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370 lines
11 KiB
C
370 lines
11 KiB
C
/*
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* DSP Group TrueSpeech compatible decoder
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* Copyright (c) 2005 Konstantin Shishkov
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "dsputil.h"
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#include "get_bits.h"
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#include "truespeech_data.h"
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/**
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* @file
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* TrueSpeech decoder.
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*/
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/**
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* TrueSpeech decoder context
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*/
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typedef struct {
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AVFrame frame;
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DSPContext dsp;
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/* input data */
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DECLARE_ALIGNED(16, uint8_t, buffer)[32];
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int16_t vector[8]; ///< input vector: 5/5/4/4/4/3/3/3
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int offset1[2]; ///< 8-bit value, used in one copying offset
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int offset2[4]; ///< 7-bit value, encodes offsets for copying and for two-point filter
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int pulseoff[4]; ///< 4-bit offset of pulse values block
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int pulsepos[4]; ///< 27-bit variable, encodes 7 pulse positions
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int pulseval[4]; ///< 7x2-bit pulse values
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int flag; ///< 1-bit flag, shows how to choose filters
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/* temporary data */
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int filtbuf[146]; // some big vector used for storing filters
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int prevfilt[8]; // filter from previous frame
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int16_t tmp1[8]; // coefficients for adding to out
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int16_t tmp2[8]; // coefficients for adding to out
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int16_t tmp3[8]; // coefficients for adding to out
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int16_t cvector[8]; // correlated input vector
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int filtval; // gain value for one function
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int16_t newvec[60]; // tmp vector
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int16_t filters[32]; // filters for every subframe
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} TSContext;
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static av_cold int truespeech_decode_init(AVCodecContext * avctx)
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{
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TSContext *c = avctx->priv_data;
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if (avctx->channels != 1) {
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av_log_ask_for_sample(avctx, "Unsupported channel count: %d\n", avctx->channels);
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return AVERROR(EINVAL);
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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ff_dsputil_init(&c->dsp, avctx);
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avcodec_get_frame_defaults(&c->frame);
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avctx->coded_frame = &c->frame;
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return 0;
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}
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static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
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{
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GetBitContext gb;
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dec->dsp.bswap_buf((uint32_t *)dec->buffer, (const uint32_t *)input, 8);
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init_get_bits(&gb, dec->buffer, 32 * 8);
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dec->vector[7] = ts_codebook[7][get_bits(&gb, 3)];
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dec->vector[6] = ts_codebook[6][get_bits(&gb, 3)];
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dec->vector[5] = ts_codebook[5][get_bits(&gb, 3)];
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dec->vector[4] = ts_codebook[4][get_bits(&gb, 4)];
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dec->vector[3] = ts_codebook[3][get_bits(&gb, 4)];
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dec->vector[2] = ts_codebook[2][get_bits(&gb, 4)];
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dec->vector[1] = ts_codebook[1][get_bits(&gb, 5)];
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dec->vector[0] = ts_codebook[0][get_bits(&gb, 5)];
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dec->flag = get_bits1(&gb);
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dec->offset1[0] = get_bits(&gb, 4) << 4;
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dec->offset2[3] = get_bits(&gb, 7);
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dec->offset2[2] = get_bits(&gb, 7);
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dec->offset2[1] = get_bits(&gb, 7);
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dec->offset2[0] = get_bits(&gb, 7);
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dec->offset1[1] = get_bits(&gb, 4);
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dec->pulseval[1] = get_bits(&gb, 14);
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dec->pulseval[0] = get_bits(&gb, 14);
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dec->offset1[1] |= get_bits(&gb, 4) << 4;
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dec->pulseval[3] = get_bits(&gb, 14);
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dec->pulseval[2] = get_bits(&gb, 14);
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dec->offset1[0] |= get_bits1(&gb);
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dec->pulsepos[0] = get_bits_long(&gb, 27);
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dec->pulseoff[0] = get_bits(&gb, 4);
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dec->offset1[0] |= get_bits1(&gb) << 1;
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dec->pulsepos[1] = get_bits_long(&gb, 27);
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dec->pulseoff[1] = get_bits(&gb, 4);
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dec->offset1[0] |= get_bits1(&gb) << 2;
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dec->pulsepos[2] = get_bits_long(&gb, 27);
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dec->pulseoff[2] = get_bits(&gb, 4);
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dec->offset1[0] |= get_bits1(&gb) << 3;
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dec->pulsepos[3] = get_bits_long(&gb, 27);
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dec->pulseoff[3] = get_bits(&gb, 4);
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}
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static void truespeech_correlate_filter(TSContext *dec)
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{
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int16_t tmp[8];
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int i, j;
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for(i = 0; i < 8; i++){
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if(i > 0){
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memcpy(tmp, dec->cvector, i * sizeof(*tmp));
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for(j = 0; j < i; j++)
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dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) +
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(dec->cvector[j] << 15) + 0x4000) >> 15;
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}
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dec->cvector[i] = (8 - dec->vector[i]) >> 3;
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}
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for(i = 0; i < 8; i++)
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dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
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dec->filtval = dec->vector[0];
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}
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static void truespeech_filters_merge(TSContext *dec)
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{
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int i;
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if(!dec->flag){
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for(i = 0; i < 8; i++){
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dec->filters[i + 0] = dec->prevfilt[i];
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dec->filters[i + 8] = dec->prevfilt[i];
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}
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}else{
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for(i = 0; i < 8; i++){
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dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
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dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
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}
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}
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for(i = 0; i < 8; i++){
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dec->filters[i + 16] = dec->cvector[i];
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dec->filters[i + 24] = dec->cvector[i];
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}
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}
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static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
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{
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int16_t tmp[146 + 60], *ptr0, *ptr1;
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const int16_t *filter;
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int i, t, off;
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t = dec->offset2[quart];
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if(t == 127){
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memset(dec->newvec, 0, 60 * sizeof(*dec->newvec));
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return;
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}
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for(i = 0; i < 146; i++)
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tmp[i] = dec->filtbuf[i];
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off = (t / 25) + dec->offset1[quart >> 1] + 18;
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off = av_clip(off, 0, 145);
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ptr0 = tmp + 145 - off;
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ptr1 = tmp + 146;
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filter = (const int16_t*)ts_order2_coeffs + (t % 25) * 2;
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for(i = 0; i < 60; i++){
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t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
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ptr0++;
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dec->newvec[i] = t;
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ptr1[i] = t;
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}
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}
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static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
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{
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int16_t tmp[7];
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int i, j, t;
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const int16_t *ptr1;
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int16_t *ptr2;
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int coef;
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memset(out, 0, 60 * sizeof(*out));
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for(i = 0; i < 7; i++) {
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t = dec->pulseval[quart] & 3;
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dec->pulseval[quart] >>= 2;
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tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
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}
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coef = dec->pulsepos[quart] >> 15;
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ptr1 = (const int16_t*)ts_pulse_values + 30;
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ptr2 = tmp;
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for(i = 0, j = 3; (i < 30) && (j > 0); i++){
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t = *ptr1++;
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if(coef >= t)
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coef -= t;
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else{
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out[i] = *ptr2++;
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ptr1 += 30;
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j--;
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}
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}
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coef = dec->pulsepos[quart] & 0x7FFF;
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ptr1 = (const int16_t*)ts_pulse_values;
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for(i = 30, j = 4; (i < 60) && (j > 0); i++){
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t = *ptr1++;
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if(coef >= t)
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coef -= t;
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else{
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out[i] = *ptr2++;
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ptr1 += 30;
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j--;
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}
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}
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}
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static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
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{
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int i;
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memmove(dec->filtbuf, &dec->filtbuf[60], 86 * sizeof(*dec->filtbuf));
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for(i = 0; i < 60; i++){
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dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
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out[i] += dec->newvec[i];
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}
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}
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static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
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{
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int i,k;
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int t[8];
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int16_t *ptr0, *ptr1;
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ptr0 = dec->tmp1;
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ptr1 = dec->filters + quart * 8;
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for(i = 0; i < 60; i++){
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int sum = 0;
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for(k = 0; k < 8; k++)
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sum += ptr0[k] * ptr1[k];
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sum = (sum + (out[i] << 12) + 0x800) >> 12;
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out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
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for(k = 7; k > 0; k--)
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ptr0[k] = ptr0[k - 1];
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ptr0[0] = out[i];
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}
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for(i = 0; i < 8; i++)
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t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
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ptr0 = dec->tmp2;
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for(i = 0; i < 60; i++){
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int sum = 0;
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for(k = 0; k < 8; k++)
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sum += ptr0[k] * t[k];
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for(k = 7; k > 0; k--)
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ptr0[k] = ptr0[k - 1];
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ptr0[0] = out[i];
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out[i] = ((out[i] << 12) - sum) >> 12;
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}
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for(i = 0; i < 8; i++)
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t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
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ptr0 = dec->tmp3;
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for(i = 0; i < 60; i++){
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int sum = out[i] << 12;
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for(k = 0; k < 8; k++)
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sum += ptr0[k] * t[k];
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for(k = 7; k > 0; k--)
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ptr0[k] = ptr0[k - 1];
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ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
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sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
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sum = sum - (sum >> 3);
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out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
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}
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}
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static void truespeech_save_prevvec(TSContext *c)
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{
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int i;
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for(i = 0; i < 8; i++)
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c->prevfilt[i] = c->cvector[i];
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}
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static int truespeech_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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TSContext *c = avctx->priv_data;
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int i, j;
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int16_t *samples;
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int iterations, ret;
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iterations = buf_size / 32;
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if (!iterations) {
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av_log(avctx, AV_LOG_ERROR,
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"Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
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return -1;
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}
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/* get output buffer */
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c->frame.nb_samples = iterations * 240;
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if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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samples = (int16_t *)c->frame.data[0];
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memset(samples, 0, iterations * 240 * sizeof(*samples));
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for(j = 0; j < iterations; j++) {
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truespeech_read_frame(c, buf);
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buf += 32;
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truespeech_correlate_filter(c);
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truespeech_filters_merge(c);
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for(i = 0; i < 4; i++) {
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truespeech_apply_twopoint_filter(c, i);
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truespeech_place_pulses (c, samples, i);
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truespeech_update_filters(c, samples, i);
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truespeech_synth (c, samples, i);
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samples += 60;
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}
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truespeech_save_prevvec(c);
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}
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*got_frame_ptr = 1;
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*(AVFrame *)data = c->frame;
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return buf_size;
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}
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AVCodec ff_truespeech_decoder = {
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.name = "truespeech",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_TRUESPEECH,
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.priv_data_size = sizeof(TSContext),
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.init = truespeech_decode_init,
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.decode = truespeech_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
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};
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