/*
  * 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/intreadwrite.h"

 #include "avcodec.h"

 #include "dsputil.h"
 #include "get_bits.h"

 
 #include "truespeech_data.h"
 /**

  * @file

  * TrueSpeech decoder.
  */
 
 /**
  * TrueSpeech decoder context
  */
 typedef struct {

     AVFrame frame;

     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) {
         av_log_ask_for_sample(avctx, "Unsupported channel count: %d\n", avctx->channels);
         return AVERROR(EINVAL);
     }

     avctx->sample_fmt = AV_SAMPLE_FMT_S16;

     ff_dsputil_init(&c->dsp, avctx);

     avcodec_get_frame_defaults(&c->frame);
     avctx->coded_frame = &c->frame;
 

     return 0;
 }
 

 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);
 
     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]) +
                                    (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 = (const int16_t*)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 = (const int16_t*)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 = (const int16_t*)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)

 {

     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 */
     c->frame.nb_samples = iterations * 240;
     if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;

     }

     samples = (int16_t *)c->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;
     *(AVFrame *)data = c->frame;

     return buf_size;

 }
 

 AVCodec ff_truespeech_decoder = {

     .name           = "truespeech",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = CODEC_ID_TRUESPEECH,
     .priv_data_size = sizeof(TSContext),
     .init           = truespeech_decode_init,
     .decode         = truespeech_decode_frame,

     .capabilities   = CODEC_CAP_DR1,

     .long_name      = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),

 };