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a247ac640d
Given that the AVCodec.next pointer has now been removed, most of the AVCodecs are not modified at all any more and can therefore be made const (as this patch does); the only exceptions are the very few codecs for external libraries that have a init_static_data callback. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com> Signed-off-by: James Almer <jamrial@gmail.com>
576 lines
19 KiB
C
576 lines
19 KiB
C
/*
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* SIPR / ACELP.NET decoder
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*
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* Copyright (c) 2008 Vladimir Voroshilov
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* Copyright (c) 2009 Vitor Sessak
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <math.h>
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#include <stdint.h>
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#include <string.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/mathematics.h"
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#define BITSTREAM_READER_LE
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#include "avcodec.h"
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#include "get_bits.h"
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#include "internal.h"
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#include "lsp.h"
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#include "acelp_vectors.h"
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#include "acelp_pitch_delay.h"
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#include "acelp_filters.h"
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#include "celp_filters.h"
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#define MAX_SUBFRAME_COUNT 5
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#include "sipr.h"
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#include "siprdata.h"
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typedef struct SiprModeParam {
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const char *mode_name;
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uint16_t bits_per_frame;
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uint8_t subframe_count;
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uint8_t frames_per_packet;
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float pitch_sharp_factor;
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/* bitstream parameters */
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uint8_t number_of_fc_indexes;
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uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor
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/** size in bits of the i-th stage vector of quantizer */
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uint8_t vq_indexes_bits[5];
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/** size in bits of the adaptive-codebook index for every subframe */
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uint8_t pitch_delay_bits[5];
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uint8_t gp_index_bits;
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uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
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uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes
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} SiprModeParam;
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static const SiprModeParam modes[MODE_COUNT] = {
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[MODE_16k] = {
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.mode_name = "16k",
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.bits_per_frame = 160,
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.subframe_count = SUBFRAME_COUNT_16k,
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.frames_per_packet = 1,
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.pitch_sharp_factor = 0.00,
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.number_of_fc_indexes = 10,
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.ma_predictor_bits = 1,
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.vq_indexes_bits = {7, 8, 7, 7, 7},
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.pitch_delay_bits = {9, 6},
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.gp_index_bits = 4,
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.fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
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.gc_index_bits = 5
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},
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[MODE_8k5] = {
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.mode_name = "8k5",
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.bits_per_frame = 152,
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.subframe_count = 3,
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.frames_per_packet = 1,
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.pitch_sharp_factor = 0.8,
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.number_of_fc_indexes = 3,
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.ma_predictor_bits = 0,
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.vq_indexes_bits = {6, 7, 7, 7, 5},
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.pitch_delay_bits = {8, 5, 5},
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.gp_index_bits = 0,
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.fc_index_bits = {9, 9, 9},
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.gc_index_bits = 7
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},
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[MODE_6k5] = {
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.mode_name = "6k5",
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.bits_per_frame = 232,
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.subframe_count = 3,
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.frames_per_packet = 2,
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.pitch_sharp_factor = 0.8,
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.number_of_fc_indexes = 3,
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.ma_predictor_bits = 0,
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.vq_indexes_bits = {6, 7, 7, 7, 5},
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.pitch_delay_bits = {8, 5, 5},
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.gp_index_bits = 0,
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.fc_index_bits = {5, 5, 5},
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.gc_index_bits = 7
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},
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[MODE_5k0] = {
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.mode_name = "5k0",
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.bits_per_frame = 296,
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.subframe_count = 5,
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.frames_per_packet = 2,
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.pitch_sharp_factor = 0.85,
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.number_of_fc_indexes = 1,
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.ma_predictor_bits = 0,
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.vq_indexes_bits = {6, 7, 7, 7, 5},
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.pitch_delay_bits = {8, 5, 8, 5, 5},
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.gp_index_bits = 0,
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.fc_index_bits = {10},
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.gc_index_bits = 7
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}
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};
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const float ff_pow_0_5[] = {
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1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4),
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1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8),
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1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
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1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
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};
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static void dequant(float *out, const int *idx, const float * const cbs[])
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{
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int i;
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int stride = 2;
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int num_vec = 5;
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for (i = 0; i < num_vec; i++)
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memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
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}
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static void lsf_decode_fp(float *lsfnew, float *lsf_history,
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const SiprParameters *parm)
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{
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int i;
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float lsf_tmp[LP_FILTER_ORDER];
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dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
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for (i = 0; i < LP_FILTER_ORDER; i++)
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lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
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ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1);
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/* Note that a minimum distance is not enforced between the last value and
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the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
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ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1);
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lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
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memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
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for (i = 0; i < LP_FILTER_ORDER - 1; i++)
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lsfnew[i] = cos(lsfnew[i]);
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lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
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}
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/** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
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static void pitch_sharpening(int pitch_lag_int, float beta,
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float *fixed_vector)
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{
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int i;
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for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
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fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
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}
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/**
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* Extract decoding parameters from the input bitstream.
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* @param parms parameters structure
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* @param pgb pointer to initialized GetBitContext structure
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*/
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static void decode_parameters(SiprParameters* parms, GetBitContext *pgb,
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const SiprModeParam *p)
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{
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int i, j;
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if (p->ma_predictor_bits)
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parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits);
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for (i = 0; i < 5; i++)
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parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]);
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for (i = 0; i < p->subframe_count; i++) {
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parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]);
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if (p->gp_index_bits)
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parms->gp_index[i] = get_bits(pgb, p->gp_index_bits);
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for (j = 0; j < p->number_of_fc_indexes; j++)
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parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
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parms->gc_index[i] = get_bits(pgb, p->gc_index_bits);
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}
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}
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static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
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int num_subfr)
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{
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double lsfint[LP_FILTER_ORDER];
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int i,j;
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float t, t0 = 1.0 / num_subfr;
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t = t0 * 0.5;
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for (i = 0; i < num_subfr; i++) {
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for (j = 0; j < LP_FILTER_ORDER; j++)
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lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
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ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
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Az += LP_FILTER_ORDER;
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t += t0;
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}
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}
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/**
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* Evaluate the adaptive impulse response.
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*/
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static void eval_ir(const float *Az, int pitch_lag, float *freq,
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float pitch_sharp_factor)
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{
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float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
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int i;
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tmp1[0] = 1.0;
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for (i = 0; i < LP_FILTER_ORDER; i++) {
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tmp1[i+1] = Az[i] * ff_pow_0_55[i];
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tmp2[i ] = Az[i] * ff_pow_0_7 [i];
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}
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memset(tmp1 + 11, 0, 37 * sizeof(float));
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ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE,
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LP_FILTER_ORDER);
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pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
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}
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/**
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* Evaluate the convolution of a vector with a sparse vector.
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*/
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static void convolute_with_sparse(float *out, const AMRFixed *pulses,
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const float *shape, int length)
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{
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int i, j;
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memset(out, 0, length*sizeof(float));
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for (i = 0; i < pulses->n; i++)
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for (j = pulses->x[i]; j < length; j++)
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out[j] += pulses->y[i] * shape[j - pulses->x[i]];
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}
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/**
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* Apply postfilter, very similar to AMR one.
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*/
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static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
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{
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float buf[SUBFR_SIZE + LP_FILTER_ORDER];
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float *pole_out = buf + LP_FILTER_ORDER;
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float lpc_n[LP_FILTER_ORDER];
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float lpc_d[LP_FILTER_ORDER];
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int i;
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for (i = 0; i < LP_FILTER_ORDER; i++) {
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lpc_d[i] = lpc[i] * ff_pow_0_75[i];
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lpc_n[i] = lpc[i] * ff_pow_0_5 [i];
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};
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memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
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LP_FILTER_ORDER*sizeof(float));
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ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE,
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LP_FILTER_ORDER);
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memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
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LP_FILTER_ORDER*sizeof(float));
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ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
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memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
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LP_FILTER_ORDER*sizeof(*pole_out));
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memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
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LP_FILTER_ORDER*sizeof(*pole_out));
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ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE,
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LP_FILTER_ORDER);
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}
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static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
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SiprMode mode, int low_gain)
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{
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int i;
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switch (mode) {
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case MODE_6k5:
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for (i = 0; i < 3; i++) {
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fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
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fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
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}
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fixed_sparse->n = 3;
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break;
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case MODE_8k5:
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for (i = 0; i < 3; i++) {
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fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
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fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i;
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fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0;
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fixed_sparse->y[2*i + 1] =
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(fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
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-fixed_sparse->y[2*i ] : fixed_sparse->y[2*i];
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}
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fixed_sparse->n = 6;
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break;
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case MODE_5k0:
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default:
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if (low_gain) {
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int offset = (pulses[0] & 0x200) ? 2 : 0;
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int val = pulses[0];
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for (i = 0; i < 3; i++) {
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int index = (val & 0x7) * 6 + 4 - i*2;
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fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
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fixed_sparse->x[i] = index;
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val >>= 3;
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}
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fixed_sparse->n = 3;
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} else {
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int pulse_subset = (pulses[0] >> 8) & 1;
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fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
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fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1;
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fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
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fixed_sparse->y[1] = -fixed_sparse->y[0];
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fixed_sparse->n = 2;
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}
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break;
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}
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}
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static void decode_frame(SiprContext *ctx, SiprParameters *params,
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float *out_data)
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{
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int i, j;
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int subframe_count = modes[ctx->mode].subframe_count;
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int frame_size = subframe_count * SUBFR_SIZE;
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float Az[LP_FILTER_ORDER * MAX_SUBFRAME_COUNT];
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float *excitation;
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float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
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float lsf_new[LP_FILTER_ORDER];
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float *impulse_response = ir_buf + LP_FILTER_ORDER;
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float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
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// memory alignment
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int t0_first = 0;
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AMRFixed fixed_cb;
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memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
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lsf_decode_fp(lsf_new, ctx->lsf_history, params);
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sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count);
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memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
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excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
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for (i = 0; i < subframe_count; i++) {
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float *pAz = Az + i*LP_FILTER_ORDER;
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float fixed_vector[SUBFR_SIZE];
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int T0,T0_frac;
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float pitch_gain, gain_code, avg_energy;
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ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
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ctx->mode == MODE_5k0, 6);
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if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
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t0_first = T0;
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ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
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ff_b60_sinc, 6,
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2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
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SUBFR_SIZE);
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decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
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ctx->past_pitch_gain < 0.8);
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eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor);
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convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
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SUBFR_SIZE);
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avg_energy = (0.01 + avpriv_scalarproduct_float_c(fixed_vector,
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fixed_vector,
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SUBFR_SIZE)) /
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SUBFR_SIZE;
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ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
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gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
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avg_energy, ctx->energy_history,
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34 - 15.0/(0.05*M_LN10/M_LN2),
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pred);
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ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
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pitch_gain, gain_code, SUBFR_SIZE);
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pitch_gain *= 0.5 * pitch_gain;
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pitch_gain = FFMIN(pitch_gain, 0.4);
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ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
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ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
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gain_code *= ctx->gain_mem;
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for (j = 0; j < SUBFR_SIZE; j++)
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fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
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if (ctx->mode == MODE_5k0) {
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postfilter_5k0(ctx, pAz, fixed_vector);
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ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
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pAz, excitation, SUBFR_SIZE,
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LP_FILTER_ORDER);
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}
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ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
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SUBFR_SIZE, LP_FILTER_ORDER);
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excitation += SUBFR_SIZE;
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}
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|
memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
|
|
LP_FILTER_ORDER * sizeof(float));
|
|
|
|
if (ctx->mode == MODE_5k0) {
|
|
for (i = 0; i < subframe_count; i++) {
|
|
float energy = avpriv_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
|
|
ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
|
|
SUBFR_SIZE);
|
|
ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
|
|
&synth[i * SUBFR_SIZE], energy,
|
|
SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
|
|
}
|
|
|
|
memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
|
|
LP_FILTER_ORDER*sizeof(float));
|
|
}
|
|
memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
|
|
(PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
|
|
|
|
ff_acelp_apply_order_2_transfer_function(out_data, synth,
|
|
(const float[2]) {-1.99997 , 1.000000000},
|
|
(const float[2]) {-1.93307352, 0.935891986},
|
|
0.939805806,
|
|
ctx->highpass_filt_mem,
|
|
frame_size);
|
|
}
|
|
|
|
static av_cold int sipr_decoder_init(AVCodecContext * avctx)
|
|
{
|
|
SiprContext *ctx = avctx->priv_data;
|
|
int i;
|
|
|
|
switch (avctx->block_align) {
|
|
case 20: ctx->mode = MODE_16k; break;
|
|
case 19: ctx->mode = MODE_8k5; break;
|
|
case 29: ctx->mode = MODE_6k5; break;
|
|
case 37: ctx->mode = MODE_5k0; break;
|
|
default:
|
|
if (avctx->bit_rate > 12200) ctx->mode = MODE_16k;
|
|
else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5;
|
|
else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5;
|
|
else ctx->mode = MODE_5k0;
|
|
av_log(avctx, AV_LOG_WARNING,
|
|
"Invalid block_align: %d. Mode %s guessed based on bitrate: %"PRId64"\n",
|
|
avctx->block_align, modes[ctx->mode].mode_name, avctx->bit_rate);
|
|
}
|
|
|
|
av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
|
|
|
|
if (ctx->mode == MODE_16k) {
|
|
ff_sipr_init_16k(ctx);
|
|
ctx->decode_frame = ff_sipr_decode_frame_16k;
|
|
} else {
|
|
ctx->decode_frame = decode_frame;
|
|
}
|
|
|
|
for (i = 0; i < LP_FILTER_ORDER; i++)
|
|
ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
|
|
|
|
for (i = 0; i < 4; i++)
|
|
ctx->energy_history[i] = -14;
|
|
|
|
avctx->channels = 1;
|
|
avctx->channel_layout = AV_CH_LAYOUT_MONO;
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int sipr_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
SiprContext *ctx = avctx->priv_data;
|
|
AVFrame *frame = data;
|
|
const uint8_t *buf=avpkt->data;
|
|
SiprParameters parm;
|
|
const SiprModeParam *mode_par = &modes[ctx->mode];
|
|
GetBitContext gb;
|
|
float *samples;
|
|
int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
|
|
int i, ret;
|
|
|
|
ctx->avctx = avctx;
|
|
if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"Error processing packet: packet size (%d) too small\n",
|
|
avpkt->size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
/* get output buffer */
|
|
frame->nb_samples = mode_par->frames_per_packet * subframe_size *
|
|
mode_par->subframe_count;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
samples = (float *)frame->data[0];
|
|
|
|
init_get_bits(&gb, buf, mode_par->bits_per_frame);
|
|
|
|
for (i = 0; i < mode_par->frames_per_packet; i++) {
|
|
decode_parameters(&parm, &gb, mode_par);
|
|
|
|
ctx->decode_frame(ctx, &parm, samples);
|
|
|
|
samples += subframe_size * mode_par->subframe_count;
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return mode_par->bits_per_frame >> 3;
|
|
}
|
|
|
|
const AVCodec ff_sipr_decoder = {
|
|
.name = "sipr",
|
|
.long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_SIPR,
|
|
.priv_data_size = sizeof(SiprContext),
|
|
.init = sipr_decoder_init,
|
|
.decode = sipr_decode_frame,
|
|
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
|
|
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
|
|
};
|