/* * 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 * 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 "libavutil/channel_layout.h"#include "avcodec.h"#include "libavutil/common.h"#include "libavutil/avassert.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 "amr.h"#include "codec_internal.h"#include "decode.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     ACELPFContext                     acelpf_ctx; ///< context for filters for ACELP-based codecs    ACELPVContext                     acelpv_ctx; ///< context for vector operations for ACELP-based codecs    CELPFContext                       celpf_ctx; ///< context for filters for CELP-based codecs    CELPMContext                       celpm_ctx; ///< context for fixed point math operations } AMRContext; typedef struct AMRChannelsContext {    AMRContext ch[2];} AMRChannelsContext; /** 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){    AMRChannelsContext *s = avctx->priv_data;    int i;     if (avctx->ch_layout.nb_channels > 2) {        avpriv_report_missing_feature(avctx, ">2 channel AMR");        return AVERROR_PATCHWELCOME;    }     if (!avctx->ch_layout.nb_channels) {        av_channel_layout_uninit(&avctx->ch_layout);        avctx->ch_layout      = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;    }    if (!avctx->sample_rate)        avctx->sample_rate = 8000;    avctx->sample_fmt     = AV_SAMPLE_FMT_FLTP;     for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {        AMRContext *p = &s->ch[ch];        // 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;         ff_acelp_filter_init(&p->acelpf_ctx);        ff_acelp_vectors_init(&p->acelpv_ctx);        ff_celp_filter_init(&p->celpf_ctx);        ff_celp_math_init(&p->celpm_ctx);    }     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){    enum Mode mode;     // Decode the first octet.    mode = buf[0] >> 3 & 0x0F;                      // frame type    p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit     if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {        return NO_DATA;    }     if (mode < MODE_DTX)        ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,                           amr_unpacking_bitmaps_per_mode[mode]);     return mode;}  /// @name AMR pitch LPC coefficient decoding functions/// @{ /** * Interpolate the LSF vector (used for fixed gain smoothing). * The interpolation is done over all four subframes even in MODE_12k2. * * @param[in]     ctx       The Context * @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(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new){    int i;     for (i = 0; i < 4; i++)        ctx->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(*lsf_r));     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->acelpv_ctx, p->lsf_q, lsf_q);     ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);} /** * 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->acelpv_ctx, p->lsf_q, lsf_q);    memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));     ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);     // 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;} /// @}  /// @name 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));        pitch_lag_frac *= 2;    }     p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t     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.   */    p->acelpf_ctx.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));} /// @}  /// @name 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

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