mirror of https://github.com/FFmpeg/FFmpeg.git
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
367 lines
11 KiB
367 lines
11 KiB
/* |
|
* 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, |
|
};
|
|
|