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FFmpeg/libavcodec/truespeech.c
Andreas Rheinhardt 48286d4d98 avcodec/codec_internal: Add macro to set AVCodec.long_name 
It reduces typing: Before this patch, there were 105 codecs
whose long_name-definition exceeded the 80 char line length
limit. Now there are only nine of them.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-09-03 15:42:57 +02:00

368 lines
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/*
* 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/mem_internal.h"
#include "avcodec.h"
#include "bswapdsp.h"
#include "codec_internal.h"
#include "decode.h"
#include "get_bits.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->ch_layout.nb_channels != 1) {
avpriv_request_sample(avctx, "Channel count %d", avctx->ch_layout.nb_channels);
return AVERROR_PATCHWELCOME;
}
av_channel_layout_uninit(&avctx->ch_layout);
avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_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, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
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 FFCodec ff_truespeech_decoder = {
.p.name = "truespeech",
CODEC_LONG_NAME("DSP Group TrueSpeech"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_TRUESPEECH,
.priv_data_size = sizeof(TSContext),
.init = truespeech_decode_init,
FF_CODEC_DECODE_CB(truespeech_decode_frame),
.p.capabilities = AV_CODEC_CAP_DR1,
};
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