/* * 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 "libavutil/mem_internal.h" #include "avcodec.h" #include "bswapdsp.h" #include "get_bits.h" #include "internal.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, 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; 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 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 = AV_CODEC_CAP_DR1, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, };