TrueSpeech compatible audio decoder by Konstantin Shishkov

Originally committed as revision 4803 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Diego Biurrun 19 years ago
parent 27c748b576
commit bf3027c87b
  1. 1
      Changelog
  2. 3
      libavcodec/Makefile
  3. 3
      libavcodec/allcodecs.c
  4. 2
      libavcodec/avcodec.h
  5. 377
      libavcodec/truespeech.c
  6. 136
      libavcodec/truespeech_data.h
  7. 1
      libavformat/wav.c

@ -26,6 +26,7 @@ version <next>
- ffserver fixed, it should now be usable again
- QDM2 audio decoder
- cook audio decoder
- TrueSpeech audio decoder
- wma2 audio decoder fixed, now all files should play correctly
- JPEG-LS decoder (unfinished)
- build system improvements

@ -161,6 +161,9 @@ endif
ifeq ($(CONFIG_TRUEMOTION2_DECODER),yes)
OBJS+= truemotion2.o
endif
ifeq ($(CONFIG_TRUESPEECH_DECODER),yes)
OBJS+= truespeech.o
endif
ifeq ($(CONFIG_TSCC_DECODER),yes)
OBJS+= tscc.o
endif

@ -497,6 +497,9 @@ void avcodec_register_all(void)
#ifdef CONFIG_COOK_DECODER
register_avcodec(&cook_decoder);
#endif //CONFIG_COOK_DECODER
#ifdef CONFIG_TRUESPEECH_DECODER
register_avcodec(&truespeech_decoder);
#endif //CONFIG_TRUESPEECH_DECODER
#ifdef CONFIG_RAWVIDEO_DECODER
register_avcodec(&rawvideo_decoder);
#endif //CONFIG_RAWVIDEO_DECODER

@ -189,6 +189,7 @@ enum CodecID {
CODEC_ID_GSM,
CODEC_ID_QDM2,
CODEC_ID_COOK,
CODEC_ID_TRUESPEECH,
CODEC_ID_OGGTHEORA= 0x16000,
@ -2134,6 +2135,7 @@ extern AVCodec mp3adu_decoder;
extern AVCodec mp3on4_decoder;
extern AVCodec qdm2_decoder;
extern AVCodec cook_decoder;
extern AVCodec truespeech_decoder;
extern AVCodec mace3_decoder;
extern AVCodec mace6_decoder;
extern AVCodec huffyuv_decoder;

@ -0,0 +1,377 @@
/*
* DSP Group TrueSpeech compatible decoder
* Copyright (c) 2005 Konstantin Shishkov
*
* This library 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 of the License, or (at your option) any later version.
*
* This library 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 this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "avcodec.h"
#include "truespeech_data.h"
/**
* @file truespeech.c
* TrueSpeech decoder.
*/
/**
* TrueSpeech decoder context
*/
typedef struct {
/* input data */
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 int truespeech_decode_init(AVCodecContext * avctx)
{
// TSContext *c = avctx->priv_data;
return 0;
}
static void truespeech_read_frame(TSContext *dec, uint8_t *input)
{
uint32_t t;
/* first dword */
t = LE_32(input);
input += 4;
dec->flag = t & 1;
dec->vector[0] = ts_codebook[0][(t >> 1) & 0x1F];
dec->vector[1] = ts_codebook[1][(t >> 6) & 0x1F];
dec->vector[2] = ts_codebook[2][(t >> 11) & 0xF];
dec->vector[3] = ts_codebook[3][(t >> 15) & 0xF];
dec->vector[4] = ts_codebook[4][(t >> 19) & 0xF];
dec->vector[5] = ts_codebook[5][(t >> 23) & 0x7];
dec->vector[6] = ts_codebook[6][(t >> 26) & 0x7];
dec->vector[7] = ts_codebook[7][(t >> 29) & 0x7];
/* second dword */
t = LE_32(input);
input += 4;
dec->offset2[0] = (t >> 0) & 0x7F;
dec->offset2[1] = (t >> 7) & 0x7F;
dec->offset2[2] = (t >> 14) & 0x7F;
dec->offset2[3] = (t >> 21) & 0x7F;
dec->offset1[0] = ((t >> 28) & 0xF) << 4;
/* third dword */
t = LE_32(input);
input += 4;
dec->pulseval[0] = (t >> 0) & 0x3FFF;
dec->pulseval[1] = (t >> 14) & 0x3FFF;
dec->offset1[1] = (t >> 28) & 0x0F;
/* fourth dword */
t = LE_32(input);
input += 4;
dec->pulseval[2] = (t >> 0) & 0x3FFF;
dec->pulseval[3] = (t >> 14) & 0x3FFF;
dec->offset1[1] |= ((t >> 28) & 0x0F) << 4;
/* fifth dword */
t = LE_32(input);
input += 4;
dec->pulsepos[0] = (t >> 4) & 0x7FFFFFF;
dec->pulseoff[0] = (t >> 0) & 0xF;
dec->offset1[0] |= (t >> 31) & 1;
/* sixth dword */
t = LE_32(input);
input += 4;
dec->pulsepos[1] = (t >> 4) & 0x7FFFFFF;
dec->pulseoff[1] = (t >> 0) & 0xF;
dec->offset1[0] |= ((t >> 31) & 1) << 1;
/* seventh dword */
t = LE_32(input);
input += 4;
dec->pulsepos[2] = (t >> 4) & 0x7FFFFFF;
dec->pulseoff[2] = (t >> 0) & 0xF;
dec->offset1[0] |= ((t >> 31) & 1) << 2;
/* eighth dword */
t = LE_32(input);
input += 4;
dec->pulsepos[3] = (t >> 4) & 0x7FFFFFF;
dec->pulseoff[3] = (t >> 0) & 0xF;
dec->offset1[0] |= ((t >> 31) & 1) << 3;
}
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 * 2);
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_230[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, *filter;
int i, t, off;
t = dec->offset2[quart];
if(t == 127){
memset(dec->newvec, 0, 60 * 2);
return;
}
for(i = 0; i < 146; i++)
tmp[i] = dec->filtbuf[i];
off = (t / 25) + dec->offset1[quart >> 1] + 18;
ptr0 = tmp + 145 - off;
ptr1 = tmp + 146;
filter = ts_240 + (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;
int16_t *ptr1, *ptr2;
int coef;
memset(out, 0, 60 * 2);
for(i = 0; i < 7; i++) {
t = dec->pulseval[quart] & 3;
dec->pulseval[quart] >>= 2;
tmp[6 - i] = ts_562[dec->pulseoff[quart] * 4 + t];
}
coef = dec->pulsepos[quart] >> 15;
ptr1 = ts_140 + 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_140;
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;
for(i = 0; i < 86; i++)
dec->filtbuf[i] = dec->filtbuf[i + 60];
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] * ptr1[k];
sum = (sum + (out[i] << 12) + 0x800) >> 12;
out[i] = 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_5E2[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] = ((out[i] << 12) - sum) >> 12;
}
for(i = 0; i < 8; i++)
t[i] = (ts_5F2[i] * ptr1[i]) >> 15;
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] = clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
sum = sum - (sum >> 3);
out[i] = 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 *data_size,
uint8_t *buf, int buf_size)
{
TSContext *c = avctx->priv_data;
int i;
short *samples = data;
int consumed = 0;
int16_t out_buf[240];
if (!buf_size)
return 0;
while (consumed < buf_size) {
truespeech_read_frame(c, buf + consumed);
consumed += 32;
truespeech_correlate_filter(c);
truespeech_filters_merge(c);
memset(out_buf, 0, 240 * 2);
for(i = 0; i < 4; i++) {
truespeech_apply_twopoint_filter(c, i);
truespeech_place_pulses(c, out_buf + i * 60, i);
truespeech_update_filters(c, out_buf + i * 60, i);
truespeech_synth(c, out_buf + i * 60, i);
}
truespeech_save_prevvec(c);
/* finally output decoded frame */
for(i = 0; i < 240; i++)
*samples++ = out_buf[i];
}
*data_size = consumed * 15;
return buf_size;
}
AVCodec truespeech_decoder = {
"truespeech",
CODEC_TYPE_AUDIO,
CODEC_ID_TRUESPEECH,
sizeof(TSContext),
truespeech_decode_init,
NULL,
NULL,
truespeech_decode_frame,
};

@ -0,0 +1,136 @@
#ifndef __TRUESPEECH_DATA__
#define __TRUESPEECH_DATA__
/* codebooks fo expanding input filter */
const int16_t ts_cb_0[32] = {
0x8240, 0x8364, 0x84CE, 0x865D, 0x8805, 0x89DE, 0x8BD7, 0x8DF4,
0x9051, 0x92E2, 0x95DE, 0x990F, 0x9C81, 0xA079, 0xA54C, 0xAAD2,
0xB18A, 0xB90A, 0xC124, 0xC9CC, 0xD339, 0xDDD3, 0xE9D6, 0xF893,
0x096F, 0x1ACA, 0x29EC, 0x381F, 0x45F9, 0x546A, 0x63C3, 0x73B5,
};
const int16_t ts_cb_1[32] = {
0x9F65, 0xB56B, 0xC583, 0xD371, 0xE018, 0xEBB4, 0xF61C, 0xFF59,
0x085B, 0x1106, 0x1952, 0x214A, 0x28C9, 0x2FF8, 0x36E6, 0x3D92,
0x43DF, 0x49BB, 0x4F46, 0x5467, 0x5930, 0x5DA3, 0x61EC, 0x65F9,
0x69D4, 0x6D5A, 0x709E, 0x73AD, 0x766B, 0x78F0, 0x7B5A, 0x7DA5,
};
const int16_t ts_cb_2[16] = {
0x96F8, 0xA3B4, 0xAF45, 0xBA53, 0xC4B1, 0xCECC, 0xD86F, 0xE21E,
0xEBF3, 0xF640, 0x00F7, 0x0C20, 0x1881, 0x269A, 0x376B, 0x4D60,
};
const int16_t ts_cb_3[16] = {
0xC654, 0xDEF2, 0xEFAA, 0xFD94, 0x096A, 0x143F, 0x1E7B, 0x282C,
0x3176, 0x3A89, 0x439F, 0x4CA2, 0x557F, 0x5E50, 0x6718, 0x6F8D,
};
const int16_t ts_cb_4[16] = {
0xABE7, 0xBBA8, 0xC81C, 0xD326, 0xDD0E, 0xE5D4, 0xEE22, 0xF618,
0xFE28, 0x064F, 0x0EB7, 0x17B8, 0x21AA, 0x2D8B, 0x3BA2, 0x4DF9,
};
const int16_t ts_cb_5[8] = {
0xD51B, 0xF12E, 0x042E, 0x13C7, 0x2260, 0x311B, 0x40DE, 0x5385,
};
const int16_t ts_cb_6[8] = {
0xB550, 0xC825, 0xD980, 0xE997, 0xF883, 0x0752, 0x1811, 0x2E18,
};
const int16_t ts_cb_7[8] = {
0xCEF0, 0xE4F9, 0xF6BB, 0x0646, 0x14F5, 0x23FF, 0x356F, 0x4A8D,
};
const int16_t *ts_codebook[8] = {
ts_cb_0, ts_cb_1, ts_cb_2, ts_cb_3, ts_cb_4, ts_cb_5, ts_cb_6, ts_cb_7
};
/* table used for decoding pulse positions */
const int16_t ts_140[120] = {
0x0E46, 0x0CCC, 0x0B6D, 0x0A28, 0x08FC, 0x07E8, 0x06EB, 0x0604,
0x0532, 0x0474, 0x03C9, 0x0330, 0x02A8, 0x0230, 0x01C7, 0x016C,
0x011E, 0x00DC, 0x00A5, 0x0078, 0x0054, 0x0038, 0x0023, 0x0014,
0x000A, 0x0004, 0x0001, 0x0000, 0x0000, 0x0000,
0x0196, 0x017A, 0x015F, 0x0145, 0x012C, 0x0114, 0x00FD, 0x00E7,
0x00D2, 0x00BE, 0x00AB, 0x0099, 0x0088, 0x0078, 0x0069, 0x005B,
0x004E, 0x0042, 0x0037, 0x002D, 0x0024, 0x001C, 0x0015, 0x000F,
0x000A, 0x0006, 0x0003, 0x0001, 0x0000, 0x0000,
0x001D, 0x001C, 0x001B, 0x001A, 0x0019, 0x0018, 0x0017, 0x0016,
0x0015, 0x0014, 0x0013, 0x0012, 0x0011, 0x0010, 0x000F, 0x000E,
0x000D, 0x000C, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006,
0x0005, 0x0004, 0x0003, 0x0002, 0x0001, 0x0000,
0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001,
0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001,
0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001,
0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001
};
/* filter for correlated input filter */
const int16_t ts_230[8] =
{ 0x7F3B, 0x7E78, 0x7DB6, 0x7CF5, 0x7C35, 0x7B76, 0x7AB8, 0x79FC };
/* two-point filters table */
const int16_t ts_240[25 * 2] = {
0xED2F, 0x5239,
0x54F1, 0xE4A9,
0x2620, 0xEE3E,
0x09D6, 0x2C40,
0xEFB5, 0x2BE0,
0x3FE1, 0x3339,
0x442F, 0xE6FE,
0x4458, 0xF9DF,
0xF231, 0x43DB,
0x3DB0, 0xF705,
0x4F7B, 0xFEFB,
0x26AD, 0x0CDC,
0x33C2, 0x0739,
0x12BE, 0x43A2,
0x1BDF, 0x1F3E,
0x0211, 0x0796,
0x2AEB, 0x163F,
0x050D, 0x3A38,
0x0D1E, 0x0D78,
0x150F, 0x3346,
0x38A4, 0x0B7D,
0x2D5D, 0x1FDF,
0x19B7, 0x2822,
0x0D99, 0x1F12,
0x194C, 0x0CE6
};
/* possible pulse values */
const int16_t ts_562[64] = {
0x0002, 0x0006, 0xFFFE, 0xFFFA,
0x0004, 0x000C, 0xFFFC, 0xFFF4,
0x0006, 0x0012, 0xFFFA, 0xFFEE,
0x000A, 0x001E, 0xFFF6, 0xFFE2,
0x0010, 0x0030, 0xFFF0, 0xFFD0,
0x0019, 0x004B, 0xFFE7, 0xFFB5,
0x0028, 0x0078, 0xFFD8, 0xFF88,
0x0040, 0x00C0, 0xFFC0, 0xFF40,
0x0065, 0x012F, 0xFF9B, 0xFED1,
0x00A1, 0x01E3, 0xFF5F, 0xFE1D,
0x0100, 0x0300, 0xFF00, 0xFD00,
0x0196, 0x04C2, 0xFE6A, 0xFB3E,
0x0285, 0x078F, 0xFD7B, 0xF871,
0x0400, 0x0C00, 0xFC00, 0xF400,
0x0659, 0x130B, 0xF9A7, 0xECF5,
0x0A14, 0x1E3C, 0xF5EC, 0xE1C4
};
/* filters used in final output calculations */
const int16_t ts_5E2[8] =
{ 0x4666, 0x26B8, 0x154C, 0x0BB6, 0x0671, 0x038B, 0x01F3, 0x0112 };
const int16_t ts_5F2[8] =
{ 0x6000, 0x4800, 0x3600, 0x2880, 0x1E60, 0x16C8, 0x1116, 0x0CD1 };
#endif

@ -44,6 +44,7 @@ const CodecTag codec_wav_tags[] = {
{ CODEC_ID_SONIC_LS, 0x2048 },
{ CODEC_ID_ADPCM_CT, 0x200 },
{ CODEC_ID_ADPCM_SWF, ('S'<<8)+'F' },
{ CODEC_ID_TRUESPEECH, 0x22 },
{ 0, 0 },
};

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