0,0 → 1,366 |
/* |
* 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 "avcodec.h" |
#include "dsputil.h" |
#include "get_bits.h" |
#include "internal.h" |
|
#include "truespeech_data.h" |
/** |
* @file |
* TrueSpeech decoder. |
*/ |
|
/** |
* TrueSpeech decoder context |
*/ |
typedef struct { |
DSPContext dsp; |
/* 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; |
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ff_dsputil_init(&c->dsp, avctx); |
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return 0; |
} |
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static void truespeech_read_frame(TSContext *dec, const uint8_t *input) |
{ |
GetBitContext gb; |
|
dec->dsp.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); |
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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); |
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dec->offset1[1] = get_bits(&gb, 4); |
dec->pulseval[1] = get_bits(&gb, 14); |
dec->pulseval[0] = get_bits(&gb, 14); |
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dec->offset1[1] |= get_bits(&gb, 4) << 4; |
dec->pulseval[3] = get_bits(&gb, 14); |
dec->pulseval[2] = get_bits(&gb, 14); |
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dec->offset1[0] |= get_bits1(&gb); |
dec->pulsepos[0] = get_bits_long(&gb, 27); |
dec->pulseoff[0] = get_bits(&gb, 4); |
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dec->offset1[0] |= get_bits1(&gb) << 1; |
dec->pulsepos[1] = get_bits_long(&gb, 27); |
dec->pulseoff[1] = get_bits(&gb, 4); |
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dec->offset1[0] |= get_bits1(&gb) << 2; |
dec->pulsepos[2] = get_bits_long(&gb, 27); |
dec->pulseoff[2] = get_bits(&gb, 4); |
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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]) + |
(dec->cvector[j] << 15) + 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; |
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dec->filtval = dec->vector[0]; |
} |
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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--; |
} |
} |
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} |
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static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart) |
{ |
int i; |
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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]; |
} |
} |
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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] * ptr1[k]; |
sum = (sum + (out[i] << 12) + 0x800) >> 12; |
out[i] = av_clip(sum, -0x7FFE, 0x7FFE); |
for(k = 7; k > 0; k--) |
ptr0[k] = ptr0[k - 1]; |
ptr0[0] = out[i]; |
} |
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for(i = 0; i < 8; i++) |
t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15; |
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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] = ((out[i] << 12) - sum) >> 12; |
} |
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for(i = 0; i < 8; i++) |
t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15; |
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ptr0 = dec->tmp3; |
for(i = 0; i < 60; i++){ |
int sum = out[i] << 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); |
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sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum; |
sum = sum - (sum >> 3); |
out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE); |
} |
} |
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static void truespeech_save_prevvec(TSContext *c) |
{ |
int i; |
|
for(i = 0; i < 8; i++) |
c->prevfilt[i] = c->cvector[i]; |
} |
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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; |
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iterations = buf_size / 32; |
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if (!iterations) { |
av_log(avctx, AV_LOG_ERROR, |
"Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size); |
return -1; |
} |
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/* 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]; |
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memset(samples, 0, iterations * 240 * sizeof(*samples)); |
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for(j = 0; j < iterations; j++) { |
truespeech_read_frame(c, buf); |
buf += 32; |
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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; |
} |
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truespeech_save_prevvec(c); |
} |
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*got_frame_ptr = 1; |
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return buf_size; |
} |
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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 = CODEC_CAP_DR1, |
}; |