Mercurial > libavcodec.hg
view amrnbdec.c @ 12138:41f078d6869d libavcodec
Do not try to decode less than one frame of data in TrueSpeech decoder.
This should solve issue 2085.
| author | kostya |
|---|---|
| date | Sun, 11 Jul 2010 14:50:20 +0000 |
| parents | ceec2fb08b8e |
| children | 2dd67ed2f947 |
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/* * 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 "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; 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) { return 1; } return 0; } /// @} /// @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_adaptive_gain_control(buf_out, 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, buf_out, highpass_zeros, highpass_poles, highpass_gain * AMR_SAMPLE_SCALE, p->high_pass_mem, AMR_BLOCK_SIZE); /* 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 = AVMEDIA_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}, };