/*
  * DCA encoder
  * Copyright (C) 2008 Alexander E. Patrakov
  *               2010 Benjamin Larsson
  *               2011 Xiang Wang
  *
  * 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/common.h"
 #include "libavutil/avassert.h"
 #include "avcodec.h"

 #include "internal.h"

 #include "put_bits.h"
 #include "dcaenc.h"
 #include "dcadata.h"

 #include "dca.h"

 
 #undef NDEBUG
 
 #define MAX_CHANNELS 6
 #define DCA_SUBBANDS_32 32
 #define DCA_MAX_FRAME_SIZE 16383
 #define DCA_HEADER_SIZE 13
 
 #define DCA_SUBBANDS 32 ///< Subband activity count
 #define QUANTIZER_BITS 16
 #define SUBFRAMES 1
 #define SUBSUBFRAMES 4
 #define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
 #define LFE_BITS 8
 #define LFE_INTERPOLATION 64
 #define LFE_PRESENT 2
 #define LFE_MISSING 0
 
 static const int8_t dca_lfe_index[] = {
     1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
 };
 
 static const int8_t dca_channel_reorder_lfe[][9] = {
     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1,  2, -1, -1, -1, -1, -1 },
     { 1,  2,  0, -1,  3, -1, -1, -1, -1 },
     { 0,  1, -1,  2,  3, -1, -1, -1, -1 },
     { 1,  2,  0, -1,  3,  4, -1, -1, -1 },
     { 2,  3, -1,  0,  1,  4,  5, -1, -1 },
     { 1,  2,  0, -1,  3,  4,  5, -1, -1 },
     { 0, -1,  4,  5,  2,  3,  1, -1, -1 },
     { 3,  4,  1, -1,  0,  2,  5,  6, -1 },
     { 2,  3, -1,  5,  7,  0,  1,  4,  6 },
     { 3,  4,  1, -1,  0,  2,  5,  7,  6 },
 };
 
 static const int8_t dca_channel_reorder_nolfe[][9] = {
     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
     { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },
     { 1,  2,  0,  3, -1, -1, -1, -1, -1 },
     { 0,  1,  2,  3, -1, -1, -1, -1, -1 },
     { 1,  2,  0,  3,  4, -1, -1, -1, -1 },
     { 2,  3,  0,  1,  4,  5, -1, -1, -1 },
     { 1,  2,  0,  3,  4,  5, -1, -1, -1 },
     { 0,  4,  5,  2,  3,  1, -1, -1, -1 },
     { 3,  4,  1,  0,  2,  5,  6, -1, -1 },
     { 2,  3,  5,  7,  0,  1,  4,  6, -1 },
     { 3,  4,  1,  0,  2,  5,  7,  6, -1 },
 };
 
 typedef struct {
     PutBitContext pb;
     int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
     int start[MAX_CHANNELS];
     int frame_size;
     int prim_channels;
     int lfe_channel;
     int sample_rate_code;
     int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
     int lfe_scale_factor;
     int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
 
     int a_mode;                         ///< audio channels arrangement
     int num_channel;
     int lfe_state;
     int lfe_offset;
     const int8_t *channel_order_tab;    ///< channel reordering table, lfe and non lfe
 
     int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
     int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
 } DCAContext;
 
 static int32_t cos_table[128];
 
 static inline int32_t mul32(int32_t a, int32_t b)
 {
     int64_t r = (int64_t) a * b;
     /* round the result before truncating - improves accuracy */
     return (r + 0x80000000) >> 32;
 }
 
 /* Integer version of the cosine modulated Pseudo QMF */
 
 static void qmf_init(void)
 {
     int i;
     int32_t c[17], s[17];
     s[0] = 0;           /* sin(index * PI / 64) * 0x7fffffff */
     c[0] = 0x7fffffff;  /* cos(index * PI / 64) * 0x7fffffff */
 
     for (i = 1; i <= 16; i++) {
         s[i] = 2 * (mul32(c[i - 1], 105372028)  + mul32(s[i - 1], 2144896908));
         c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
     }
 
     for (i = 0; i < 16; i++) {
         cos_table[i      ]  =  c[i]      >> 3; /* avoid output overflow */
         cos_table[i +  16]  =  s[16 - i] >> 3;
         cos_table[i +  32]  = -s[i]      >> 3;
         cos_table[i +  48]  = -c[16 - i] >> 3;
         cos_table[i +  64]  = -c[i]      >> 3;
         cos_table[i +  80]  = -s[16 - i] >> 3;
         cos_table[i +  96]  =  s[i]      >> 3;
         cos_table[i + 112]  =  c[16 - i] >> 3;
     }
 }
 
 static int32_t band_delta_factor(int band, int sample_num)
 {
     int index = band * (2 * sample_num + 1);
     if (band == 0)
         return 0x07ffffff;
     else
         return cos_table[index & 127];
 }
 
 static void add_new_samples(DCAContext *c, const int32_t *in,
                             int count, int channel)
 {
     int i;
 
     /* Place new samples into the history buffer */
     for (i = 0; i < count; i++) {
         c->history[channel][c->start[channel] + i] = in[i];
         av_assert0(c->start[channel] + i < 512);
     }
     c->start[channel] += count;
     if (c->start[channel] == 512)
         c->start[channel] = 0;
     av_assert0(c->start[channel] < 512);
 }
 
 static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
                           int channel)
 {
     int band, i, j, k;
     int32_t resp;
     int32_t accum[DCA_SUBBANDS_32] = {0};
 
     add_new_samples(c, in, DCA_SUBBANDS_32, channel);
 
     /* Calculate the dot product of the signal with the (possibly inverted)
        reference decoder's response to this vector:
        (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
        so that -1.0 cancels 1.0 from the previous step */
 
     for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
         accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
     for (i = 0; i < c->start[channel]; k++, j++, i++)
         accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
 
     resp = 0;
     /* TODO: implement FFT instead of this naive calculation */
     for (band = 0; band < DCA_SUBBANDS_32; band++) {
         for (j = 0; j < 32; j++)
             resp += mul32(accum[j], band_delta_factor(band, j));
 
         out[band] = (band & 2) ? (-resp) : resp;
     }
 }
 
 static int32_t lfe_fir_64i[512];
 static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
 {
     int i, j;
     int channel = c->prim_channels;
     int32_t accum = 0;
 
     add_new_samples(c, in, LFE_INTERPOLATION, channel);
     for (i = c->start[channel], j = 0; i < 512; i++, j++)
         accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
     for (i = 0; i < c->start[channel]; i++, j++)
         accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
     return accum;
 }
 
 static void init_lfe_fir(void)
 {
     static int initialized = 0;
     int i;
     if (initialized)
         return;
 
     for (i = 0; i < 512; i++)
         lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
     initialized = 1;
 }
 
 static void put_frame_header(DCAContext *c)
 {
     /* SYNC */
     put_bits(&c->pb, 16, 0x7ffe);
     put_bits(&c->pb, 16, 0x8001);
 
     /* Frame type: normal */
     put_bits(&c->pb, 1, 1);
 
     /* Deficit sample count: none */
     put_bits(&c->pb, 5, 31);
 
     /* CRC is not present */
     put_bits(&c->pb, 1, 0);
 
     /* Number of PCM sample blocks */
     put_bits(&c->pb, 7, PCM_SAMPLES-1);
 
     /* Primary frame byte size */
     put_bits(&c->pb, 14, c->frame_size-1);
 
     /* Audio channel arrangement: L + R (stereo) */
     put_bits(&c->pb, 6, c->num_channel);
 
     /* Core audio sampling frequency */
     put_bits(&c->pb, 4, c->sample_rate_code);
 
     /* Transmission bit rate: 1411.2 kbps */
     put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
 
     /* Embedded down mix: disabled */
     put_bits(&c->pb, 1, 0);
 
     /* Embedded dynamic range flag: not present */
     put_bits(&c->pb, 1, 0);
 
     /* Embedded time stamp flag: not present */
     put_bits(&c->pb, 1, 0);
 
     /* Auxiliary data flag: not present */
     put_bits(&c->pb, 1, 0);
 
     /* HDCD source: no */
     put_bits(&c->pb, 1, 0);
 
     /* Extension audio ID: N/A */
     put_bits(&c->pb, 3, 0);
 
     /* Extended audio data: not present */
     put_bits(&c->pb, 1, 0);
 
     /* Audio sync word insertion flag: after each sub-frame */
     put_bits(&c->pb, 1, 0);
 
     /* Low frequency effects flag: not present or interpolation factor=64 */
     put_bits(&c->pb, 2, c->lfe_state);
 
     /* Predictor history switch flag: on */
     put_bits(&c->pb, 1, 1);
 
     /* No CRC */
     /* Multirate interpolator switch: non-perfect reconstruction */
     put_bits(&c->pb, 1, 0);
 
     /* Encoder software revision: 7 */
     put_bits(&c->pb, 4, 7);
 
     /* Copy history: 0 */
     put_bits(&c->pb, 2, 0);
 
     /* Source PCM resolution: 16 bits, not DTS ES */
     put_bits(&c->pb, 3, 0);
 
     /* Front sum/difference coding: no */
     put_bits(&c->pb, 1, 0);
 
     /* Surrounds sum/difference coding: no */
     put_bits(&c->pb, 1, 0);
 
     /* Dialog normalization: 0 dB */
     put_bits(&c->pb, 4, 0);
 }
 
 static void put_primary_audio_header(DCAContext *c)
 {
     static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
     static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
 
     int ch, i;
     /* Number of subframes */
     put_bits(&c->pb, 4, SUBFRAMES - 1);
 
     /* Number of primary audio channels */
     put_bits(&c->pb, 3, c->prim_channels - 1);
 
     /* Subband activity count */
     for (ch = 0; ch < c->prim_channels; ch++)
         put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
 
     /* High frequency VQ start subband */
     for (ch = 0; ch < c->prim_channels; ch++)
         put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
 
     /* Joint intensity coding index: 0, 0 */
     for (ch = 0; ch < c->prim_channels; ch++)
         put_bits(&c->pb, 3, 0);
 
     /* Transient mode codebook: A4, A4 (arbitrary) */
     for (ch = 0; ch < c->prim_channels; ch++)
         put_bits(&c->pb, 2, 0);
 
     /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
     for (ch = 0; ch < c->prim_channels; ch++)
         put_bits(&c->pb, 3, 6);
 
     /* Bit allocation quantizer select: linear 5-bit */
     for (ch = 0; ch < c->prim_channels; ch++)
         put_bits(&c->pb, 3, 6);
 
     /* Quantization index codebook select: dummy data
        to avoid transmission of scale factor adjustment */
 
     for (i = 1; i < 11; i++)
         for (ch = 0; ch < c->prim_channels; ch++)
             put_bits(&c->pb, bitlen[i], thr[i]);
 
     /* Scale factor adjustment index: not transmitted */
 }
 
 /**
  * 8-23 bits quantization
  * @param sample
  * @param bits
  */
 static inline uint32_t quantize(int32_t sample, int bits)
 {
     av_assert0(sample <    1 << (bits - 1));
     av_assert0(sample >= -(1 << (bits - 1)));

     return sample & ((1 << bits) - 1);

 }
 
 static inline int find_scale_factor7(int64_t max_value, int bits)
 {
     int i = 0, j = 128, q;
     max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
     while (i < j) {
         q = (i + j) >> 1;
         if (max_value < scale_factor_quant7[q])
             j = q;
         else
             i = q + 1;
     }
     av_assert1(i < 128);
     return i;
 }
 
 static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
                                int scale_factor)
 {
     sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
     put_bits(&c->pb, bits, quantize((int) sample, bits));
 }
 
 static void put_subframe(DCAContext *c,
                          int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
                          int subframe)
 {
     int i, sub, ss, ch, max_value;
     int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
 
     /* Subsubframes count */
     put_bits(&c->pb, 2, SUBSUBFRAMES -1);
 
     /* Partial subsubframe sample count: dummy */
     put_bits(&c->pb, 3, 0);
 
     /* Prediction mode: no ADPCM, in each channel and subband */
     for (ch = 0; ch < c->prim_channels; ch++)
         for (sub = 0; sub < DCA_SUBBANDS; sub++)
             put_bits(&c->pb, 1, 0);
 
     /* Prediction VQ addres: not transmitted */
     /* Bit allocation index */
     for (ch = 0; ch < c->prim_channels; ch++)
         for (sub = 0; sub < DCA_SUBBANDS; sub++)
             put_bits(&c->pb, 5, QUANTIZER_BITS+3);
 
     if (SUBSUBFRAMES > 1) {
         /* Transition mode: none for each channel and subband */
         for (ch = 0; ch < c->prim_channels; ch++)
             for (sub = 0; sub < DCA_SUBBANDS; sub++)
                 put_bits(&c->pb, 1, 0); /* codebook A4 */
     }
 
     /* Determine scale_factor */
     for (ch = 0; ch < c->prim_channels; ch++)
         for (sub = 0; sub < DCA_SUBBANDS; sub++) {
             max_value = 0;
             for (i = 0; i < 8 * SUBSUBFRAMES; i++)
                 max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
             c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
         }
 
     if (c->lfe_channel) {
         max_value = 0;
         for (i = 0; i < 4 * SUBSUBFRAMES; i++)
             max_value = FFMAX(max_value, FFABS(lfe_data[i]));
         c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
     }
 
     /* Scale factors: the same for each channel and subband,
        encoded according to Table D.1.2 */
     for (ch = 0; ch < c->prim_channels; ch++)
         for (sub = 0; sub < DCA_SUBBANDS; sub++)
             put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
 
     /* Joint subband scale factor codebook select: not transmitted */
     /* Scale factors for joint subband coding: not transmitted */
     /* Stereo down-mix coefficients: not transmitted */
     /* Dynamic range coefficient: not transmitted */
     /* Stde information CRC check word: not transmitted */
     /* VQ encoded high frequency subbands: not transmitted */
 
     /* LFE data */
     if (c->lfe_channel) {
         for (i = 0; i < 4 * SUBSUBFRAMES; i++)
             put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
         put_bits(&c->pb, 8, c->lfe_scale_factor);
     }
 
     /* Audio data (subsubframes) */
 
     for (ss = 0; ss < SUBSUBFRAMES ; ss++)
         for (ch = 0; ch < c->prim_channels; ch++)
             for (sub = 0; sub < DCA_SUBBANDS; sub++)
                 for (i = 0; i < 8; i++)
                     put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
 
     /* DSYNC */
     put_bits(&c->pb, 16, 0xffff);
 }
 
 static void put_frame(DCAContext *c,
                       int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
                       uint8_t *frame)
 {
     int i;
     init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
 
     put_primary_audio_header(c);
     for (i = 0; i < SUBFRAMES; i++)
         put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
 
     flush_put_bits(&c->pb);
     c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
 
     init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
     put_frame_header(c);
     flush_put_bits(&c->pb);
 }
 

 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                         const AVFrame *frame, int *got_packet_ptr)

 {
     int i, k, channel;
     DCAContext *c = avctx->priv_data;

     const int16_t *samples;
     int ret, real_channel = 0;

     if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)) < 0)

         return ret;
 
     samples = (const int16_t *)frame->data[0];

     for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
         for (channel = 0; channel < c->prim_channels + 1; channel++) {
             real_channel = c->channel_order_tab[channel];
             if (real_channel >= 0) {

                 /* Get 32 PCM samples */
                 for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
                     c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
                 }
                 /* Put subband samples into the proper place */

                 qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
             }
         }
     }
 
     if (c->lfe_channel) {
         for (i = 0; i < PCM_SAMPLES / 2; i++) {
             for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
                 c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
             c->lfe_data[i] = lfe_downsample(c, c->pcm);
         }
     }
 

     put_frame(c, c->subband, avpkt->data);

     avpkt->size     = c->frame_size;
     *got_packet_ptr = 1;
     return 0;

 }
 
 static int encode_init(AVCodecContext *avctx)
 {
     DCAContext *c = avctx->priv_data;
     int i;

     uint64_t layout = avctx->channel_layout;

 
     c->prim_channels = avctx->channels;
     c->lfe_channel   = (avctx->channels == 3 || avctx->channels == 6);
 

     if (!layout) {
         av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
                                       "encoder will guess the layout, but it "
                                       "might be incorrect.\n");
         layout = av_get_default_channel_layout(avctx->channels);
     }
     switch (layout) {

     case AV_CH_LAYOUT_STEREO:       c->a_mode = 2; c->num_channel = 2; break;
     case AV_CH_LAYOUT_5POINT0:      c->a_mode = 9; c->num_channel = 9; break;
     case AV_CH_LAYOUT_5POINT1:      c->a_mode = 9; c->num_channel = 9; break;
     case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
     case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
     default:
     av_log(avctx, AV_LOG_ERROR,
            "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
     return AVERROR_PATCHWELCOME;
     }
 
     if (c->lfe_channel) {
         init_lfe_fir();
         c->prim_channels--;
         c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
         c->lfe_state         = LFE_PRESENT;
         c->lfe_offset        = dca_lfe_index[c->a_mode];
     } else {
         c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
         c->lfe_state         = LFE_MISSING;
     }
 
     for (i = 0; i < 16; i++) {

         if (avpriv_dca_sample_rates[i] && (avpriv_dca_sample_rates[i] == avctx->sample_rate))

             break;
     }
     if (i == 16) {
         av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
         for (i = 0; i < 16; i++)

             av_log(avctx, AV_LOG_ERROR, "%d, ", avpriv_dca_sample_rates[i]);

         av_log(avctx, AV_LOG_ERROR, "supported.\n");
         return -1;
     }
     c->sample_rate_code = i;
 
     avctx->frame_size = 32 * PCM_SAMPLES;
 
     if (!cos_table[127])
         qmf_init();
     return 0;
 }
 
 AVCodec ff_dca_encoder = {
     .name           = "dca",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_DTS,

     .priv_data_size = sizeof(DCAContext),
     .init           = encode_init,

     .encode2        = encode_frame,

     .capabilities   = CODEC_CAP_EXPERIMENTAL,

     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                      AV_SAMPLE_FMT_NONE },

     .long_name      = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

 };