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FFmpeg/libavcodec/dcaenc.c

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  1. /*

  2.  * DCA encoder

  3.  * Copyright (C) 2008-2012 Alexander E. Patrakov

  4.  *               2010 Benjamin Larsson

  5.  *               2011 Xiang Wang

  6.  *

  7.  * This file is part of FFmpeg.

  8.  *

  9.  * FFmpeg is free software; you can redistribute it and/or

  10.  * modify it under the terms of the GNU Lesser General Public

  11.  * License as published by the Free Software Foundation; either

  12.  * version 2.1 of the License, or (at your option) any later version.

  13.  *

  14.  * FFmpeg is distributed in the hope that it will be useful,

  15.  * but WITHOUT ANY WARRANTY; without even the implied warranty of

  16.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

  17.  * Lesser General Public License for more details.

  18.  *

  19.  * You should have received a copy of the GNU Lesser General Public

  20.  * License along with FFmpeg; if not, write to the Free Software

  21.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

  22.  */

  23. 

  24. #include "libavutil/avassert.h"

  25. #include "libavutil/channel_layout.h"

  26. #include "libavutil/common.h"

  27. #include "avcodec.h"

  28. #include "dca.h"

  29. #include "dcadata.h"

  30. #include "dcaenc.h"

  31. #include "internal.h"

  32. #include "mathops.h"

  33. #include "put_bits.h"

  34. 

  35. #define MAX_CHANNELS 6

  36. #define DCA_MAX_FRAME_SIZE 16384

  37. #define DCA_HEADER_SIZE 13

  38. #define DCA_LFE_SAMPLES 8

  39. 

  40. #define DCAENC_SUBBANDS 32

  41. #define SUBFRAMES 1

  42. #define SUBSUBFRAMES 2

  43. #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)

  44. #define AUBANDS 25

  45. 

  46. typedef struct DCAEncContext {

  47.     PutBitContext pb;

  48.     int frame_size;

  49.     int frame_bits;

  50.     int fullband_channels;

  51.     int channels;

  52.     int lfe_channel;

  53.     int samplerate_index;

  54.     int bitrate_index;

  55.     int channel_config;

  56.     const int32_t *band_interpolation;

  57.     const int32_t *band_spectrum;

  58.     int lfe_scale_factor;

  59.     softfloat lfe_quant;

  60.     int32_t lfe_peak_cb;

  61. 

  62.     int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */

  63.     int32_t subband[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];

  64.     int32_t quantized[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];

  65.     int32_t peak_cb[DCAENC_SUBBANDS][MAX_CHANNELS];

  66.     int32_t downsampled_lfe[DCA_LFE_SAMPLES];

  67.     int32_t masking_curve_cb[SUBSUBFRAMES][256];

  68.     int abits[DCAENC_SUBBANDS][MAX_CHANNELS];

  69.     int scale_factor[DCAENC_SUBBANDS][MAX_CHANNELS];

  70.     softfloat quant[DCAENC_SUBBANDS][MAX_CHANNELS];

  71.     int32_t eff_masking_curve_cb[256];

  72.     int32_t band_masking_cb[32];

  73.     int32_t worst_quantization_noise;

  74.     int32_t worst_noise_ever;

  75.     int consumed_bits;

  76. } DCAEncContext;

  77. 

  78. static int32_t cos_table[2048];

  79. static int32_t band_interpolation[2][512];

  80. static int32_t band_spectrum[2][8];

  81. static int32_t auf[9][AUBANDS][256];

  82. static int32_t cb_to_add[256];

  83. static int32_t cb_to_level[2048];

  84. static int32_t lfe_fir_64i[512];

  85. 

  86. /* Transfer function of outer and middle ear, Hz -> dB */

  87. static double hom(double f)

  88. {

  89.     double f1 = f / 1000;

  90. 

  91.     return -3.64 * pow(f1, -0.8)

  92.            + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))

  93.            - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))

  94.            - 0.0006 * (f1 * f1) * (f1 * f1);

  95. }

  96. 

  97. static double gammafilter(int i, double f)

  98. {

  99.     double h = (f - fc[i]) / erb[i];

  100. 

  101.     h = 1 + h * h;

  102.     h = 1 / (h * h);

  103.     return 20 * log10(h);

  104. }

  105. 

  106. static int encode_init(AVCodecContext *avctx)

  107. {

  108.     DCAEncContext *c = avctx->priv_data;

  109.     uint64_t layout = avctx->channel_layout;

  110.     int i, min_frame_bits;

  111. 

  112.     c->fullband_channels = c->channels = avctx->channels;

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

  114.     c->band_interpolation = band_interpolation[1];

  115.     c->band_spectrum = band_spectrum[1];

  116.     c->worst_quantization_noise = -2047;

  117.     c->worst_noise_ever = -2047;

  118. 

  119.     if (!layout) {

  120.         av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "

  121.                                       "encoder will guess the layout, but it "

  122.                                       "might be incorrect.\n");

  123.         layout = av_get_default_channel_layout(avctx->channels);

  124.     }

  125.     switch (layout) {

  126.     case AV_CH_LAYOUT_MONO:         c->channel_config = 0; break;

  127.     case AV_CH_LAYOUT_STEREO:       c->channel_config = 2; break;

  128.     case AV_CH_LAYOUT_2_2:          c->channel_config = 8; break;

  129.     case AV_CH_LAYOUT_5POINT0:      c->channel_config = 9; break;

  130.     case AV_CH_LAYOUT_5POINT1:      c->channel_config = 9; break;

  131.     default:

  132.         av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");

  133.         return AVERROR_PATCHWELCOME;

  134.     }

  135. 

  136.     if (c->lfe_channel)

  137.         c->fullband_channels--;

  138. 

  139.     for (i = 0; i < 9; i++) {

  140.         if (sample_rates[i] == avctx->sample_rate)

  141.             break;

  142.     }

  143.     if (i == 9)

  144.         return AVERROR(EINVAL);

  145.     c->samplerate_index = i;

  146. 

  147.     if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {

  148.         av_log(avctx, AV_LOG_ERROR, "Bit rate %i not supported.", avctx->bit_rate);

  149.         return AVERROR(EINVAL);

  150.     }

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

  152.         ;

  153.     c->bitrate_index = i;

  154.     avctx->bit_rate = ff_dca_bit_rates[i];

  155.     c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);

  156.     min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;

  157.     if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))

  158.         return AVERROR(EINVAL);

  159. 

  160.     c->frame_size = (c->frame_bits + 7) / 8;

  161. 

  162.     avctx->frame_size = 32 * SUBBAND_SAMPLES;

  163. 

  164.     if (!cos_table[0]) {

  165.         int j, k;

  166. 

  167.         for (i = 0; i < 2048; i++) {

  168.             cos_table[i]   = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));

  169.             cb_to_level[i] = (int32_t)(0x7fffffff * pow(10, -0.005 * i));

  170.         }

  171. 

  172.         /* FIXME: probably incorrect */

  173.         for (i = 0; i < 256; i++) {

  174.             lfe_fir_64i[i] = (int32_t)(0x01ffffff * ff_dca_lfe_fir_64[i]);

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

  176.         }

  177. 

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

  179.             band_interpolation[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);

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

  181.         }

  182. 

  183.         for (i = 0; i < 9; i++) {

  184.             for (j = 0; j < AUBANDS; j++) {

  185.                 for (k = 0; k < 256; k++) {

  186.                     double freq = sample_rates[i] * (k + 0.5) / 512;

  187. 

  188.                     auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));

  189.                 }

  190.             }

  191.         }

  192. 

  193.         for (i = 0; i < 256; i++) {

  194.             double add = 1 + pow(10, -0.01 * i);

  195.             cb_to_add[i] = (int32_t)(100 * log10(add));

  196.         }

  197.         for (j = 0; j < 8; j++) {

  198.             double accum = 0;

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

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

  201.                 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);

  202.             }

  203.             band_spectrum[0][j] = (int32_t)(200 * log10(accum));

  204.         }

  205.         for (j = 0; j < 8; j++) {

  206.             double accum = 0;

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

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

  209.                 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);

  210.             }

  211.             band_spectrum[1][j] = (int32_t)(200 * log10(accum));

  212.         }

  213.     }

  214.     return 0;

  215. }

  216. 

  217. static inline int32_t cos_t(int x)

  218. {

  219.     return cos_table[x & 2047];

  220. }

  221. 

  222. static inline int32_t sin_t(int x)

  223. {

  224.     return cos_t(x - 512);

  225. }

  226. 

  227. static inline int32_t half32(int32_t a)

  228. {

  229.     return (a + 1) >> 1;

  230. }

  231. 

  232. static inline int32_t mul32(int32_t a, int32_t b)

  233. {

  234.     int64_t r = (int64_t)a * b + 0x80000000ULL;

  235.     return r >> 32;

  236. }

  237. 

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

  239. {

  240.     int ch, subs, i, k, j;

  241. 

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

  243.         /* History is copied because it is also needed for PSY */

  244.         int32_t hist[512];

  245.         int hist_start = 0;

  246. 

  247.         for (i = 0; i < 512; i++)

  248.             hist[i] = c->history[i][ch];

  249. 

  250.         for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {

  251.             int32_t accum[64];

  252.             int32_t resp;

  253.             int band;

  254. 

  255.             /* Calculate the convolutions at once */

  256.             for (i = 0; i < 64; i++)

  257.                 accum[i] = 0;

  258. 

  259.             for (k = 0, i = hist_start, j = 0;

  260.                     i < 512; k = (k + 1) & 63, i++, j++)

  261.                 accum[k] += mul32(hist[i], c->band_interpolation[j]);

  262.             for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)

  263.                 accum[k] += mul32(hist[i], c->band_interpolation[j]);

  264. 

  265.             for (k = 16; k < 32; k++)

  266.                 accum[k] = accum[k] - accum[31 - k];

  267.             for (k = 32; k < 48; k++)

  268.                 accum[k] = accum[k] + accum[95 - k];

  269. 

  270.             for (band = 0; band < 32; band++) {

  271.                 resp = 0;

  272.                 for (i = 16; i < 48; i++) {

  273.                     int s = (2 * band + 1) * (2 * (i + 16) + 1);

  274.                     resp += mul32(accum[i], cos_t(s << 3)) >> 3;

  275.                 }

  276. 

  277.                 c->subband[subs][band][ch] = ((band + 1) & 2) ? -resp : resp;

  278.             }

  279. 

  280.             /* Copy in 32 new samples from input */

  281.             for (i = 0; i < 32; i++)

  282.                 hist[i + hist_start] = input[(subs * 32 + i) * c->channels + ch];

  283.             hist_start = (hist_start + 32) & 511;

  284.         }

  285.     }

  286. }

  287. 

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

  289. {

  290.     /* FIXME: make 128x LFE downsampling possible */

  291.     int i, j, lfes;

  292.     int32_t hist[512];

  293.     int32_t accum;

  294.     int hist_start = 0;

  295. 

  296.     for (i = 0; i < 512; i++)

  297.         hist[i] = c->history[i][c->channels - 1];

  298. 

  299.     for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {

  300.         /* Calculate the convolution */

  301.         accum = 0;

  302. 

  303.         for (i = hist_start, j = 0; i < 512; i++, j++)

  304.             accum += mul32(hist[i], lfe_fir_64i[j]);

  305.         for (i = 0; i < hist_start; i++, j++)

  306.             accum += mul32(hist[i], lfe_fir_64i[j]);

  307. 

  308.         c->downsampled_lfe[lfes] = accum;

  309. 

  310.         /* Copy in 64 new samples from input */

  311.         for (i = 0; i < 64; i++)

  312.             hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + c->channels - 1];

  313. 

  314.         hist_start = (hist_start + 64) & 511;

  315.     }

  316. }

  317. 

  318. typedef struct {

  319.     int32_t re;

  320.     int32_t im;

  321. } cplx32;

  322. 

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

  324. {

  325.     cplx32 buf[256], rin[256], rout[256];

  326.     int i, j, k, l;

  327. 

  328.     /* do two transforms in parallel */

  329.     for (i = 0; i < 256; i++) {

  330.         /* Apply the Hann window */

  331.         rin[i].re = mul32(in[2 * i], 0x3fffffff - (cos_t(8 * i + 2) >> 1));

  332.         rin[i].im = mul32(in[2 * i + 1], 0x3fffffff - (cos_t(8 * i + 6) >> 1));

  333.     }

  334.     /* pre-rotation */

  335.     for (i = 0; i < 256; i++) {

  336.         buf[i].re = mul32(cos_t(4 * i + 2), rin[i].re)

  337.                   - mul32(sin_t(4 * i + 2), rin[i].im);

  338.         buf[i].im = mul32(cos_t(4 * i + 2), rin[i].im)

  339.                   + mul32(sin_t(4 * i + 2), rin[i].re);

  340.     }

  341. 

  342.     for (j = 256, l = 1; j != 1; j >>= 1, l <<= 1) {

  343.         for (k = 0; k < 256; k += j) {

  344.             for (i = k; i < k + j / 2; i++) {

  345.                 cplx32 sum, diff;

  346.                 int t = 8 * l * i;

  347. 

  348.                 sum.re = buf[i].re + buf[i + j / 2].re;

  349.                 sum.im = buf[i].im + buf[i + j / 2].im;

  350. 

  351.                 diff.re = buf[i].re - buf[i + j / 2].re;

  352.                 diff.im = buf[i].im - buf[i + j / 2].im;

  353. 

  354.                 buf[i].re = half32(sum.re);

  355.                 buf[i].im = half32(sum.im);

  356. 

  357.                 buf[i + j / 2].re = mul32(diff.re, cos_t(t))

  358.                                   - mul32(diff.im, sin_t(t));

  359.                 buf[i + j / 2].im = mul32(diff.im, cos_t(t))

  360.                                   + mul32(diff.re, sin_t(t));

  361.             }

  362.         }

  363.     }

  364.     /* post-rotation */

  365.     for (i = 0; i < 256; i++) {

  366.         int b = ff_reverse[i];

  367.         rout[i].re = mul32(buf[b].re, cos_t(4 * i))

  368.                    - mul32(buf[b].im, sin_t(4 * i));

  369.         rout[i].im = mul32(buf[b].im, cos_t(4 * i))

  370.                    + mul32(buf[b].re, sin_t(4 * i));

  371.     }

  372.     for (i = 0; i < 256; i++) {

  373.         /* separate the results of the two transforms */

  374.         cplx32 o1, o2;

  375. 

  376.         o1.re =  rout[i].re - rout[255 - i].re;

  377.         o1.im =  rout[i].im + rout[255 - i].im;

  378. 

  379.         o2.re =  rout[i].im - rout[255 - i].im;

  380.         o2.im = -rout[i].re - rout[255 - i].re;

  381. 

  382.         /* combine them into one long transform */

  383.         out[i].re = mul32( o1.re + o2.re, cos_t(2 * i + 1))

  384.                   + mul32( o1.im - o2.im, sin_t(2 * i + 1));

  385.         out[i].im = mul32( o1.im + o2.im, cos_t(2 * i + 1))

  386.                   + mul32(-o1.re + o2.re, sin_t(2 * i + 1));

  387.     }

  388. }

  389. 

  390. static int32_t get_cb(int32_t in)

  391. {

  392.     int i, res;

  393. 

  394.     res = 0;

  395.     if (in < 0)

  396.         in = -in;

  397.     for (i = 1024; i > 0; i >>= 1) {

  398.         if (cb_to_level[i + res] >= in)

  399.             res += i;

  400.     }

  401.     return -res;

  402. }

  403. 

  404. static int32_t add_cb(int32_t a, int32_t b)

  405. {

  406.     if (a < b)

  407.         FFSWAP(int32_t, a, b);

  408. 

  409.     if (a - b >= 256)

  410.         return a;

  411.     return a + cb_to_add[a - b];

  412. }

  413. 

  414. static void adjust_jnd(int samplerate_index,

  415.                        const int32_t in[512], int32_t out_cb[256])

  416. {

  417.     int32_t power[256];

  418.     cplx32 out[256];

  419.     int32_t out_cb_unnorm[256];

  420.     int32_t denom;

  421.     const int32_t ca_cb = -1114;

  422.     const int32_t cs_cb = 928;

  423.     int i, j;

  424. 

  425.     fft(in, out);

  426. 

  427.     for (j = 0; j < 256; j++) {

  428.         power[j] = add_cb(get_cb(out[j].re), get_cb(out[j].im));

  429.         out_cb_unnorm[j] = -2047; /* and can only grow */

  430.     }

  431. 

  432.     for (i = 0; i < AUBANDS; i++) {

  433.         denom = ca_cb; /* and can only grow */

  434.         for (j = 0; j < 256; j++)

  435.             denom = add_cb(denom, power[j] + auf[samplerate_index][i][j]);

  436.         for (j = 0; j < 256; j++)

  437.             out_cb_unnorm[j] = add_cb(out_cb_unnorm[j],

  438.                     -denom + auf[samplerate_index][i][j]);

  439.     }

  440. 

  441.     for (j = 0; j < 256; j++)

  442.         out_cb[j] = add_cb(out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);

  443. }

  444. 

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

  446.                             int32_t spectrum1, int32_t spectrum2, int channel,

  447.                             int32_t * arg);

  448. 

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

  450.                           walk_band_t walk, int32_t *arg)

  451. {

  452.     int f;

  453. 

  454.     if (band == 0) {

  455.         for (f = 0; f < 4; f++)

  456.             walk(c, 0, 0, f, 0, -2047, channel, arg);

  457.     } else {

  458.         for (f = 0; f < 8; f++)

  459.             walk(c, band, band - 1, 8 * band - 4 + f,

  460.                     c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);

  461.     }

  462. }

  463. 

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

  465.                            walk_band_t walk, int32_t *arg)

  466. {

  467.     int f;

  468. 

  469.     if (band == 31) {

  470.         for (f = 0; f < 4; f++)

  471.             walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);

  472.     } else {

  473.         for (f = 0; f < 8; f++)

  474.             walk(c, band, band + 1, 8 * band + 4 + f,

  475.                     c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);

  476.     }

  477. }

  478. 

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

  480.                                 int f, int32_t spectrum1, int32_t spectrum2,

  481.                                 int channel, int32_t * arg)

  482. {

  483.     int32_t value = c->eff_masking_curve_cb[f] - spectrum1;

  484. 

  485.     if (value < c->band_masking_cb[band1])

  486.         c->band_masking_cb[band1] = value;

  487. }

  488. 

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

  490. {

  491.     int i, k, band, ch, ssf;

  492.     int32_t data[512];

  493. 

  494.     for (i = 0; i < 256; i++)

  495.         for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)

  496.             c->masking_curve_cb[ssf][i] = -2047;

  497. 

  498.     for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)

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

  500.             for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)

  501.                 data[i] = c->history[k][ch];

  502.             for (k -= 512; i < 512; i++, k++)

  503.                 data[i] = input[k * c->channels + ch];

  504.             adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);

  505.         }

  506.     for (i = 0; i < 256; i++) {

  507.         int32_t m = 2048;

  508. 

  509.         for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)

  510.             if (c->masking_curve_cb[ssf][i] < m)

  511.                 m = c->masking_curve_cb[ssf][i];

  512.         c->eff_masking_curve_cb[i] = m;

  513.     }

  514. 

  515.     for (band = 0; band < 32; band++) {

  516.         c->band_masking_cb[band] = 2048;

  517.         walk_band_low(c, band, 0, update_band_masking, NULL);

  518.         walk_band_high(c, band, 0, update_band_masking, NULL);

  519.     }

  520. }

  521. 

  522. static void find_peaks(DCAEncContext *c)

  523. {

  524.     int band, ch;

  525. 

  526.     for (band = 0; band < 32; band++)

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

  528.             int sample;

  529.             int32_t m = 0;

  530. 

  531.             for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {

  532.                 int32_t s = abs(c->subband[sample][band][ch]);

  533.                 if (m < s)

  534.                     m = s;

  535.             }

  536.             c->peak_cb[band][ch] = get_cb(m);

  537.         }

  538. 

  539.     if (c->lfe_channel) {

  540.         int sample;

  541.         int32_t m = 0;

  542. 

  543.         for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)

  544.             if (m < abs(c->downsampled_lfe[sample]))

  545.                 m = abs(c->downsampled_lfe[sample]);

  546.         c->lfe_peak_cb = get_cb(m);

  547.     }

  548. }

  549. 

  550. static const int snr_fudge = 128;

  551. #define USED_1ABITS 1

  552. #define USED_NABITS 2

  553. #define USED_26ABITS 4

  554. 

  555. static int init_quantization_noise(DCAEncContext *c, int noise)

  556. {

  557.     int ch, band, ret = 0;

  558. 

  559.     c->consumed_bits = 132 + 493 * c->fullband_channels;

  560.     if (c->lfe_channel)

  561.         c->consumed_bits += 72;

  562. 

  563.     /* attempt to guess the bit distribution based on the prevoius frame */

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

  565.         for (band = 0; band < 32; band++) {

  566.             int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;

  567. 

  568.             if (snr_cb >= 1312) {

  569.                 c->abits[band][ch] = 26;

  570.                 ret |= USED_26ABITS;

  571.             } else if (snr_cb >= 222) {

  572.                 c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);

  573.                 ret |= USED_NABITS;

  574.             } else if (snr_cb >= 0) {

  575.                 c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);

  576.                 ret |= USED_NABITS;

  577.             } else {

  578.                 c->abits[band][ch] = 1;

  579.                 ret |= USED_1ABITS;

  580.             }

  581.         }

  582.     }

  583. 

  584.     for (band = 0; band < 32; band++)

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

  586.             c->consumed_bits += bit_consumption[c->abits[band][ch]];

  587.         }

  588. 

  589.     return ret;

  590. }

  591. 

  592. static void assign_bits(DCAEncContext *c)

  593. {

  594.     /* Find the bounds where the binary search should work */

  595.     int low, high, down;

  596.     int used_abits = 0;

  597. 

  598.     init_quantization_noise(c, c->worst_quantization_noise);

  599.     low = high = c->worst_quantization_noise;

  600.     if (c->consumed_bits > c->frame_bits) {

  601.         while (c->consumed_bits > c->frame_bits) {

  602.             av_assert0(used_abits != USED_1ABITS);

  603.             low = high;

  604.             high += snr_fudge;

  605.             used_abits = init_quantization_noise(c, high);

  606.         }

  607.     } else {

  608.         while (c->consumed_bits <= c->frame_bits) {

  609.             high = low;

  610.             if (used_abits == USED_26ABITS)

  611.                 goto out; /* The requested bitrate is too high, pad with zeros */

  612.             low -= snr_fudge;

  613.             used_abits = init_quantization_noise(c, low);

  614.         }

  615.     }

  616. 

  617.     /* Now do a binary search between low and high to see what fits */

  618.     for (down = snr_fudge >> 1; down; down >>= 1) {

  619.         init_quantization_noise(c, high - down);

  620.         if (c->consumed_bits <= c->frame_bits)

  621.             high -= down;

  622.     }

  623.     init_quantization_noise(c, high);

  624. out:

  625.     c->worst_quantization_noise = high;

  626.     if (high > c->worst_noise_ever)

  627.         c->worst_noise_ever = high;

  628. }

  629. 

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

  631. {

  632.     int k, ch;

  633. 

  634.     for (k = 0; k < 512; k++)

  635.         for (ch = 0; ch < c->channels; ch++)

  636.             c->history[k][ch] = input[k * c->channels + ch];

  637. }

  638. 

  639. static int32_t quantize_value(int32_t value, softfloat quant)

  640. {

  641.     int32_t offset = 1 << (quant.e - 1);

  642. 

  643.     value = mul32(value, quant.m) + offset;

  644.     value = value >> quant.e;

  645.     return value;

  646. }

  647. 

  648. static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)

  649. {

  650.     int32_t peak;

  651.     int our_nscale, try_remove;

  652.     softfloat our_quant;

  653. 

  654.     av_assert0(peak_cb <= 0);

  655.     av_assert0(peak_cb >= -2047);

  656. 

  657.     our_nscale = 127;

  658.     peak = cb_to_level[-peak_cb];

  659. 

  660.     for (try_remove = 64; try_remove > 0; try_remove >>= 1) {

  661.         if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)

  662.             continue;

  663.         our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);

  664.         our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;

  665.         if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))

  666.             continue;

  667.         our_nscale -= try_remove;

  668.     }

  669. 

  670.     if (our_nscale >= 125)

  671.         our_nscale = 124;

  672. 

  673.     quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);

  674.     quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;

  675.     av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));

  676. 

  677.     return our_nscale;

  678. }

  679. 

  680. static void calc_scales(DCAEncContext *c)

  681. {

  682.     int band, ch;

  683. 

  684.     for (band = 0; band < 32; band++)

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

  686.             c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],

  687.                                                        c->abits[band][ch],

  688.                                                        &c->quant[band][ch]);

  689. 

  690.     if (c->lfe_channel)

  691.         c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);

  692. }

  693. 

  694. static void quantize_all(DCAEncContext *c)

  695. {

  696.     int sample, band, ch;

  697. 

  698.     for (sample = 0; sample < SUBBAND_SAMPLES; sample++)

  699.         for (band = 0; band < 32; band++)

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

  701.                 c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);

  702. }

  703. 

  704. static void put_frame_header(DCAEncContext *c)

  705. {

  706.     /* SYNC */

  707.     put_bits(&c->pb, 16, 0x7ffe);

  708.     put_bits(&c->pb, 16, 0x8001);

  709. 

  710.     /* Frame type: normal */

  711.     put_bits(&c->pb, 1, 1);

  712. 

  713.     /* Deficit sample count: none */

  714.     put_bits(&c->pb, 5, 31);

  715. 

  716.     /* CRC is not present */

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

  718. 

  719.     /* Number of PCM sample blocks */

  720.     put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);

  721. 

  722.     /* Primary frame byte size */

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

  724. 

  725.     /* Audio channel arrangement */

  726.     put_bits(&c->pb, 6, c->channel_config);

  727. 

  728.     /* Core audio sampling frequency */

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

  730. 

  731.     /* Transmission bit rate */

  732.     put_bits(&c->pb, 5, c->bitrate_index);

  733. 

  734.     /* Embedded down mix: disabled */

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

  736. 

  737.     /* Embedded dynamic range flag: not present */

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

  739. 

  740.     /* Embedded time stamp flag: not present */

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

  742. 

  743.     /* Auxiliary data flag: not present */

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

  745. 

  746.     /* HDCD source: no */

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

  748. 

  749.     /* Extension audio ID: N/A */

  750.     put_bits(&c->pb, 3, 0);

  751. 

  752.     /* Extended audio data: not present */

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

  754. 

  755.     /* Audio sync word insertion flag: after each sub-frame */

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

  757. 

  758.     /* Low frequency effects flag: not present or 64x subsampling */

  759.     put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);

  760. 

  761.     /* Predictor history switch flag: on */

  762.     put_bits(&c->pb, 1, 1);

  763. 

  764.     /* No CRC */

  765.     /* Multirate interpolator switch: non-perfect reconstruction */

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

  767. 

  768.     /* Encoder software revision: 7 */

  769.     put_bits(&c->pb, 4, 7);

  770. 

  771.     /* Copy history: 0 */

  772.     put_bits(&c->pb, 2, 0);

  773. 

  774.     /* Source PCM resolution: 16 bits, not DTS ES */

  775.     put_bits(&c->pb, 3, 0);

  776. 

  777.     /* Front sum/difference coding: no */

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

  779. 

  780.     /* Surrounds sum/difference coding: no */

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

  782. 

  783.     /* Dialog normalization: 0 dB */

  784.     put_bits(&c->pb, 4, 0);

  785. }

  786. 

  787. static void put_primary_audio_header(DCAEncContext *c)

  788. {

  789.     static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };

  790.     static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

  791. 

  792.     int ch, i;

  793.     /* Number of subframes */

  794.     put_bits(&c->pb, 4, SUBFRAMES - 1);

  795. 

  796.     /* Number of primary audio channels */

  797.     put_bits(&c->pb, 3, c->fullband_channels - 1);

  798. 

  799.     /* Subband activity count */

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

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

  802. 

  803.     /* High frequency VQ start subband */

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

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

  806. 

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

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

  809.         put_bits(&c->pb, 3, 0);

  810. 

  811.     /* Transient mode codebook: A4, A4 (arbitrary) */

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

  813.         put_bits(&c->pb, 2, 0);

  814. 

  815.     /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */

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

  817.         put_bits(&c->pb, 3, 6);

  818. 

  819.     /* Bit allocation quantizer select: linear 5-bit */

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

  821.         put_bits(&c->pb, 3, 6);

  822. 

  823.     /* Quantization index codebook select: dummy data

  824.        to avoid transmission of scale factor adjustment */

  825.     for (i = 1; i < 11; i++)

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

  827.             put_bits(&c->pb, bitlen[i], thr[i]);

  828. 

  829.     /* Scale factor adjustment index: not transmitted */

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

  831. }

  832. 

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

  834. {

  835.     if (c->abits[band][ch] <= 7) {

  836.         int sum, i, j;

  837.         for (i = 0; i < 8; i += 4) {

  838.             sum = 0;

  839.             for (j = 3; j >= 0; j--) {

  840.                 sum *= quant_levels[c->abits[band][ch]];

  841.                 sum += c->quantized[ss * 8 + i + j][band][ch];

  842.                 sum += (quant_levels[c->abits[band][ch]] - 1) / 2;

  843.             }

  844.             put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);

  845.         }

  846.     } else {

  847.         int i;

  848.         for (i = 0; i < 8; i++) {

  849.             int bits = bit_consumption[c->abits[band][ch]] / 16;

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

  851.         }

  852.     }

  853. }

  854. 

  855. static void put_subframe(DCAEncContext *c, int subframe)

  856. {

  857.     int i, band, ss, ch;

  858. 

  859.     /* Subsubframes count */

  860.     put_bits(&c->pb, 2, SUBSUBFRAMES -1);

  861. 

  862.     /* Partial subsubframe sample count: dummy */

  863.     put_bits(&c->pb, 3, 0);

  864. 

  865.     /* Prediction mode: no ADPCM, in each channel and subband */

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

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

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

  869. 

  870.     /* Prediction VQ address: not transmitted */

  871.     /* Bit allocation index */

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

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

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

  875. 

  876.     if (SUBSUBFRAMES > 1) {

  877.         /* Transition mode: none for each channel and subband */

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

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

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

  881.     }

  882. 

  883.     /* Scale factors */

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

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

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

  887. 

  888.     /* Joint subband scale factor codebook select: not transmitted */

  889.     /* Scale factors for joint subband coding: not transmitted */

  890.     /* Stereo down-mix coefficients: not transmitted */

  891.     /* Dynamic range coefficient: not transmitted */

  892.     /* Stde information CRC check word: not transmitted */

  893.     /* VQ encoded high frequency subbands: not transmitted */

  894. 

  895.     /* LFE data: 8 samples and scalefactor */

  896.     if (c->lfe_channel) {

  897.         for (i = 0; i < DCA_LFE_SAMPLES; i++)

  898.             put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);

  899.         put_bits(&c->pb, 8, c->lfe_scale_factor);

  900.     }

  901. 

  902.     /* Audio data (subsubframes) */

  903.     for (ss = 0; ss < SUBSUBFRAMES ; ss++)

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

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

  906.                     put_subframe_samples(c, ss, band, ch);

  907. 

  908.     /* DSYNC */

  909.     put_bits(&c->pb, 16, 0xffff);

  910. }

  911. 

  912. static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

  913.                         const AVFrame *frame, int *got_packet_ptr)

  914. {

  915.     DCAEncContext *c = avctx->priv_data;

  916.     const int32_t *samples;

  917.     int ret, i;

  918. 

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

  920.         return ret;

  921. 

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

  923. 

  924.     subband_transform(c, samples);

  925.     if (c->lfe_channel)

  926.         lfe_downsample(c, samples);

  927. 

  928.     calc_masking(c, samples);

  929.     find_peaks(c);

  930.     assign_bits(c);

  931.     calc_scales(c);

  932.     quantize_all(c);

  933.     shift_history(c, samples);

  934. 

  935.     init_put_bits(&c->pb, avpkt->data, avpkt->size);

  936.     put_frame_header(c);

  937.     put_primary_audio_header(c);

  938.     for (i = 0; i < SUBFRAMES; i++)

  939.         put_subframe(c, i);

  940. 

  941. 

  942.     for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++)

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

  944. 

  945.     flush_put_bits(&c->pb);

  946. 

  947.     avpkt->pts      = frame->pts;

  948.     avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);

  949.     avpkt->size     = c->frame_size + 1;

  950.     *got_packet_ptr = 1;

  951.     return 0;

  952. }

  953. 

  954. static const AVCodecDefault defaults[] = {

  955.     { "b",          "1411200" },

  956.     { NULL },

  957. };

  958. 

  959. AVCodec ff_dca_encoder = {

  960.     .name                  = "dca",

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

  962.     .type                  = AVMEDIA_TYPE_AUDIO,

  963.     .id                    = AV_CODEC_ID_DTS,

  964.     .priv_data_size        = sizeof(DCAEncContext),

  965.     .init                  = encode_init,

  966.     .encode2               = encode_frame,

  967.     .capabilities          = AV_CODEC_CAP_EXPERIMENTAL,

  968.     .sample_fmts           = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,

  969.                                                             AV_SAMPLE_FMT_NONE },

  970.     .supported_samplerates = sample_rates,

  971.     .channel_layouts       = (const uint64_t[]) { AV_CH_LAYOUT_MONO,

  972.                                                   AV_CH_LAYOUT_STEREO,

  973.                                                   AV_CH_LAYOUT_2_2,

  974.                                                   AV_CH_LAYOUT_5POINT0,

  975.                                                   AV_CH_LAYOUT_5POINT1,

  976.                                                   0 },

  977.     .defaults              = defaults,

  978. };