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08bebeb1be
Some callers assume that item_name is always set, so this may be
considered an API break.
This reverts commit 0c6203c97a.
1128 lines
37 KiB
C
1128 lines
37 KiB
C
/*
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* G.723.1 compatible decoder
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* Copyright (c) 2006 Benjamin Larsson
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* Copyright (c) 2010 Mohamed Naufal Basheer
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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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/**
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* @file
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* G.723.1 compatible decoder
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#define BITSTREAM_READER_LE
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#include "acelp_vectors.h"
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#include "avcodec.h"
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#include "celp_filters.h"
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#include "codec_internal.h"
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#include "decode.h"
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#include "get_bits.h"
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#include "g723_1.h"
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#define CNG_RANDOM_SEED 12345
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/**
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* Postfilter gain weighting factors scaled by 2^15
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*/
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static const int16_t ppf_gain_weight[2] = {0x1800, 0x2000};
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static const int16_t pitch_contrib[340] = {
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60, 0, 0, 2489, 60, 0, 0, 5217,
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1, 6171, 0, 3953, 0, 10364, 1, 9357,
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-1, 8843, 1, 9396, 0, 5794, -1, 10816,
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2, 11606, -2, 12072, 0, 8616, 1, 12170,
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0, 14440, 0, 7787, -1, 13721, 0, 18205,
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0, 14471, 0, 15807, 1, 15275, 0, 13480,
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-1, 18375, -1, 0, 1, 11194, -1, 13010,
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1, 18836, -2, 20354, 1, 16233, -1, 0,
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60, 0, 0, 12130, 0, 13385, 1, 17834,
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1, 20875, 0, 21996, 1, 0, 1, 18277,
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-1, 21321, 1, 13738, -1, 19094, -1, 20387,
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-1, 0, 0, 21008, 60, 0, -2, 22807,
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0, 15900, 1, 0, 0, 17989, -1, 22259,
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1, 24395, 1, 23138, 0, 23948, 1, 22997,
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2, 22604, -1, 25942, 0, 26246, 1, 25321,
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0, 26423, 0, 24061, 0, 27247, 60, 0,
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-1, 25572, 1, 23918, 1, 25930, 2, 26408,
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-1, 19049, 1, 27357, -1, 24538, 60, 0,
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-1, 25093, 0, 28549, 1, 0, 0, 22793,
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-1, 25659, 0, 29377, 0, 30276, 0, 26198,
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1, 22521, -1, 28919, 0, 27384, 1, 30162,
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-1, 0, 0, 24237, -1, 30062, 0, 21763,
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1, 30917, 60, 0, 0, 31284, 0, 29433,
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1, 26821, 1, 28655, 0, 31327, 2, 30799,
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1, 31389, 0, 32322, 1, 31760, -2, 31830,
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0, 26936, -1, 31180, 1, 30875, 0, 27873,
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-1, 30429, 1, 31050, 0, 0, 0, 31912,
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1, 31611, 0, 31565, 0, 25557, 0, 31357,
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60, 0, 1, 29536, 1, 28985, -1, 26984,
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-1, 31587, 2, 30836, -2, 31133, 0, 30243,
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-1, 30742, -1, 32090, 60, 0, 2, 30902,
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60, 0, 0, 30027, 0, 29042, 60, 0,
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0, 31756, 0, 24553, 0, 25636, -2, 30501,
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60, 0, -1, 29617, 0, 30649, 60, 0,
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0, 29274, 2, 30415, 0, 27480, 0, 31213,
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-1, 28147, 0, 30600, 1, 31652, 2, 29068,
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60, 0, 1, 28571, 1, 28730, 1, 31422,
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0, 28257, 0, 24797, 60, 0, 0, 0,
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60, 0, 0, 22105, 0, 27852, 60, 0,
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60, 0, -1, 24214, 0, 24642, 0, 23305,
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60, 0, 60, 0, 1, 22883, 0, 21601,
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60, 0, 2, 25650, 60, 0, -2, 31253,
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-2, 25144, 0, 17998
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};
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/**
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* Size of the MP-MLQ fixed excitation codebooks
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*/
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static const int32_t max_pos[4] = {593775, 142506, 593775, 142506};
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/**
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* 0.65^i (Zero part) and 0.75^i (Pole part) scaled by 2^15
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*/
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static const int16_t postfilter_tbl[2][LPC_ORDER] = {
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/* Zero */
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{21299, 13844, 8999, 5849, 3802, 2471, 1606, 1044, 679, 441},
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/* Pole */
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{24576, 18432, 13824, 10368, 7776, 5832, 4374, 3281, 2460, 1845}
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};
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static const int cng_adaptive_cb_lag[4] = { 1, 0, 1, 3 };
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static const int cng_filt[4] = { 273, 998, 499, 333 };
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static const int cng_bseg[3] = { 2048, 18432, 231233 };
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static av_cold int g723_1_decode_init(AVCodecContext *avctx)
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{
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G723_1_Context *s = avctx->priv_data;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
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if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n",
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avctx->ch_layout.nb_channels);
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return AVERROR(EINVAL);
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}
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for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
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G723_1_ChannelContext *p = &s->ch[ch];
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p->pf_gain = 1 << 12;
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memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
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p->cng_random_seed = CNG_RANDOM_SEED;
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p->past_frame_type = SID_FRAME;
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}
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return 0;
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}
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/**
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* Unpack the frame into parameters.
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*
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* @param p the context
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* @param buf pointer to the input buffer
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* @param buf_size size of the input buffer
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*/
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static int unpack_bitstream(G723_1_ChannelContext *p, const uint8_t *buf,
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int buf_size)
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{
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GetBitContext gb;
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int ad_cb_len;
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int temp, info_bits, i;
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int ret;
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ret = init_get_bits8(&gb, buf, buf_size);
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if (ret < 0)
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return ret;
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/* Extract frame type and rate info */
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info_bits = get_bits(&gb, 2);
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if (info_bits == 3) {
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p->cur_frame_type = UNTRANSMITTED_FRAME;
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return 0;
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}
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/* Extract 24 bit lsp indices, 8 bit for each band */
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p->lsp_index[2] = get_bits(&gb, 8);
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p->lsp_index[1] = get_bits(&gb, 8);
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p->lsp_index[0] = get_bits(&gb, 8);
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if (info_bits == 2) {
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p->cur_frame_type = SID_FRAME;
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p->subframe[0].amp_index = get_bits(&gb, 6);
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return 0;
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}
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/* Extract the info common to both rates */
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p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
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p->cur_frame_type = ACTIVE_FRAME;
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p->pitch_lag[0] = get_bits(&gb, 7);
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if (p->pitch_lag[0] > 123) /* test if forbidden code */
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return -1;
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p->pitch_lag[0] += PITCH_MIN;
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p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
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p->pitch_lag[1] = get_bits(&gb, 7);
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if (p->pitch_lag[1] > 123)
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return -1;
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p->pitch_lag[1] += PITCH_MIN;
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p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
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p->subframe[0].ad_cb_lag = 1;
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p->subframe[2].ad_cb_lag = 1;
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for (i = 0; i < SUBFRAMES; i++) {
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/* Extract combined gain */
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temp = get_bits(&gb, 12);
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ad_cb_len = 170;
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p->subframe[i].dirac_train = 0;
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if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
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p->subframe[i].dirac_train = temp >> 11;
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temp &= 0x7FF;
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ad_cb_len = 85;
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}
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p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
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if (p->subframe[i].ad_cb_gain < ad_cb_len) {
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p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
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GAIN_LEVELS;
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} else {
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return -1;
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}
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}
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p->subframe[0].grid_index = get_bits1(&gb);
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p->subframe[1].grid_index = get_bits1(&gb);
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p->subframe[2].grid_index = get_bits1(&gb);
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p->subframe[3].grid_index = get_bits1(&gb);
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if (p->cur_rate == RATE_6300) {
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skip_bits1(&gb); /* skip reserved bit */
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/* Compute pulse_pos index using the 13-bit combined position index */
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temp = get_bits(&gb, 13);
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p->subframe[0].pulse_pos = temp / 810;
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temp -= p->subframe[0].pulse_pos * 810;
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p->subframe[1].pulse_pos = FASTDIV(temp, 90);
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temp -= p->subframe[1].pulse_pos * 90;
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p->subframe[2].pulse_pos = FASTDIV(temp, 9);
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p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
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p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
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get_bits(&gb, 16);
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p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
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get_bits(&gb, 14);
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p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
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get_bits(&gb, 16);
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p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
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get_bits(&gb, 14);
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p->subframe[0].pulse_sign = get_bits(&gb, 6);
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p->subframe[1].pulse_sign = get_bits(&gb, 5);
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p->subframe[2].pulse_sign = get_bits(&gb, 6);
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p->subframe[3].pulse_sign = get_bits(&gb, 5);
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} else { /* 5300 bps */
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p->subframe[0].pulse_pos = get_bits(&gb, 12);
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p->subframe[1].pulse_pos = get_bits(&gb, 12);
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p->subframe[2].pulse_pos = get_bits(&gb, 12);
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p->subframe[3].pulse_pos = get_bits(&gb, 12);
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p->subframe[0].pulse_sign = get_bits(&gb, 4);
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p->subframe[1].pulse_sign = get_bits(&gb, 4);
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p->subframe[2].pulse_sign = get_bits(&gb, 4);
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p->subframe[3].pulse_sign = get_bits(&gb, 4);
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}
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return 0;
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}
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/**
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* Bitexact implementation of sqrt(val/2).
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*/
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static int16_t square_root(unsigned val)
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{
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av_assert2(!(val & 0x80000000));
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return (ff_sqrt(val << 1) >> 1) & (~1);
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}
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/**
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* Generate fixed codebook excitation vector.
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*
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* @param vector decoded excitation vector
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* @param subfrm current subframe
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* @param cur_rate current bitrate
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* @param pitch_lag closed loop pitch lag
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* @param index current subframe index
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*/
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static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
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enum Rate cur_rate, int pitch_lag, int index)
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{
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int temp, i, j;
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memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
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if (cur_rate == RATE_6300) {
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if (subfrm->pulse_pos >= max_pos[index])
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return;
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/* Decode amplitudes and positions */
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j = PULSE_MAX - pulses[index];
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temp = subfrm->pulse_pos;
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for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
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temp -= ff_g723_1_combinatorial_table[j][i];
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if (temp >= 0)
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continue;
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temp += ff_g723_1_combinatorial_table[j++][i];
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if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
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vector[subfrm->grid_index + GRID_SIZE * i] =
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-ff_g723_1_fixed_cb_gain[subfrm->amp_index];
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} else {
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vector[subfrm->grid_index + GRID_SIZE * i] =
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ff_g723_1_fixed_cb_gain[subfrm->amp_index];
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}
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if (j == PULSE_MAX)
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break;
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}
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if (subfrm->dirac_train == 1)
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ff_g723_1_gen_dirac_train(vector, pitch_lag);
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} else { /* 5300 bps */
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int cb_gain = ff_g723_1_fixed_cb_gain[subfrm->amp_index];
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int cb_shift = subfrm->grid_index;
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int cb_sign = subfrm->pulse_sign;
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int cb_pos = subfrm->pulse_pos;
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int offset, beta, lag;
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for (i = 0; i < 8; i += 2) {
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offset = ((cb_pos & 7) << 3) + cb_shift + i;
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vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
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cb_pos >>= 3;
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cb_sign >>= 1;
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}
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/* Enhance harmonic components */
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lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
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subfrm->ad_cb_lag - 1;
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beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
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if (lag < SUBFRAME_LEN - 2) {
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for (i = lag; i < SUBFRAME_LEN; i++)
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vector[i] += beta * vector[i - lag] >> 15;
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}
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}
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}
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/**
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* Estimate maximum auto-correlation around pitch lag.
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*
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* @param buf buffer with offset applied
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* @param offset offset of the excitation vector
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* @param ccr_max pointer to the maximum auto-correlation
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* @param pitch_lag decoded pitch lag
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* @param length length of autocorrelation
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* @param dir forward lag(1) / backward lag(-1)
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*/
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static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
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int pitch_lag, int length, int dir)
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{
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int limit, ccr, lag = 0;
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int i;
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pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
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if (dir > 0)
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limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
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else
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limit = pitch_lag + 3;
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for (i = pitch_lag - 3; i <= limit; i++) {
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ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
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if (ccr > *ccr_max) {
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*ccr_max = ccr;
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lag = i;
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}
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}
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return lag;
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}
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/**
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* Calculate pitch postfilter optimal and scaling gains.
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*
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* @param lag pitch postfilter forward/backward lag
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* @param ppf pitch postfilter parameters
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* @param cur_rate current bitrate
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* @param tgt_eng target energy
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* @param ccr cross-correlation
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* @param res_eng residual energy
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*/
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static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
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int tgt_eng, int ccr, int res_eng)
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{
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int pf_residual; /* square of postfiltered residual */
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int temp1, temp2;
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ppf->index = lag;
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temp1 = tgt_eng * res_eng >> 1;
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temp2 = ccr * ccr << 1;
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if (temp2 > temp1) {
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if (ccr >= res_eng) {
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ppf->opt_gain = ppf_gain_weight[cur_rate];
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} else {
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ppf->opt_gain = (ccr << 15) / res_eng *
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ppf_gain_weight[cur_rate] >> 15;
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}
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/* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
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temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
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temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
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pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
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if (tgt_eng >= pf_residual << 1) {
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temp1 = 0x7fff;
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} else {
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temp1 = (tgt_eng << 14) / pf_residual;
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}
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/* scaling_gain = sqrt(tgt_eng/pf_res^2) */
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ppf->sc_gain = square_root(temp1 << 16);
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} else {
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ppf->opt_gain = 0;
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ppf->sc_gain = 0x7fff;
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}
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ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
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}
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/**
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* Calculate pitch postfilter parameters.
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*
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* @param p the context
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* @param offset offset of the excitation vector
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* @param pitch_lag decoded pitch lag
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* @param ppf pitch postfilter parameters
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* @param cur_rate current bitrate
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*/
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static void comp_ppf_coeff(G723_1_ChannelContext *p, int offset, int pitch_lag,
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PPFParam *ppf, enum Rate cur_rate)
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{
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int16_t scale;
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int i;
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int temp1, temp2;
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/*
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* 0 - target energy
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* 1 - forward cross-correlation
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* 2 - forward residual energy
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|
* 3 - backward cross-correlation
|
|
* 4 - backward residual energy
|
|
*/
|
|
int energy[5] = {0, 0, 0, 0, 0};
|
|
int16_t *buf = p->audio + LPC_ORDER + offset;
|
|
int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
|
|
SUBFRAME_LEN, 1);
|
|
int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
|
|
SUBFRAME_LEN, -1);
|
|
|
|
ppf->index = 0;
|
|
ppf->opt_gain = 0;
|
|
ppf->sc_gain = 0x7fff;
|
|
|
|
/* Case 0, Section 3.6 */
|
|
if (!back_lag && !fwd_lag)
|
|
return;
|
|
|
|
/* Compute target energy */
|
|
energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);
|
|
|
|
/* Compute forward residual energy */
|
|
if (fwd_lag)
|
|
energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
|
|
SUBFRAME_LEN);
|
|
|
|
/* Compute backward residual energy */
|
|
if (back_lag)
|
|
energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
|
|
SUBFRAME_LEN);
|
|
|
|
/* Normalize and shorten */
|
|
temp1 = 0;
|
|
for (i = 0; i < 5; i++)
|
|
temp1 = FFMAX(energy[i], temp1);
|
|
|
|
scale = ff_g723_1_normalize_bits(temp1, 31);
|
|
for (i = 0; i < 5; i++)
|
|
energy[i] = (energy[i] << scale) >> 16;
|
|
|
|
if (fwd_lag && !back_lag) { /* Case 1 */
|
|
comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
|
|
energy[2]);
|
|
} else if (!fwd_lag) { /* Case 2 */
|
|
comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
|
|
energy[4]);
|
|
} else { /* Case 3 */
|
|
|
|
/*
|
|
* Select the largest of energy[1]^2/energy[2]
|
|
* and energy[3]^2/energy[4]
|
|
*/
|
|
temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
|
|
temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
|
|
if (temp1 >= temp2) {
|
|
comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
|
|
energy[2]);
|
|
} else {
|
|
comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
|
|
energy[4]);
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Classify frames as voiced/unvoiced.
|
|
*
|
|
* @param p the context
|
|
* @param pitch_lag decoded pitch_lag
|
|
* @param exc_eng excitation energy estimation
|
|
* @param scale scaling factor of exc_eng
|
|
*
|
|
* @return residual interpolation index if voiced, 0 otherwise
|
|
*/
|
|
static int comp_interp_index(G723_1_ChannelContext *p, int pitch_lag,
|
|
int *exc_eng, int *scale)
|
|
{
|
|
int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
|
|
int16_t *buf = p->audio + LPC_ORDER;
|
|
|
|
int index, ccr, tgt_eng, best_eng, temp;
|
|
|
|
*scale = ff_g723_1_scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
|
|
buf += offset;
|
|
|
|
/* Compute maximum backward cross-correlation */
|
|
ccr = 0;
|
|
index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
|
|
ccr = av_sat_add32(ccr, 1 << 15) >> 16;
|
|
|
|
/* Compute target energy */
|
|
tgt_eng = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);
|
|
*exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
|
|
|
|
if (ccr <= 0)
|
|
return 0;
|
|
|
|
/* Compute best energy */
|
|
best_eng = ff_g723_1_dot_product(buf - index, buf - index,
|
|
SUBFRAME_LEN * 2);
|
|
best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
|
|
|
|
temp = best_eng * *exc_eng >> 3;
|
|
|
|
if (temp < ccr * ccr) {
|
|
return index;
|
|
} else
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Perform residual interpolation based on frame classification.
|
|
*
|
|
* @param buf decoded excitation vector
|
|
* @param out output vector
|
|
* @param lag decoded pitch lag
|
|
* @param gain interpolated gain
|
|
* @param rseed seed for random number generator
|
|
*/
|
|
static void residual_interp(int16_t *buf, int16_t *out, int lag,
|
|
int gain, int *rseed)
|
|
{
|
|
int i;
|
|
if (lag) { /* Voiced */
|
|
int16_t *vector_ptr = buf + PITCH_MAX;
|
|
/* Attenuate */
|
|
for (i = 0; i < lag; i++)
|
|
out[i] = vector_ptr[i - lag] * 3 >> 2;
|
|
av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
|
|
(FRAME_LEN - lag) * sizeof(*out));
|
|
} else { /* Unvoiced */
|
|
for (i = 0; i < FRAME_LEN; i++) {
|
|
*rseed = (int16_t)(*rseed * 521 + 259);
|
|
out[i] = gain * *rseed >> 15;
|
|
}
|
|
memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Perform IIR filtering.
|
|
*
|
|
* @param fir_coef FIR coefficients
|
|
* @param iir_coef IIR coefficients
|
|
* @param src source vector
|
|
* @param dest destination vector
|
|
* @param width width of the output, 16 bits(0) / 32 bits(1)
|
|
*/
|
|
#define iir_filter(fir_coef, iir_coef, src, dest, width)\
|
|
{\
|
|
int m, n;\
|
|
int res_shift = 16 & ~-(width);\
|
|
int in_shift = 16 - res_shift;\
|
|
\
|
|
for (m = 0; m < SUBFRAME_LEN; m++) {\
|
|
int64_t filter = 0;\
|
|
for (n = 1; n <= LPC_ORDER; n++) {\
|
|
filter -= (fir_coef)[n - 1] * (src)[m - n] -\
|
|
(iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
|
|
}\
|
|
\
|
|
(dest)[m] = av_clipl_int32(((src)[m] * 65536) + (filter * 8) +\
|
|
(1 << 15)) >> res_shift;\
|
|
}\
|
|
}
|
|
|
|
/**
|
|
* Adjust gain of postfiltered signal.
|
|
*
|
|
* @param p the context
|
|
* @param buf postfiltered output vector
|
|
* @param energy input energy coefficient
|
|
*/
|
|
static void gain_scale(G723_1_ChannelContext *p, int16_t * buf, int energy)
|
|
{
|
|
int num, denom, gain, bits1, bits2;
|
|
int i;
|
|
|
|
num = energy;
|
|
denom = 0;
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
int temp = buf[i] >> 2;
|
|
temp *= temp;
|
|
denom = av_sat_dadd32(denom, temp);
|
|
}
|
|
|
|
if (num && denom) {
|
|
bits1 = ff_g723_1_normalize_bits(num, 31);
|
|
bits2 = ff_g723_1_normalize_bits(denom, 31);
|
|
num = num << bits1 >> 1;
|
|
denom <<= bits2;
|
|
|
|
bits2 = 5 + bits1 - bits2;
|
|
bits2 = av_clip_uintp2(bits2, 5);
|
|
|
|
gain = (num >> 1) / (denom >> 16);
|
|
gain = square_root(gain << 16 >> bits2);
|
|
} else {
|
|
gain = 1 << 12;
|
|
}
|
|
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
|
|
buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
|
|
(1 << 10)) >> 11);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Perform formant filtering.
|
|
*
|
|
* @param p the context
|
|
* @param lpc quantized lpc coefficients
|
|
* @param buf input buffer
|
|
* @param dst output buffer
|
|
*/
|
|
static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc,
|
|
int16_t *buf, int16_t *dst)
|
|
{
|
|
int16_t filter_coef[2][LPC_ORDER];
|
|
int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
|
|
int i, j, k;
|
|
|
|
memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
|
|
memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
|
|
|
|
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
|
|
for (k = 0; k < LPC_ORDER; k++) {
|
|
filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
|
|
(1 << 14)) >> 15;
|
|
filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
|
|
(1 << 14)) >> 15;
|
|
}
|
|
iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);
|
|
lpc += LPC_ORDER;
|
|
}
|
|
|
|
memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
|
|
memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
|
|
|
|
buf += LPC_ORDER;
|
|
signal_ptr = filter_signal + LPC_ORDER;
|
|
for (i = 0; i < SUBFRAMES; i++) {
|
|
int temp;
|
|
int auto_corr[2];
|
|
int scale, energy;
|
|
|
|
/* Normalize */
|
|
scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);
|
|
|
|
/* Compute auto correlation coefficients */
|
|
auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
|
|
auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN);
|
|
|
|
/* Compute reflection coefficient */
|
|
temp = auto_corr[1] >> 16;
|
|
if (temp) {
|
|
temp = (auto_corr[0] >> 2) / temp;
|
|
}
|
|
p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
|
|
temp = -p->reflection_coef >> 1 & ~3;
|
|
|
|
/* Compensation filter */
|
|
for (j = 0; j < SUBFRAME_LEN; j++) {
|
|
dst[j] = av_sat_dadd32(signal_ptr[j],
|
|
(signal_ptr[j - 1] >> 16) * temp) >> 16;
|
|
}
|
|
|
|
/* Compute normalized signal energy */
|
|
temp = 2 * scale + 4;
|
|
if (temp < 0) {
|
|
energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
|
|
} else
|
|
energy = auto_corr[1] >> temp;
|
|
|
|
gain_scale(p, dst, energy);
|
|
|
|
buf += SUBFRAME_LEN;
|
|
signal_ptr += SUBFRAME_LEN;
|
|
dst += SUBFRAME_LEN;
|
|
}
|
|
}
|
|
|
|
static int sid_gain_to_lsp_index(int gain)
|
|
{
|
|
if (gain < 0x10)
|
|
return gain << 6;
|
|
else if (gain < 0x20)
|
|
return gain - 8 << 7;
|
|
else
|
|
return gain - 20 << 8;
|
|
}
|
|
|
|
static inline int cng_rand(int *state, int base)
|
|
{
|
|
*state = (*state * 521 + 259) & 0xFFFF;
|
|
return (*state & 0x7FFF) * base >> 15;
|
|
}
|
|
|
|
static int estimate_sid_gain(G723_1_ChannelContext *p)
|
|
{
|
|
int i, shift, seg, seg2, t, val, val_add, x, y;
|
|
|
|
shift = 16 - p->cur_gain * 2;
|
|
if (shift > 0) {
|
|
if (p->sid_gain == 0) {
|
|
t = 0;
|
|
} else if (shift >= 31 || (int32_t)((uint32_t)p->sid_gain << shift) >> shift != p->sid_gain) {
|
|
if (p->sid_gain < 0) t = INT32_MIN;
|
|
else t = INT32_MAX;
|
|
} else
|
|
t = p->sid_gain * (1 << shift);
|
|
} else if(shift < -31) {
|
|
t = (p->sid_gain < 0) ? -1 : 0;
|
|
}else
|
|
t = p->sid_gain >> -shift;
|
|
x = av_clipl_int32(t * (int64_t)cng_filt[0] >> 16);
|
|
|
|
if (x >= cng_bseg[2])
|
|
return 0x3F;
|
|
|
|
if (x >= cng_bseg[1]) {
|
|
shift = 4;
|
|
seg = 3;
|
|
} else {
|
|
shift = 3;
|
|
seg = (x >= cng_bseg[0]);
|
|
}
|
|
seg2 = FFMIN(seg, 3);
|
|
|
|
val = 1 << shift;
|
|
val_add = val >> 1;
|
|
for (i = 0; i < shift; i++) {
|
|
t = seg * 32 + (val << seg2);
|
|
t *= t;
|
|
if (x >= t)
|
|
val += val_add;
|
|
else
|
|
val -= val_add;
|
|
val_add >>= 1;
|
|
}
|
|
|
|
t = seg * 32 + (val << seg2);
|
|
y = t * t - x;
|
|
if (y <= 0) {
|
|
t = seg * 32 + (val + 1 << seg2);
|
|
t = t * t - x;
|
|
val = (seg2 - 1) * 16 + val;
|
|
if (t >= y)
|
|
val++;
|
|
} else {
|
|
t = seg * 32 + (val - 1 << seg2);
|
|
t = t * t - x;
|
|
val = (seg2 - 1) * 16 + val;
|
|
if (t >= y)
|
|
val--;
|
|
}
|
|
|
|
return val;
|
|
}
|
|
|
|
static void generate_noise(G723_1_ChannelContext *p)
|
|
{
|
|
int i, j, idx, t;
|
|
int off[SUBFRAMES];
|
|
int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
|
|
int tmp[SUBFRAME_LEN * 2];
|
|
int16_t *vector_ptr;
|
|
int64_t sum;
|
|
int b0, c, delta, x, shift;
|
|
|
|
p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
|
|
p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
|
|
|
|
for (i = 0; i < SUBFRAMES; i++) {
|
|
p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
|
|
p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i];
|
|
}
|
|
|
|
for (i = 0; i < SUBFRAMES / 2; i++) {
|
|
t = cng_rand(&p->cng_random_seed, 1 << 13);
|
|
off[i * 2] = t & 1;
|
|
off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
|
|
t >>= 2;
|
|
for (j = 0; j < 11; j++) {
|
|
signs[i * 11 + j] = ((t & 1) * 2 - 1) * (1 << 14);
|
|
t >>= 1;
|
|
}
|
|
}
|
|
|
|
idx = 0;
|
|
for (i = 0; i < SUBFRAMES; i++) {
|
|
for (j = 0; j < SUBFRAME_LEN / 2; j++)
|
|
tmp[j] = j;
|
|
t = SUBFRAME_LEN / 2;
|
|
for (j = 0; j < pulses[i]; j++, idx++) {
|
|
int idx2 = cng_rand(&p->cng_random_seed, t);
|
|
|
|
pos[idx] = tmp[idx2] * 2 + off[i];
|
|
tmp[idx2] = tmp[--t];
|
|
}
|
|
}
|
|
|
|
vector_ptr = p->audio + LPC_ORDER;
|
|
memcpy(vector_ptr, p->prev_excitation,
|
|
PITCH_MAX * sizeof(*p->excitation));
|
|
for (i = 0; i < SUBFRAMES; i += 2) {
|
|
ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
|
|
p->pitch_lag[i >> 1], &p->subframe[i],
|
|
p->cur_rate);
|
|
ff_g723_1_gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
|
|
vector_ptr + SUBFRAME_LEN,
|
|
p->pitch_lag[i >> 1], &p->subframe[i + 1],
|
|
p->cur_rate);
|
|
|
|
t = 0;
|
|
for (j = 0; j < SUBFRAME_LEN * 2; j++)
|
|
t |= FFABS(vector_ptr[j]);
|
|
t = FFMIN(t, 0x7FFF);
|
|
if (!t) {
|
|
shift = 0;
|
|
} else {
|
|
shift = -10 + av_log2(t);
|
|
if (shift < -2)
|
|
shift = -2;
|
|
}
|
|
sum = 0;
|
|
if (shift < 0) {
|
|
for (j = 0; j < SUBFRAME_LEN * 2; j++) {
|
|
t = vector_ptr[j] * (1 << -shift);
|
|
sum += t * t;
|
|
tmp[j] = t;
|
|
}
|
|
} else {
|
|
for (j = 0; j < SUBFRAME_LEN * 2; j++) {
|
|
t = vector_ptr[j] >> shift;
|
|
sum += t * t;
|
|
tmp[j] = t;
|
|
}
|
|
}
|
|
|
|
b0 = 0;
|
|
for (j = 0; j < 11; j++)
|
|
b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
|
|
b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
|
|
|
|
c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
|
|
if (shift * 2 + 3 >= 0)
|
|
c >>= shift * 2 + 3;
|
|
else
|
|
c <<= -(shift * 2 + 3);
|
|
c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
|
|
|
|
delta = b0 * b0 * 2 - c;
|
|
if (delta <= 0) {
|
|
x = -b0;
|
|
} else {
|
|
delta = square_root(delta);
|
|
x = delta - b0;
|
|
t = delta + b0;
|
|
if (FFABS(t) < FFABS(x))
|
|
x = -t;
|
|
}
|
|
shift++;
|
|
if (shift < 0)
|
|
x >>= -shift;
|
|
else
|
|
x *= 1 << shift;
|
|
x = av_clip(x, -10000, 10000);
|
|
|
|
for (j = 0; j < 11; j++) {
|
|
idx = (i / 2) * 11 + j;
|
|
vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
|
|
(x * signs[idx] >> 15));
|
|
}
|
|
|
|
/* copy decoded data to serve as a history for the next decoded subframes */
|
|
memcpy(vector_ptr + PITCH_MAX, vector_ptr,
|
|
sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
|
|
vector_ptr += SUBFRAME_LEN * 2;
|
|
}
|
|
/* Save the excitation for the next frame */
|
|
memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
|
|
PITCH_MAX * sizeof(*p->excitation));
|
|
}
|
|
|
|
static int g723_1_decode_frame(AVCodecContext *avctx, AVFrame *frame,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
G723_1_Context *s = avctx->priv_data;
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
int dec_mode = buf[0] & 3;
|
|
int channels = avctx->ch_layout.nb_channels;
|
|
|
|
PPFParam ppf[SUBFRAMES];
|
|
int16_t cur_lsp[LPC_ORDER];
|
|
int16_t lpc[SUBFRAMES * LPC_ORDER];
|
|
int16_t acb_vector[SUBFRAME_LEN];
|
|
int16_t *out;
|
|
int bad_frame = 0, i, j, ret;
|
|
|
|
if (buf_size < frame_size[dec_mode] * channels) {
|
|
if (buf_size)
|
|
av_log(avctx, AV_LOG_WARNING,
|
|
"Expected %d bytes, got %d - skipping packet\n",
|
|
frame_size[dec_mode], buf_size);
|
|
*got_frame_ptr = 0;
|
|
return buf_size;
|
|
}
|
|
|
|
frame->nb_samples = FRAME_LEN;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
|
|
for (int ch = 0; ch < channels; ch++) {
|
|
G723_1_ChannelContext *p = &s->ch[ch];
|
|
int16_t *audio = p->audio;
|
|
|
|
if (unpack_bitstream(p, buf + ch * (buf_size / channels),
|
|
buf_size / channels) < 0) {
|
|
bad_frame = 1;
|
|
if (p->past_frame_type == ACTIVE_FRAME)
|
|
p->cur_frame_type = ACTIVE_FRAME;
|
|
else
|
|
p->cur_frame_type = UNTRANSMITTED_FRAME;
|
|
}
|
|
|
|
out = (int16_t *)frame->extended_data[ch];
|
|
|
|
if (p->cur_frame_type == ACTIVE_FRAME) {
|
|
if (!bad_frame)
|
|
p->erased_frames = 0;
|
|
else if (p->erased_frames != 3)
|
|
p->erased_frames++;
|
|
|
|
ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
|
|
ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
|
|
|
|
/* Save the lsp_vector for the next frame */
|
|
memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
|
|
|
|
/* Generate the excitation for the frame */
|
|
memcpy(p->excitation, p->prev_excitation,
|
|
PITCH_MAX * sizeof(*p->excitation));
|
|
if (!p->erased_frames) {
|
|
int16_t *vector_ptr = p->excitation + PITCH_MAX;
|
|
|
|
/* Update interpolation gain memory */
|
|
p->interp_gain = ff_g723_1_fixed_cb_gain[(p->subframe[2].amp_index +
|
|
p->subframe[3].amp_index) >> 1];
|
|
for (i = 0; i < SUBFRAMES; i++) {
|
|
gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
|
|
p->pitch_lag[i >> 1], i);
|
|
ff_g723_1_gen_acb_excitation(acb_vector,
|
|
&p->excitation[SUBFRAME_LEN * i],
|
|
p->pitch_lag[i >> 1],
|
|
&p->subframe[i], p->cur_rate);
|
|
/* Get the total excitation */
|
|
for (j = 0; j < SUBFRAME_LEN; j++) {
|
|
int v = av_clip_int16(vector_ptr[j] * 2);
|
|
vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
|
|
}
|
|
vector_ptr += SUBFRAME_LEN;
|
|
}
|
|
|
|
vector_ptr = p->excitation + PITCH_MAX;
|
|
|
|
p->interp_index = comp_interp_index(p, p->pitch_lag[1],
|
|
&p->sid_gain, &p->cur_gain);
|
|
|
|
/* Perform pitch postfiltering */
|
|
if (s->postfilter) {
|
|
i = PITCH_MAX;
|
|
for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
|
comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
|
|
ppf + j, p->cur_rate);
|
|
|
|
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
|
ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
|
|
vector_ptr + i,
|
|
vector_ptr + i + ppf[j].index,
|
|
ppf[j].sc_gain,
|
|
ppf[j].opt_gain,
|
|
1 << 14, 15, SUBFRAME_LEN);
|
|
} else {
|
|
audio = vector_ptr - LPC_ORDER;
|
|
}
|
|
|
|
/* Save the excitation for the next frame */
|
|
memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
|
|
PITCH_MAX * sizeof(*p->excitation));
|
|
} else {
|
|
p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
|
|
if (p->erased_frames == 3) {
|
|
/* Mute output */
|
|
memset(p->excitation, 0,
|
|
(FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
|
|
memset(p->prev_excitation, 0,
|
|
PITCH_MAX * sizeof(*p->excitation));
|
|
memset(frame->data[0], 0,
|
|
(FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
|
|
} else {
|
|
int16_t *buf = p->audio + LPC_ORDER;
|
|
|
|
/* Regenerate frame */
|
|
residual_interp(p->excitation, buf, p->interp_index,
|
|
p->interp_gain, &p->random_seed);
|
|
|
|
/* Save the excitation for the next frame */
|
|
memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
|
|
PITCH_MAX * sizeof(*p->excitation));
|
|
}
|
|
}
|
|
p->cng_random_seed = CNG_RANDOM_SEED;
|
|
} else {
|
|
if (p->cur_frame_type == SID_FRAME) {
|
|
p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
|
|
ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
|
|
} else if (p->past_frame_type == ACTIVE_FRAME) {
|
|
p->sid_gain = estimate_sid_gain(p);
|
|
}
|
|
|
|
if (p->past_frame_type == ACTIVE_FRAME)
|
|
p->cur_gain = p->sid_gain;
|
|
else
|
|
p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
|
|
generate_noise(p);
|
|
ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
|
|
/* Save the lsp_vector for the next frame */
|
|
memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
|
|
}
|
|
|
|
p->past_frame_type = p->cur_frame_type;
|
|
|
|
memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
|
|
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
|
ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
|
|
audio + i, SUBFRAME_LEN, LPC_ORDER,
|
|
0, 1, 1 << 12);
|
|
memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
|
|
|
|
if (s->postfilter) {
|
|
formant_postfilter(p, lpc, p->audio, out);
|
|
} else { // if output is not postfiltered it should be scaled by 2
|
|
for (i = 0; i < FRAME_LEN; i++)
|
|
out[i] = av_clip_int16(2 * p->audio[LPC_ORDER + i]);
|
|
}
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return frame_size[dec_mode] * channels;
|
|
}
|
|
|
|
#define OFFSET(x) offsetof(G723_1_Context, x)
|
|
#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
|
|
|
|
static const AVOption options[] = {
|
|
{ "postfilter", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL,
|
|
{ .i64 = 1 }, 0, 1, AD },
|
|
{ NULL }
|
|
};
|
|
|
|
|
|
static const AVClass g723_1dec_class = {
|
|
.class_name = "G.723.1 decoder",
|
|
.item_name = av_default_item_name,
|
|
.option = options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
const FFCodec ff_g723_1_decoder = {
|
|
.p.name = "g723_1",
|
|
CODEC_LONG_NAME("G.723.1"),
|
|
.p.type = AVMEDIA_TYPE_AUDIO,
|
|
.p.id = AV_CODEC_ID_G723_1,
|
|
.priv_data_size = sizeof(G723_1_Context),
|
|
.init = g723_1_decode_init,
|
|
FF_CODEC_DECODE_CB(g723_1_decode_frame),
|
|
.p.capabilities =
|
|
#if FF_API_SUBFRAMES
|
|
AV_CODEC_CAP_SUBFRAMES |
|
|
#endif
|
|
AV_CODEC_CAP_DR1,
|
|
.p.priv_class = &g723_1dec_class,
|
|
};
|