/*
  * DCA encoder

  * Copyright (C) 2008-2012 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/avassert.h"

 #include "libavutil/channel_layout.h"

 #include "libavutil/common.h"
 #include "avcodec.h"

 #include "dca.h"
 #include "dcadata.h"
 #include "dcaenc.h"

 #include "internal.h"

 #include "mathops.h"

 #include "put_bits.h"
 
 #define MAX_CHANNELS 6

 #define DCA_MAX_FRAME_SIZE 16384

 #define DCA_HEADER_SIZE 13

 #define DCA_LFE_SAMPLES 8

 #define DCAENC_SUBBANDS 32

 #define SUBFRAMES 1

 #define SUBSUBFRAMES 2
 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
 #define AUBANDS 25

 typedef struct DCAEncContext {

     PutBitContext pb;
     int frame_size;

     int frame_bits;
     int fullband_channels;
     int channels;

     int lfe_channel;

     int samplerate_index;
     int bitrate_index;
     int channel_config;
     const int32_t *band_interpolation;
     const int32_t *band_spectrum;

     int lfe_scale_factor;

     softfloat lfe_quant;
     int32_t lfe_peak_cb;
 
     int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */

     int32_t subband[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
     int32_t quantized[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
     int32_t peak_cb[DCAENC_SUBBANDS][MAX_CHANNELS];

     int32_t downsampled_lfe[DCA_LFE_SAMPLES];
     int32_t masking_curve_cb[SUBSUBFRAMES][256];

     int abits[DCAENC_SUBBANDS][MAX_CHANNELS];
     int scale_factor[DCAENC_SUBBANDS][MAX_CHANNELS];
     softfloat quant[DCAENC_SUBBANDS][MAX_CHANNELS];

     int32_t eff_masking_curve_cb[256];
     int32_t band_masking_cb[32];
     int32_t worst_quantization_noise;
     int32_t worst_noise_ever;
     int consumed_bits;

 } DCAEncContext;

 static int32_t cos_table[2048];
 static int32_t band_interpolation[2][512];
 static int32_t band_spectrum[2][8];
 static int32_t auf[9][AUBANDS][256];
 static int32_t cb_to_add[256];
 static int32_t cb_to_level[2048];
 static int32_t lfe_fir_64i[512];

 /* Transfer function of outer and middle ear, Hz -> dB */
 static double hom(double f)
 {
     double f1 = f / 1000;
 
     return -3.64 * pow(f1, -0.8)
            + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
            - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
            - 0.0006 * (f1 * f1) * (f1 * f1);
 }
 
 static double gammafilter(int i, double f)
 {
     double h = (f - fc[i]) / erb[i];
 
     h = 1 + h * h;
     h = 1 / (h * h);
     return 20 * log10(h);
 }
 
 static int encode_init(AVCodecContext *avctx)
 {

     DCAEncContext *c = avctx->priv_data;

     uint64_t layout = avctx->channel_layout;
     int i, min_frame_bits;
 
     c->fullband_channels = c->channels = avctx->channels;
     c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
     c->band_interpolation = band_interpolation[1];
     c->band_spectrum = band_spectrum[1];
     c->worst_quantization_noise = -2047;
     c->worst_noise_ever = -2047;
 
     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_MONO:         c->channel_config = 0; break;
     case AV_CH_LAYOUT_STEREO:       c->channel_config = 2; break;
     case AV_CH_LAYOUT_2_2:          c->channel_config = 8; break;
     case AV_CH_LAYOUT_5POINT0:      c->channel_config = 9; break;
     case AV_CH_LAYOUT_5POINT1:      c->channel_config = 9; break;
     default:
         av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");
         return AVERROR_PATCHWELCOME;
     }
 
     if (c->lfe_channel)
         c->fullband_channels--;
 
     for (i = 0; i < 9; i++) {
         if (sample_rates[i] == avctx->sample_rate)
             break;
     }
     if (i == 9)
         return AVERROR(EINVAL);
     c->samplerate_index = i;
 
     if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {
         av_log(avctx, AV_LOG_ERROR, "Bit rate %i not supported.", avctx->bit_rate);
         return AVERROR(EINVAL);
     }

     for (i = 0; ff_dca_bit_rates[i] < avctx->bit_rate; i++)

         ;
     c->bitrate_index = i;

     avctx->bit_rate = ff_dca_bit_rates[i];

     c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);
     min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;
     if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))
         return AVERROR(EINVAL);
 
     c->frame_size = (c->frame_bits + 7) / 8;
 
     avctx->frame_size = 32 * SUBBAND_SAMPLES;
 
     if (!cos_table[0]) {
         int j, k;
 
         for (i = 0; i < 2048; i++) {
             cos_table[i]   = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
             cb_to_level[i] = (int32_t)(0x7fffffff * pow(10, -0.005 * i));
         }
 

         /* FIXME: probably incorrect */
         for (i = 0; i < 256; i++) {

             lfe_fir_64i[i] = (int32_t)(0x01ffffff * ff_dca_lfe_fir_64[i]);
             lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * ff_dca_lfe_fir_64[i]);

         }
 

         for (i = 0; i < 512; i++) {

             band_interpolation[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
             band_interpolation[1][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_nonperfect[i]);

         }
 
         for (i = 0; i < 9; i++) {
             for (j = 0; j < AUBANDS; j++) {
                 for (k = 0; k < 256; k++) {
                     double freq = sample_rates[i] * (k + 0.5) / 512;
 
                     auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));
                 }
             }
         }
 
         for (i = 0; i < 256; i++) {
             double add = 1 + pow(10, -0.01 * i);
             cb_to_add[i] = (int32_t)(100 * log10(add));
         }
         for (j = 0; j < 8; j++) {
             double accum = 0;
             for (i = 0; i < 512; i++) {

                 double reconst = ff_dca_fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);

                 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
             }
             band_spectrum[0][j] = (int32_t)(200 * log10(accum));
         }
         for (j = 0; j < 8; j++) {
             double accum = 0;
             for (i = 0; i < 512; i++) {

                 double reconst = ff_dca_fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);

                 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
             }
             band_spectrum[1][j] = (int32_t)(200 * log10(accum));
         }
     }
     return 0;
 }
 
 static inline int32_t cos_t(int x)
 {
     return cos_table[x & 2047];
 }
 
 static inline int32_t sin_t(int x)
 {
     return cos_t(x - 512);
 }

 static inline int32_t half32(int32_t a)
 {
     return (a + 1) >> 1;
 }

 
 static inline int32_t mul32(int32_t a, int32_t b)
 {

     int64_t r = (int64_t)a * b + 0x80000000ULL;
     return r >> 32;
 }
 

 static void subband_transform(DCAEncContext *c, const int32_t *input)

 {
     int ch, subs, i, k, j;
 
     for (ch = 0; ch < c->fullband_channels; ch++) {
         /* History is copied because it is also needed for PSY */
         int32_t hist[512];
         int hist_start = 0;
 
         for (i = 0; i < 512; i++)
             hist[i] = c->history[i][ch];
 
         for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
             int32_t accum[64];
             int32_t resp;
             int band;
 
             /* Calculate the convolutions at once */
             for (i = 0; i < 64; i++)
                 accum[i] = 0;
 
             for (k = 0, i = hist_start, j = 0;
                     i < 512; k = (k + 1) & 63, i++, j++)
                 accum[k] += mul32(hist[i], c->band_interpolation[j]);
             for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)
                 accum[k] += mul32(hist[i], c->band_interpolation[j]);
 
             for (k = 16; k < 32; k++)
                 accum[k] = accum[k] - accum[31 - k];
             for (k = 32; k < 48; k++)
                 accum[k] = accum[k] + accum[95 - k];
 
             for (band = 0; band < 32; band++) {
                 resp = 0;
                 for (i = 16; i < 48; i++) {
                     int s = (2 * band + 1) * (2 * (i + 16) + 1);
                     resp += mul32(accum[i], cos_t(s << 3)) >> 3;
                 }
 
                 c->subband[subs][band][ch] = ((band + 1) & 2) ? -resp : resp;
             }
 
             /* Copy in 32 new samples from input */
             for (i = 0; i < 32; i++)
                 hist[i + hist_start] = input[(subs * 32 + i) * c->channels + ch];
             hist_start = (hist_start + 32) & 511;
         }
     }
 }
 

 static void lfe_downsample(DCAEncContext *c, const int32_t *input)

 {
     /* FIXME: make 128x LFE downsampling possible */
     int i, j, lfes;
     int32_t hist[512];
     int32_t accum;
     int hist_start = 0;
 
     for (i = 0; i < 512; i++)
         hist[i] = c->history[i][c->channels - 1];
 
     for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
         /* Calculate the convolution */
         accum = 0;
 
         for (i = hist_start, j = 0; i < 512; i++, j++)
             accum += mul32(hist[i], lfe_fir_64i[j]);
         for (i = 0; i < hist_start; i++, j++)
             accum += mul32(hist[i], lfe_fir_64i[j]);
 
         c->downsampled_lfe[lfes] = accum;
 
         /* Copy in 64 new samples from input */
         for (i = 0; i < 64; i++)
             hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + c->channels - 1];
 
         hist_start = (hist_start + 64) & 511;
     }

 }
 

 typedef struct {
     int32_t re;
     int32_t im;
 } cplx32;

 static void fft(const int32_t in[2 * 256], cplx32 out[256])

 {

     cplx32 buf[256], rin[256], rout[256];
     int i, j, k, l;
 
     /* do two transforms in parallel */
     for (i = 0; i < 256; i++) {
         /* Apply the Hann window */
         rin[i].re = mul32(in[2 * i], 0x3fffffff - (cos_t(8 * i + 2) >> 1));
         rin[i].im = mul32(in[2 * i + 1], 0x3fffffff - (cos_t(8 * i + 6) >> 1));
     }
     /* pre-rotation */
     for (i = 0; i < 256; i++) {
         buf[i].re = mul32(cos_t(4 * i + 2), rin[i].re)
                   - mul32(sin_t(4 * i + 2), rin[i].im);
         buf[i].im = mul32(cos_t(4 * i + 2), rin[i].im)
                   + mul32(sin_t(4 * i + 2), rin[i].re);
     }
 
     for (j = 256, l = 1; j != 1; j >>= 1, l <<= 1) {
         for (k = 0; k < 256; k += j) {
             for (i = k; i < k + j / 2; i++) {
                 cplx32 sum, diff;
                 int t = 8 * l * i;
 
                 sum.re = buf[i].re + buf[i + j / 2].re;
                 sum.im = buf[i].im + buf[i + j / 2].im;
 
                 diff.re = buf[i].re - buf[i + j / 2].re;
                 diff.im = buf[i].im - buf[i + j / 2].im;
 
                 buf[i].re = half32(sum.re);
                 buf[i].im = half32(sum.im);
 
                 buf[i + j / 2].re = mul32(diff.re, cos_t(t))
                                   - mul32(diff.im, sin_t(t));
                 buf[i + j / 2].im = mul32(diff.im, cos_t(t))
                                   + mul32(diff.re, sin_t(t));
             }
         }
     }
     /* post-rotation */
     for (i = 0; i < 256; i++) {
         int b = ff_reverse[i];
         rout[i].re = mul32(buf[b].re, cos_t(4 * i))
                    - mul32(buf[b].im, sin_t(4 * i));
         rout[i].im = mul32(buf[b].im, cos_t(4 * i))
                    + mul32(buf[b].re, sin_t(4 * i));
     }
     for (i = 0; i < 256; i++) {
         /* separate the results of the two transforms */
         cplx32 o1, o2;
 
         o1.re =  rout[i].re - rout[255 - i].re;
         o1.im =  rout[i].im + rout[255 - i].im;
 
         o2.re =  rout[i].im - rout[255 - i].im;
         o2.im = -rout[i].re - rout[255 - i].re;
 
         /* combine them into one long transform */
         out[i].re = mul32( o1.re + o2.re, cos_t(2 * i + 1))
                   + mul32( o1.im - o2.im, sin_t(2 * i + 1));
         out[i].im = mul32( o1.im + o2.im, cos_t(2 * i + 1))
                   + mul32(-o1.re + o2.re, sin_t(2 * i + 1));

     }

 }

 static int32_t get_cb(int32_t in)
 {
     int i, res;
 
     res = 0;
     if (in < 0)
         in = -in;
     for (i = 1024; i > 0; i >>= 1) {
         if (cb_to_level[i + res] >= in)
             res += i;

     }

     return -res;

 }
 

 static int32_t add_cb(int32_t a, int32_t b)

 {

     if (a < b)
         FFSWAP(int32_t, a, b);
 
     if (a - b >= 256)
         return a;
     return a + cb_to_add[a - b];

 }
 

 static void adjust_jnd(int samplerate_index,
                        const int32_t in[512], int32_t out_cb[256])

 {

     int32_t power[256];
     cplx32 out[256];
     int32_t out_cb_unnorm[256];
     int32_t denom;
     const int32_t ca_cb = -1114;
     const int32_t cs_cb = 928;
     int i, j;
 
     fft(in, out);

     for (j = 0; j < 256; j++) {
         power[j] = add_cb(get_cb(out[j].re), get_cb(out[j].im));
         out_cb_unnorm[j] = -2047; /* and can only grow */

     }

 
     for (i = 0; i < AUBANDS; i++) {
         denom = ca_cb; /* and can only grow */
         for (j = 0; j < 256; j++)
             denom = add_cb(denom, power[j] + auf[samplerate_index][i][j]);
         for (j = 0; j < 256; j++)
             out_cb_unnorm[j] = add_cb(out_cb_unnorm[j],
                     -denom + auf[samplerate_index][i][j]);
     }
 
     for (j = 0; j < 256; j++)
         out_cb[j] = add_cb(out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);

 }
 

 typedef void (*walk_band_t)(DCAEncContext *c, int band1, int band2, int f,

                             int32_t spectrum1, int32_t spectrum2, int channel,
                             int32_t * arg);
 

 static void walk_band_low(DCAEncContext *c, int band, int channel,

                           walk_band_t walk, int32_t *arg)

 {

     int f;

     if (band == 0) {
         for (f = 0; f < 4; f++)
             walk(c, 0, 0, f, 0, -2047, channel, arg);
     } else {
         for (f = 0; f < 8; f++)
             walk(c, band, band - 1, 8 * band - 4 + f,
                     c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);
     }
 }

 static void walk_band_high(DCAEncContext *c, int band, int channel,

                            walk_band_t walk, int32_t *arg)
 {
     int f;

     if (band == 31) {
         for (f = 0; f < 4; f++)
             walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);
     } else {
         for (f = 0; f < 8; f++)
             walk(c, band, band + 1, 8 * band + 4 + f,
                     c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);
     }
 }

 static void update_band_masking(DCAEncContext *c, int band1, int band2,

                                 int f, int32_t spectrum1, int32_t spectrum2,
                                 int channel, int32_t * arg)
 {
     int32_t value = c->eff_masking_curve_cb[f] - spectrum1;

     if (value < c->band_masking_cb[band1])
         c->band_masking_cb[band1] = value;
 }
 

 static void calc_masking(DCAEncContext *c, const int32_t *input)

 {
     int i, k, band, ch, ssf;
     int32_t data[512];
 
     for (i = 0; i < 256; i++)
         for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
             c->masking_curve_cb[ssf][i] = -2047;
 
     for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
         for (ch = 0; ch < c->fullband_channels; ch++) {
             for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
                 data[i] = c->history[k][ch];
             for (k -= 512; i < 512; i++, k++)
                 data[i] = input[k * c->channels + ch];
             adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);
         }
     for (i = 0; i < 256; i++) {
         int32_t m = 2048;
 
         for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
             if (c->masking_curve_cb[ssf][i] < m)
                 m = c->masking_curve_cb[ssf][i];
         c->eff_masking_curve_cb[i] = m;
     }
 
     for (band = 0; band < 32; band++) {
         c->band_masking_cb[band] = 2048;
         walk_band_low(c, band, 0, update_band_masking, NULL);
         walk_band_high(c, band, 0, update_band_masking, NULL);

     }
 }
 

 static void find_peaks(DCAEncContext *c)

 {

     int band, ch;
 
     for (band = 0; band < 32; band++)
         for (ch = 0; ch < c->fullband_channels; ch++) {
             int sample;
             int32_t m = 0;
 
             for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
                 int32_t s = abs(c->subband[sample][band][ch]);
                 if (m < s)
                     m = s;
             }
             c->peak_cb[band][ch] = get_cb(m);
         }
 
     if (c->lfe_channel) {
         int sample;
         int32_t m = 0;
 
         for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)
             if (m < abs(c->downsampled_lfe[sample]))
                 m = abs(c->downsampled_lfe[sample]);
         c->lfe_peak_cb = get_cb(m);
     }

 }
 

 static const int snr_fudge = 128;
 #define USED_1ABITS 1
 #define USED_NABITS 2
 #define USED_26ABITS 4
 

 static int init_quantization_noise(DCAEncContext *c, int noise)

 {

     int ch, band, ret = 0;
 
     c->consumed_bits = 132 + 493 * c->fullband_channels;
     if (c->lfe_channel)
         c->consumed_bits += 72;
 
     /* attempt to guess the bit distribution based on the prevoius frame */
     for (ch = 0; ch < c->fullband_channels; ch++) {
         for (band = 0; band < 32; band++) {
             int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;
 
             if (snr_cb >= 1312) {
                 c->abits[band][ch] = 26;
                 ret |= USED_26ABITS;
             } else if (snr_cb >= 222) {
                 c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);
                 ret |= USED_NABITS;
             } else if (snr_cb >= 0) {
                 c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);
                 ret |= USED_NABITS;
             } else {
                 c->abits[band][ch] = 1;
                 ret |= USED_1ABITS;
             }
         }
     }

     for (band = 0; band < 32; band++)
         for (ch = 0; ch < c->fullband_channels; ch++) {
             c->consumed_bits += bit_consumption[c->abits[band][ch]];
         }
 
     return ret;
 }
 

 static void assign_bits(DCAEncContext *c)

 {
     /* Find the bounds where the binary search should work */
     int low, high, down;
     int used_abits = 0;
 
     init_quantization_noise(c, c->worst_quantization_noise);
     low = high = c->worst_quantization_noise;
     if (c->consumed_bits > c->frame_bits) {
         while (c->consumed_bits > c->frame_bits) {
             av_assert0(used_abits != USED_1ABITS);
             low = high;
             high += snr_fudge;
             used_abits = init_quantization_noise(c, high);
         }
     } else {
         while (c->consumed_bits <= c->frame_bits) {
             high = low;
             if (used_abits == USED_26ABITS)
                 goto out; /* The requested bitrate is too high, pad with zeros */
             low -= snr_fudge;
             used_abits = init_quantization_noise(c, low);
         }
     }
 
     /* Now do a binary search between low and high to see what fits */
     for (down = snr_fudge >> 1; down; down >>= 1) {
         init_quantization_noise(c, high - down);
         if (c->consumed_bits <= c->frame_bits)
             high -= down;
     }
     init_quantization_noise(c, high);
 out:
     c->worst_quantization_noise = high;
     if (high > c->worst_noise_ever)
         c->worst_noise_ever = high;
 }
 

 static void shift_history(DCAEncContext *c, const int32_t *input)

 {
     int k, ch;
 
     for (k = 0; k < 512; k++)
         for (ch = 0; ch < c->channels; ch++)
             c->history[k][ch] = input[k * c->channels + ch];
 }
 
 static int32_t quantize_value(int32_t value, softfloat quant)
 {
     int32_t offset = 1 << (quant.e - 1);
 
     value = mul32(value, quant.m) + offset;
     value = value >> quant.e;
     return value;
 }
 
 static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
 {
     int32_t peak;
     int our_nscale, try_remove;
     softfloat our_quant;
 
     av_assert0(peak_cb <= 0);
     av_assert0(peak_cb >= -2047);
 
     our_nscale = 127;
     peak = cb_to_level[-peak_cb];
 
     for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
         if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
             continue;
         our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
         our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
         if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
             continue;
         our_nscale -= try_remove;
     }
 
     if (our_nscale >= 125)
         our_nscale = 124;
 
     quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
     quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
     av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
 
     return our_nscale;
 }
 

 static void calc_scales(DCAEncContext *c)

 {
     int band, ch;
 
     for (band = 0; band < 32; band++)
         for (ch = 0; ch < c->fullband_channels; ch++)
             c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],
                                                        c->abits[band][ch],
                                                        &c->quant[band][ch]);
 
     if (c->lfe_channel)
         c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);
 }
 

 static void quantize_all(DCAEncContext *c)

 {
     int sample, band, ch;
 
     for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
         for (band = 0; band < 32; band++)
             for (ch = 0; ch < c->fullband_channels; ch++)
                 c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);

 }
 

 static void put_frame_header(DCAEncContext *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, SUBBAND_SAMPLES - 1);

 
     /* Primary frame byte size */

     put_bits(&c->pb, 14, c->frame_size - 1);

     /* Audio channel arrangement */
     put_bits(&c->pb, 6, c->channel_config);

 
     /* Core audio sampling frequency */

     put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);

     /* Transmission bit rate */
     put_bits(&c->pb, 5, c->bitrate_index);

 
     /* 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 64x subsampling */
     put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);

 
     /* 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(DCAEncContext *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->fullband_channels - 1);

 
     /* Subband activity count */

     for (ch = 0; ch < c->fullband_channels; ch++)

         put_bits(&c->pb, 5, DCAENC_SUBBANDS - 2);

 
     /* High frequency VQ start subband */

     for (ch = 0; ch < c->fullband_channels; ch++)

         put_bits(&c->pb, 5, DCAENC_SUBBANDS - 1);

 
     /* Joint intensity coding index: 0, 0 */

     for (ch = 0; ch < c->fullband_channels; ch++)

         put_bits(&c->pb, 3, 0);
 
     /* Transient mode codebook: A4, A4 (arbitrary) */

     for (ch = 0; ch < c->fullband_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->fullband_channels; ch++)

         put_bits(&c->pb, 3, 6);
 
     /* Bit allocation quantizer select: linear 5-bit */

     for (ch = 0; ch < c->fullband_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->fullband_channels; ch++)

             put_bits(&c->pb, bitlen[i], thr[i]);
 
     /* Scale factor adjustment index: not transmitted */

     /* Audio header CRC check word: not transmitted */

 }
 

 static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)

 {

     if (c->abits[band][ch] <= 7) {
         int sum, i, j;
         for (i = 0; i < 8; i += 4) {
             sum = 0;
             for (j = 3; j >= 0; j--) {
                 sum *= quant_levels[c->abits[band][ch]];
                 sum += c->quantized[ss * 8 + i + j][band][ch];
                 sum += (quant_levels[c->abits[band][ch]] - 1) / 2;
             }
             put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);
         }
     } else {
         int i;
         for (i = 0; i < 8; i++) {
             int bits = bit_consumption[c->abits[band][ch]] / 16;

             put_sbits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch]);

         }

     }
 }
 

 static void put_subframe(DCAEncContext *c, int subframe)

 {

     int i, band, ss, ch;

 
     /* 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->fullband_channels; ch++)

         for (band = 0; band < DCAENC_SUBBANDS; band++)

             put_bits(&c->pb, 1, 0);
 

     /* Prediction VQ address: not transmitted */

     /* Bit allocation index */

     for (ch = 0; ch < c->fullband_channels; ch++)

         for (band = 0; band < DCAENC_SUBBANDS; band++)

             put_bits(&c->pb, 5, c->abits[band][ch]);

 
     if (SUBSUBFRAMES > 1) {
         /* Transition mode: none for each channel and subband */

         for (ch = 0; ch < c->fullband_channels; ch++)

             for (band = 0; band < DCAENC_SUBBANDS; band++)

                 put_bits(&c->pb, 1, 0); /* codebook A4 */
     }
 

     /* Scale factors */
     for (ch = 0; ch < c->fullband_channels; ch++)

         for (band = 0; band < DCAENC_SUBBANDS; band++)

             put_bits(&c->pb, 7, c->scale_factor[band][ch]);

 
     /* 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: 8 samples and scalefactor */

     if (c->lfe_channel) {

         for (i = 0; i < DCA_LFE_SAMPLES; i++)
             put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);

         put_bits(&c->pb, 8, c->lfe_scale_factor);
     }
 
     /* Audio data (subsubframes) */
     for (ss = 0; ss < SUBSUBFRAMES ; ss++)

         for (ch = 0; ch < c->fullband_channels; ch++)

             for (band = 0; band < DCAENC_SUBBANDS; band++)

                     put_subframe_samples(c, ss, band, ch);

 
     /* DSYNC */
     put_bits(&c->pb, 16, 0xffff);
 }
 

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

 {

     DCAEncContext *c = avctx->priv_data;

     const int32_t *samples;
     int ret, i;

     if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size , 0)) < 0)

         return ret;
 

     samples = (const int32_t *)frame->data[0];

     subband_transform(c, samples);
     if (c->lfe_channel)
         lfe_downsample(c, samples);

     calc_masking(c, samples);
     find_peaks(c);
     assign_bits(c);
     calc_scales(c);
     quantize_all(c);
     shift_history(c, samples);

     init_put_bits(&c->pb, avpkt->data, avpkt->size);
     put_frame_header(c);
     put_primary_audio_header(c);
     for (i = 0; i < SUBFRAMES; i++)
         put_subframe(c, i);

 
     for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++)
         put_bits(&c->pb, 1, 0);
 

     flush_put_bits(&c->pb);

     avpkt->pts      = frame->pts;
     avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
     avpkt->size     = c->frame_size + 1;
     *got_packet_ptr = 1;

     return 0;
 }
 

 static const AVCodecDefault defaults[] = {
     { "b",          "1411200" },
     { NULL },
 };
 

 AVCodec ff_dca_encoder = {

     .name                  = "dca",

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

     .type                  = AVMEDIA_TYPE_AUDIO,
     .id                    = AV_CODEC_ID_DTS,

     .priv_data_size        = sizeof(DCAEncContext),

     .init                  = encode_init,
     .encode2               = encode_frame,

     .capabilities          = AV_CODEC_CAP_EXPERIMENTAL,

     .sample_fmts           = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
                                                             AV_SAMPLE_FMT_NONE },
     .supported_samplerates = sample_rates,
     .channel_layouts       = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
                                                   AV_CH_LAYOUT_STEREO,
                                                   AV_CH_LAYOUT_2_2,
                                                   AV_CH_LAYOUT_5POINT0,
                                                   AV_CH_LAYOUT_5POINT1,
                                                   0 },
     .defaults              = defaults,

 };