Mercurial > libavcodec.hg
view truespeech.c @ 3365:84f29207af3a libavcodec
flacenc - rice param search patch by (Justin Ruggles jruggle earthlink net
1) search for optimal rice parameters and partition order. i also
modified the stereo method estimation to use this to calculate estimated
bit count instead of using just the pure sums.
2) search for the best fixed prediction order
3) constant subframe mode (good for encoding silence)
Note that the regression test for the decoded wav file also changed.
This is due to FFmpeg's FLAC decoder truncating the file, which it did
before anyway...just at a different cutoff point. The generated FLAC
files are still 100% lossless.
With this update, FFmpeg's FLAC encoder has speed and compression
somewhere between "flac -1" and "flac -2". On my machine, it's about
15% faster than "flac -2", and about 10% slower than "flac -1". The
encoding parameters are identical to "flac -2" (fixed predictors, 1152
blocksize, partition order 0 to 3).
author | michael |
---|---|
date | Tue, 27 Jun 2006 21:01:21 +0000 |
parents | 82277c821113 |
children | c8c591fe26f8 |
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/* * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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; const int16_t *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 = (const int16_t*)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; const int16_t *ptr1; int16_t *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 = (const int16_t*)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 = (const int16_t*)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, };