoding a subframe.
+ *
+ * @param p             pointer to the AMRContext
+ */
+static void update_state(AMRContext *p)
+{
+    memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
+
+    memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
+            (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
+
+    memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
+    memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
+
+    memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
+            LP_FILTER_ORDER * sizeof(float));
+}
+
+/// @}
+
+
+/// @defgroup amr_postproc AMR Post processing functions
+/// @{
+
+/**
+ * Get the tilt factor of a formant filter from its transfer function
+ *
+ * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
+ * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
+ */
+static float tilt_factor(float *lpc_n, float *lpc_d)
+{
+    float rh0, rh1; // autocorrelation at lag 0 and 1
+
+    // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
+    float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
+    float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
+
+    hf[0] = 1.0;
+    memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
+    ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
+                                 LP_FILTER_ORDER);
+
+    rh0 = ff_dot_productf(hf, hf,     AMR_TILT_RESPONSE);
+    rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
+
+    // The spec only specifies this check for 12.2 and 10.2 kbit/s
+    // modes. But in the ref source the tilt is always non-negative.
+    return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
+}
+
+/**
+ * Perform adaptive post-filtering to enhance the quality of the speech.
+ * See section 6.2.1.
+ *
+ * @param p             pointer to the AMRContext
+ * @param lpc           interpolated LP coefficients for this subframe
+ * @param buf_out       output of the filter
+ */
+static void postfilter(AMRContext *p, float *lpc, float *buf_out)
+{
+    int i;
+    float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
+
+    float speech_gain       = ff_dot_productf(samples, samples,
+                                              AMR_SUBFRAME_SIZE);
+
+    float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
+    const float *gamma_n, *gamma_d;                       // Formant filter factor table
+    float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
+
+    if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
+        gamma_n = ff_pow_0_7;
+        gamma_d = ff_pow_0_75;
+    } else {
+        gamma_n = ff_pow_0_55;
+        gamma_d = ff_pow_0_7;
+    }
+
+    for (i = 0; i < LP_FILTER_ORDER; i++) {
+         lpc_n[i] = lpc[i] * gamma_n[i];
+         lpc_d[i] = lpc[i] * gamma_d[i];
+    }
+
+    memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
+    ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
+                                 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
+    memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
+           sizeof(float) * LP_FILTER_ORDER);
+
+    ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
+                                      pole_out + LP_FILTER_ORDER,
+                                      AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
+
+    ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
+                         AMR_SUBFRAME_SIZE);
+
+    ff_adaptative_gain_control(buf_out, speech_gain, AMR_SUBFRAME_SIZE,
+                               AMR_AGC_ALPHA, &p->postfilter_agc);
+}
+
+/// @}
+
+static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
+                              AVPacket *avpkt)
+{
+
+    AMRContext *p = avctx->priv_data;        // pointer to private data
+    const uint8_t *buf = avpkt->data;
+    int buf_size       = avpkt->size;
+    float *buf_out = data;                   // pointer to the output data buffer
+    int i, subframe;
+    float fixed_gain_factor;
+    AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
+    float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
+    float synth_fixed_gain;                  // the fixed gain that synthesis should use
+    const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
+
+    p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
+    if (p->cur_frame_mode == MODE_DTX) {
+        av_log_missing_feature(avctx, "dtx mode", 1);
+        return -1;
+    }
+
+    if (p->cur_frame_mode == MODE_12k2) {
+        lsf2lsp_5(p);
+    } else
+        lsf2lsp_3(p);
+
+    for (i = 0; i < 4; i++)
+        ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
+
+    for (subframe = 0; subframe < 4; subframe++) {
+        const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
+
+        decode_pitch_vector(p, amr_subframe, subframe);
+
+        decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
+                            p->cur_frame_mode, subframe);
+
+        // The fixed gain (section 6.1.3) depends on the fixed vector
+        // (section 6.1.2), but the fixed vector calculation uses
+        // pitch sharpening based on the on the pitch gain (section 6.1.3).
+        // So the correct order is: pitch gain, pitch sharpening, fixed gain.
+        decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
+                     &fixed_gain_factor);
+
+        pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
+
+        ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
+                            AMR_SUBFRAME_SIZE);
+
+        p->fixed_gain[4] =
+            ff_amr_set_fixed_gain(fixed_gain_factor,
+                       ff_dot_productf(p->fixed_vector, p->fixed_vector,
+                                       AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
+                       p->prediction_error,
+                       energy_mean[p->cur_frame_mode], energy_pred_fac);
+
+        // The excitation feedback is calculated without any processing such
+        // as fixed gain smoothing. This isn't mentioned in the specification.
+        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
+            p->excitation[i] *= p->pitch_gain[4];
+        ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
+                            AMR_SUBFRAME_SIZE);
+
+        // In the ref decoder, excitation is stored with no fractional bits.
+        // This step prevents buzz in silent periods. The ref encoder can
+        // emit long sequences with pitch factor greater than one. This
+        // creates unwanted feedback if the excitation vector is nonzero.
+        // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
+        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
+            p->excitation[i] = truncf(p->excitation[i]);
+
+        // Smooth fixed gain.
+        // The specification is ambiguous, but in the reference source, the
+        // smoothed value is NOT fed back into later fixed gain smoothing.
+        synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
+                                             p->lsf_avg, p->cur_frame_mode);
+
+        synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
+                                             synth_fixed_gain, spare_vector);
+
+        if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
+                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
+            // overflow detected -> rerun synthesis scaling pitch vector down
+            // by a factor of 4, skipping pitch vector contribution emphasis
+            // and adaptive gain control
+            synthesis(p, p->lpc[subframe], synth_fixed_gain,
+                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
+
+        postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
+
+        // update buffers and history
+        ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
+        update_state(p);
+    }
+
+    ff_acelp_apply_order_2_transfer_function(buf_out, highpass_zeros,
+                                             highpass_poles, highpass_gain,
+                                             p->high_pass_mem, AMR_BLOCK_SIZE);
+
+    for (i = 0; i < AMR_BLOCK_SIZE; i++)
+        buf_out[i] = av_clipf(buf_out[i] * AMR_SAMPLE_SCALE,
+                              -1.0, 32767.0 / 32768.0);
+
+    /* Update averaged lsf vector (used for fixed gain smoothing).
+     *
+     * Note that lsf_avg should not incorporate the current frame's LSFs
+     * for fixed_gain_smooth.
+     * The specification has an incorrect formula: the reference decoder uses
+     * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
+    ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
+                            0.84, 0.16, LP_FILTER_ORDER);
+
+    /* report how many samples we got */
+    *data_size = AMR_BLOCK_SIZE * sizeof(float);
+
+    /* return the amount of bytes consumed if everything was OK */
+    return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
+}
+
+
+AVCodec amrnb_decoder = {
+    .name           = "amrnb",
+    .type           = CODEC_TYPE_AUDIO,
+    .id             = CODEC_ID_AMR_NB,
+    .priv_data_size = sizeof(AMRContext),
+    .init           = amrnb_decode_init,
+    .decode         = amrnb_decode_frame,
+    .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
+    .sample_fmts    = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
+};