|
- /*
- * DSP Group TrueSpeech compatible decoder
- * Copyright (c) 2005 Konstantin Shishkov
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; 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 "bitstream.h"
- #include "bswapdsp.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->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;
-
- ff_bswapdsp_init(&c->bdsp);
-
- return 0;
- }
-
- static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
- {
- BitstreamContext bc;
-
- dec->bdsp.bswap_buf((uint32_t *) dec->buffer, (const uint32_t *) input, 8);
- bitstream_init8(&bc, dec->buffer, 32);
-
- dec->vector[7] = ts_codebook[7][bitstream_read(&bc, 3)];
- dec->vector[6] = ts_codebook[6][bitstream_read(&bc, 3)];
- dec->vector[5] = ts_codebook[5][bitstream_read(&bc, 3)];
- dec->vector[4] = ts_codebook[4][bitstream_read(&bc, 4)];
- dec->vector[3] = ts_codebook[3][bitstream_read(&bc, 4)];
- dec->vector[2] = ts_codebook[2][bitstream_read(&bc, 4)];
- dec->vector[1] = ts_codebook[1][bitstream_read(&bc, 5)];
- dec->vector[0] = ts_codebook[0][bitstream_read(&bc, 5)];
- dec->flag = bitstream_read_bit(&bc);
-
- dec->offset1[0] = bitstream_read(&bc, 4) << 4;
- dec->offset2[3] = bitstream_read(&bc, 7);
- dec->offset2[2] = bitstream_read(&bc, 7);
- dec->offset2[1] = bitstream_read(&bc, 7);
- dec->offset2[0] = bitstream_read(&bc, 7);
-
- dec->offset1[1] = bitstream_read(&bc, 4);
- dec->pulseval[1] = bitstream_read(&bc, 14);
- dec->pulseval[0] = bitstream_read(&bc, 14);
-
- dec->offset1[1] |= bitstream_read(&bc, 4) << 4;
- dec->pulseval[3] = bitstream_read(&bc, 14);
- dec->pulseval[2] = bitstream_read(&bc, 14);
-
- dec->offset1[0] |= bitstream_read_bit(&bc);
- dec->pulsepos[0] = bitstream_read(&bc, 27);
- dec->pulseoff[0] = bitstream_read(&bc, 4);
-
- dec->offset1[0] |= bitstream_read_bit(&bc) << 1;
- dec->pulsepos[1] = bitstream_read(&bc, 27);
- dec->pulseoff[1] = bitstream_read(&bc, 4);
-
- dec->offset1[0] |= bitstream_read_bit(&bc) << 2;
- dec->pulsepos[2] = bitstream_read(&bc, 27);
- dec->pulseoff[2] = bitstream_read(&bc, 4);
-
- dec->offset1[0] |= bitstream_read_bit(&bc) << 3;
- dec->pulsepos[3] = bitstream_read(&bc, 27);
- dec->pulseoff[3] = bitstream_read(&bc, 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;
-
- 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] * 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];
- }
-
- 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] = ((out[i] << 12) - 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] << 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) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
- 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;
- }
-
- 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,
- };
|