Mercurial > libavcodec.hg
diff amrnbdec.c @ 11235:5c339e441ace libavcodec
AMR-NB floating-point based decoder.
Code produced during SoC by Robert Swain and Colin McQuillan.
| author | vitor |
|---|---|
| date | Sun, 21 Feb 2010 18:01:56 +0000 |
| parents | |
| children | c2e19a511e26 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/amrnbdec.c Sun Feb 21 18:01:56 2010 +0000 @@ -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}, +};