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FFmpeg/libavcodec/truespeech.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs
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>
2021-04-27 10:43:15 -03:00

368 lines
11 KiB
C

/*
* DSP Group TrueSpeech compatible decoder
* Copyright (c) 2005 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mem_internal.h"
#include "avcodec.h"
#include "bswapdsp.h"
#include "get_bits.h"
#include "internal.h"
#include "truespeech_data.h"
/**
* @file
* TrueSpeech decoder.
*/
/**
* TrueSpeech decoder context
*/
typedef struct TSContext {
BswapDSPContext bdsp;
/* input data */
DECLARE_ALIGNED(16, uint8_t, buffer)[32];
int16_t vector[8]; ///< input vector: 5/5/4/4/4/3/3/3
int offset1[2]; ///< 8-bit value, used in one copying offset
int offset2[4]; ///< 7-bit value, encodes offsets for copying and for two-point filter
int pulseoff[4]; ///< 4-bit offset of pulse values block
int pulsepos[4]; ///< 27-bit variable, encodes 7 pulse positions
int pulseval[4]; ///< 7x2-bit pulse values
int flag; ///< 1-bit flag, shows how to choose filters
/* temporary data */
int filtbuf[146]; // some big vector used for storing filters
int prevfilt[8]; // filter from previous frame
int16_t tmp1[8]; // coefficients for adding to out
int16_t tmp2[8]; // coefficients for adding to out
int16_t tmp3[8]; // coefficients for adding to out
int16_t cvector[8]; // correlated input vector
int filtval; // gain value for one function
int16_t newvec[60]; // tmp vector
int16_t filters[32]; // filters for every subframe
} TSContext;
static av_cold int truespeech_decode_init(AVCodecContext * avctx)
{
TSContext *c = avctx->priv_data;
if (avctx->channels != 1) {
avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
return AVERROR_PATCHWELCOME;
}
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
ff_bswapdsp_init(&c->bdsp);
return 0;
}
static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
{
GetBitContext gb;
dec->bdsp.bswap_buf((uint32_t *) dec->buffer, (const uint32_t *) input, 8);
init_get_bits(&gb, dec->buffer, 32 * 8);
dec->vector[7] = ts_codebook[7][get_bits(&gb, 3)];
dec->vector[6] = ts_codebook[6][get_bits(&gb, 3)];
dec->vector[5] = ts_codebook[5][get_bits(&gb, 3)];
dec->vector[4] = ts_codebook[4][get_bits(&gb, 4)];
dec->vector[3] = ts_codebook[3][get_bits(&gb, 4)];
dec->vector[2] = ts_codebook[2][get_bits(&gb, 4)];
dec->vector[1] = ts_codebook[1][get_bits(&gb, 5)];
dec->vector[0] = ts_codebook[0][get_bits(&gb, 5)];
dec->flag = get_bits1(&gb);
dec->offset1[0] = get_bits(&gb, 4) << 4;
dec->offset2[3] = get_bits(&gb, 7);
dec->offset2[2] = get_bits(&gb, 7);
dec->offset2[1] = get_bits(&gb, 7);
dec->offset2[0] = get_bits(&gb, 7);
dec->offset1[1] = get_bits(&gb, 4);
dec->pulseval[1] = get_bits(&gb, 14);
dec->pulseval[0] = get_bits(&gb, 14);
dec->offset1[1] |= get_bits(&gb, 4) << 4;
dec->pulseval[3] = get_bits(&gb, 14);
dec->pulseval[2] = get_bits(&gb, 14);
dec->offset1[0] |= get_bits1(&gb);
dec->pulsepos[0] = get_bits_long(&gb, 27);
dec->pulseoff[0] = get_bits(&gb, 4);
dec->offset1[0] |= get_bits1(&gb) << 1;
dec->pulsepos[1] = get_bits_long(&gb, 27);
dec->pulseoff[1] = get_bits(&gb, 4);
dec->offset1[0] |= get_bits1(&gb) << 2;
dec->pulsepos[2] = get_bits_long(&gb, 27);
dec->pulseoff[2] = get_bits(&gb, 4);
dec->offset1[0] |= get_bits1(&gb) << 3;
dec->pulsepos[3] = get_bits_long(&gb, 27);
dec->pulseoff[3] = get_bits(&gb, 4);
}
static void truespeech_correlate_filter(TSContext *dec)
{
int16_t tmp[8];
int i, j;
for(i = 0; i < 8; i++){
if(i > 0){
memcpy(tmp, dec->cvector, i * sizeof(*tmp));
for(j = 0; j < i; j++)
dec->cvector[j] += (tmp[i - j - 1] * dec->vector[i] + 0x4000) >> 15;
}
dec->cvector[i] = (8 - dec->vector[i]) >> 3;
}
for(i = 0; i < 8; i++)
dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
dec->filtval = dec->vector[0];
}
static void truespeech_filters_merge(TSContext *dec)
{
int i;
if(!dec->flag){
for(i = 0; i < 8; i++){
dec->filters[i + 0] = dec->prevfilt[i];
dec->filters[i + 8] = dec->prevfilt[i];
}
}else{
for(i = 0; i < 8; i++){
dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
}
}
for(i = 0; i < 8; i++){
dec->filters[i + 16] = dec->cvector[i];
dec->filters[i + 24] = dec->cvector[i];
}
}
static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
{
int16_t tmp[146 + 60], *ptr0, *ptr1;
const int16_t *filter;
int i, t, off;
t = dec->offset2[quart];
if(t == 127){
memset(dec->newvec, 0, 60 * sizeof(*dec->newvec));
return;
}
for(i = 0; i < 146; i++)
tmp[i] = dec->filtbuf[i];
off = (t / 25) + dec->offset1[quart >> 1] + 18;
off = av_clip(off, 0, 145);
ptr0 = tmp + 145 - off;
ptr1 = tmp + 146;
filter = ts_order2_coeffs + (t % 25) * 2;
for(i = 0; i < 60; i++){
t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
ptr0++;
dec->newvec[i] = t;
ptr1[i] = t;
}
}
static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
{
int16_t tmp[7];
int i, j, t;
const int16_t *ptr1;
int16_t *ptr2;
int coef;
memset(out, 0, 60 * sizeof(*out));
for(i = 0; i < 7; i++) {
t = dec->pulseval[quart] & 3;
dec->pulseval[quart] >>= 2;
tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
}
coef = dec->pulsepos[quart] >> 15;
ptr1 = ts_pulse_values + 30;
ptr2 = tmp;
for(i = 0, j = 3; (i < 30) && (j > 0); i++){
t = *ptr1++;
if(coef >= t)
coef -= t;
else{
out[i] = *ptr2++;
ptr1 += 30;
j--;
}
}
coef = dec->pulsepos[quart] & 0x7FFF;
ptr1 = ts_pulse_values;
for(i = 30, j = 4; (i < 60) && (j > 0); i++){
t = *ptr1++;
if(coef >= t)
coef -= t;
else{
out[i] = *ptr2++;
ptr1 += 30;
j--;
}
}
}
static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
{
int i;
memmove(dec->filtbuf, &dec->filtbuf[60], 86 * sizeof(*dec->filtbuf));
for(i = 0; i < 60; i++){
dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
out[i] += dec->newvec[i];
}
}
static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
{
int i,k;
int t[8];
int16_t *ptr0, *ptr1;
ptr0 = dec->tmp1;
ptr1 = dec->filters + quart * 8;
for(i = 0; i < 60; i++){
int sum = 0;
for(k = 0; k < 8; k++)
sum += ptr0[k] * (unsigned)ptr1[k];
sum = out[i] + ((int)(sum + 0x800U) >> 12);
out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
for(k = 7; k > 0; k--)
ptr0[k] = ptr0[k - 1];
ptr0[0] = out[i];
}
for(i = 0; i < 8; i++)
t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
ptr0 = dec->tmp2;
for(i = 0; i < 60; i++){
int sum = 0;
for(k = 0; k < 8; k++)
sum += ptr0[k] * t[k];
for(k = 7; k > 0; k--)
ptr0[k] = ptr0[k - 1];
ptr0[0] = out[i];
out[i] += (- sum) >> 12;
}
for(i = 0; i < 8; i++)
t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
ptr0 = dec->tmp3;
for(i = 0; i < 60; i++){
int sum = out[i] * (1 << 12);
for(k = 0; k < 8; k++)
sum += ptr0[k] * t[k];
for(k = 7; k > 0; k--)
ptr0[k] = ptr0[k - 1];
ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
sum = sum - (sum >> 3);
out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
}
}
static void truespeech_save_prevvec(TSContext *c)
{
int i;
for(i = 0; i < 8; i++)
c->prevfilt[i] = c->cvector[i];
}
static int truespeech_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
TSContext *c = avctx->priv_data;
int i, j;
int16_t *samples;
int iterations, ret;
iterations = buf_size / 32;
if (!iterations) {
av_log(avctx, AV_LOG_ERROR,
"Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
return -1;
}
/* get output buffer */
frame->nb_samples = iterations * 240;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
samples = (int16_t *)frame->data[0];
memset(samples, 0, iterations * 240 * sizeof(*samples));
for(j = 0; j < iterations; j++) {
truespeech_read_frame(c, buf);
buf += 32;
truespeech_correlate_filter(c);
truespeech_filters_merge(c);
for(i = 0; i < 4; i++) {
truespeech_apply_twopoint_filter(c, i);
truespeech_place_pulses (c, samples, i);
truespeech_update_filters(c, samples, i);
truespeech_synth (c, samples, i);
samples += 60;
}
truespeech_save_prevvec(c);
}
*got_frame_ptr = 1;
return buf_size;
}
const AVCodec ff_truespeech_decoder = {
.name = "truespeech",
.long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_TRUESPEECH,
.priv_data_size = sizeof(TSContext),
.init = truespeech_decode_init,
.decode = truespeech_decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
};