xref: /third_party/ffmpeg/libavcodec/dcaenc.c (revision cabdff1a)
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#define FFT_FLOAT 0
25
26#include "libavutil/avassert.h"
27#include "libavutil/channel_layout.h"
28#include "libavutil/common.h"
29#include "libavutil/ffmath.h"
30#include "libavutil/mem_internal.h"
31#include "libavutil/opt.h"
32#include "avcodec.h"
33#include "codec_internal.h"
34#include "dca.h"
35#include "dcaadpcm.h"
36#include "dcamath.h"
37#include "dca_core.h"
38#include "dcadata.h"
39#include "dcaenc.h"
40#include "encode.h"
41#include "fft.h"
42#include "internal.h"
43#include "mathops.h"
44#include "put_bits.h"
45
46#define MAX_CHANNELS 6
47#define DCA_MAX_FRAME_SIZE 16384
48#define DCA_HEADER_SIZE 13
49#define DCA_LFE_SAMPLES 8
50
51#define DCAENC_SUBBANDS 32
52#define SUBFRAMES 1
53#define SUBSUBFRAMES 2
54#define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
55#define AUBANDS 25
56
57#define COS_T(x) (c->cos_table[(x) & 2047])
58
59typedef struct CompressionOptions {
60    int adpcm_mode;
61} CompressionOptions;
62
63typedef struct DCAEncContext {
64    AVClass *class;
65    PutBitContext pb;
66    DCAADPCMEncContext adpcm_ctx;
67    FFTContext mdct;
68    CompressionOptions options;
69    int frame_size;
70    int frame_bits;
71    int fullband_channels;
72    int channels;
73    int lfe_channel;
74    int samplerate_index;
75    int bitrate_index;
76    int channel_config;
77    const int32_t *band_interpolation;
78    const int32_t *band_spectrum;
79    int lfe_scale_factor;
80    softfloat lfe_quant;
81    int32_t lfe_peak_cb;
82    const int8_t *channel_order_tab;  ///< channel reordering table, lfe and non lfe
83
84    int32_t prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS];
85    int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS * 2];
86    int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
87    int32_t *subband[MAX_CHANNELS][DCAENC_SUBBANDS];
88    int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES];
89    int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS];
90    int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; ///< expected peak of residual signal
91    int32_t downsampled_lfe[DCA_LFE_SAMPLES];
92    int32_t masking_curve_cb[SUBSUBFRAMES][256];
93    int32_t bit_allocation_sel[MAX_CHANNELS];
94    int abits[MAX_CHANNELS][DCAENC_SUBBANDS];
95    int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS];
96    softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS];
97    int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS];
98    int32_t eff_masking_curve_cb[256];
99    int32_t band_masking_cb[32];
100    int32_t worst_quantization_noise;
101    int32_t worst_noise_ever;
102    int consumed_bits;
103    int consumed_adpcm_bits; ///< Number of bits to transmit ADPCM related info
104
105    int32_t cos_table[2048];
106    int32_t band_interpolation_tab[2][512];
107    int32_t band_spectrum_tab[2][8];
108    int32_t auf[9][AUBANDS][256];
109    int32_t cb_to_add[256];
110    int32_t cb_to_level[2048];
111    int32_t lfe_fir_64i[512];
112} DCAEncContext;
113
114/* Transfer function of outer and middle ear, Hz -> dB */
115static double hom(double f)
116{
117    double f1 = f / 1000;
118
119    return -3.64 * pow(f1, -0.8)
120           + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
121           - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
122           - 0.0006 * (f1 * f1) * (f1 * f1);
123}
124
125static double gammafilter(int i, double f)
126{
127    double h = (f - fc[i]) / erb[i];
128
129    h = 1 + h * h;
130    h = 1 / (h * h);
131    return 20 * log10(h);
132}
133
134static int subband_bufer_alloc(DCAEncContext *c)
135{
136    int ch, band;
137    int32_t *bufer = av_calloc(MAX_CHANNELS * DCAENC_SUBBANDS *
138                               (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS),
139                               sizeof(int32_t));
140    if (!bufer)
141        return AVERROR(ENOMEM);
142
143    /* we need a place for DCA_ADPCM_COEFF samples from previous frame
144     * to calc prediction coefficients for each subband */
145    for (ch = 0; ch < MAX_CHANNELS; ch++) {
146        for (band = 0; band < DCAENC_SUBBANDS; band++) {
147            c->subband[ch][band] = bufer +
148                                   ch * DCAENC_SUBBANDS * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) +
149                                   band * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + DCA_ADPCM_COEFFS;
150        }
151    }
152    return 0;
153}
154
155static void subband_bufer_free(DCAEncContext *c)
156{
157    if (c->subband[0][0]) {
158        int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS;
159        av_free(bufer);
160        c->subband[0][0] = NULL;
161    }
162}
163
164static int encode_init(AVCodecContext *avctx)
165{
166    DCAEncContext *c = avctx->priv_data;
167    AVChannelLayout layout = avctx->ch_layout;
168    int i, j, k, min_frame_bits;
169    int ret;
170
171    if ((ret = subband_bufer_alloc(c)) < 0)
172        return ret;
173
174    c->fullband_channels = c->channels = layout.nb_channels;
175    c->lfe_channel = (c->channels == 3 || c->channels == 6);
176    c->band_interpolation = c->band_interpolation_tab[1];
177    c->band_spectrum = c->band_spectrum_tab[1];
178    c->worst_quantization_noise = -2047;
179    c->worst_noise_ever = -2047;
180    c->consumed_adpcm_bits = 0;
181
182    if (ff_dcaadpcm_init(&c->adpcm_ctx))
183        return AVERROR(ENOMEM);
184
185    if (layout.order == AV_CHANNEL_ORDER_UNSPEC) {
186        av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
187                                      "encoder will guess the layout, but it "
188                                      "might be incorrect.\n");
189        av_channel_layout_default(&layout, layout.nb_channels);
190    }
191
192    if (!av_channel_layout_compare(&layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_MONO))
193        c->channel_config = 0;
194    else if (!av_channel_layout_compare(&layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO))
195        c->channel_config = 2;
196    else if (!av_channel_layout_compare(&layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_2_2))
197        c->channel_config = 8;
198    else if (!av_channel_layout_compare(&layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0))
199        c->channel_config = 9;
200    else if (!av_channel_layout_compare(&layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1))
201        c->channel_config = 9;
202    else {
203        av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");
204        return AVERROR_PATCHWELCOME;
205    }
206
207    if (c->lfe_channel) {
208        c->fullband_channels--;
209        c->channel_order_tab = channel_reorder_lfe[c->channel_config];
210    } else {
211        c->channel_order_tab = channel_reorder_nolfe[c->channel_config];
212    }
213
214    for (i = 0; i < MAX_CHANNELS; i++) {
215        for (j = 0; j < DCA_CODE_BOOKS; j++) {
216            c->quant_index_sel[i][j] = ff_dca_quant_index_group_size[j];
217        }
218        /* 6 - no Huffman */
219        c->bit_allocation_sel[i] = 6;
220
221        for (j = 0; j < DCAENC_SUBBANDS; j++) {
222            /* -1 - no ADPCM */
223            c->prediction_mode[i][j] = -1;
224            memset(c->adpcm_history[i][j], 0, sizeof(int32_t)*DCA_ADPCM_COEFFS);
225        }
226    }
227
228    for (i = 0; i < 9; i++) {
229        if (sample_rates[i] == avctx->sample_rate)
230            break;
231    }
232    if (i == 9)
233        return AVERROR(EINVAL);
234    c->samplerate_index = i;
235
236    if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {
237        av_log(avctx, AV_LOG_ERROR, "Bit rate %"PRId64" not supported.", avctx->bit_rate);
238        return AVERROR(EINVAL);
239    }
240    for (i = 0; ff_dca_bit_rates[i] < avctx->bit_rate; i++)
241        ;
242    c->bitrate_index = i;
243    c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);
244    min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;
245    if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))
246        return AVERROR(EINVAL);
247
248    c->frame_size = (c->frame_bits + 7) / 8;
249
250    avctx->frame_size = 32 * SUBBAND_SAMPLES;
251
252    if ((ret = ff_mdct_init(&c->mdct, 9, 0, 1.0)) < 0)
253        return ret;
254
255    /* Init all tables */
256    c->cos_table[0] = 0x7fffffff;
257    c->cos_table[512] = 0;
258    c->cos_table[1024] = -c->cos_table[0];
259    for (i = 1; i < 512; i++) {
260        c->cos_table[i]   = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
261        c->cos_table[1024-i] = -c->cos_table[i];
262        c->cos_table[1024+i] = -c->cos_table[i];
263        c->cos_table[2048-i] = +c->cos_table[i];
264    }
265
266    for (i = 0; i < 2048; i++)
267        c->cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i));
268
269    for (k = 0; k < 32; k++) {
270        for (j = 0; j < 8; j++) {
271            c->lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
272            c->lfe_fir_64i[64 * (7-j) + (63 - k)] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
273        }
274    }
275
276    for (i = 0; i < 512; i++) {
277        c->band_interpolation_tab[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
278        c->band_interpolation_tab[1][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_nonperfect[i]);
279    }
280
281    for (i = 0; i < 9; i++) {
282        for (j = 0; j < AUBANDS; j++) {
283            for (k = 0; k < 256; k++) {
284                double freq = sample_rates[i] * (k + 0.5) / 512;
285
286                c->auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));
287            }
288        }
289    }
290
291    for (i = 0; i < 256; i++) {
292        double add = 1 + ff_exp10(-0.01 * i);
293        c->cb_to_add[i] = (int32_t)(100 * log10(add));
294    }
295    for (j = 0; j < 8; j++) {
296        double accum = 0;
297        for (i = 0; i < 512; i++) {
298            double reconst = ff_dca_fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);
299            accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
300        }
301        c->band_spectrum_tab[0][j] = (int32_t)(200 * log10(accum));
302    }
303    for (j = 0; j < 8; j++) {
304        double accum = 0;
305        for (i = 0; i < 512; i++) {
306            double reconst = ff_dca_fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);
307            accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
308        }
309        c->band_spectrum_tab[1][j] = (int32_t)(200 * log10(accum));
310    }
311
312    return 0;
313}
314
315static av_cold int encode_close(AVCodecContext *avctx)
316{
317    DCAEncContext *c = avctx->priv_data;
318    ff_mdct_end(&c->mdct);
319    subband_bufer_free(c);
320    ff_dcaadpcm_free(&c->adpcm_ctx);
321
322    return 0;
323}
324
325static void subband_transform(DCAEncContext *c, const int32_t *input)
326{
327    int ch, subs, i, k, j;
328
329    for (ch = 0; ch < c->fullband_channels; ch++) {
330        /* History is copied because it is also needed for PSY */
331        int32_t hist[512];
332        int hist_start = 0;
333        const int chi = c->channel_order_tab[ch];
334
335        memcpy(hist, &c->history[ch][0], 512 * sizeof(int32_t));
336
337        for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
338            int32_t accum[64];
339            int32_t resp;
340            int band;
341
342            /* Calculate the convolutions at once */
343            memset(accum, 0, 64 * sizeof(int32_t));
344
345            for (k = 0, i = hist_start, j = 0;
346                    i < 512; k = (k + 1) & 63, i++, j++)
347                accum[k] += mul32(hist[i], c->band_interpolation[j]);
348            for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)
349                accum[k] += mul32(hist[i], c->band_interpolation[j]);
350
351            for (k = 16; k < 32; k++)
352                accum[k] = accum[k] - accum[31 - k];
353            for (k = 32; k < 48; k++)
354                accum[k] = accum[k] + accum[95 - k];
355
356            for (band = 0; band < 32; band++) {
357                resp = 0;
358                for (i = 16; i < 48; i++) {
359                    int s = (2 * band + 1) * (2 * (i + 16) + 1);
360                    resp += mul32(accum[i], COS_T(s << 3)) >> 3;
361                }
362
363                c->subband[ch][band][subs] = ((band + 1) & 2) ? -resp : resp;
364            }
365
366            /* Copy in 32 new samples from input */
367            for (i = 0; i < 32; i++)
368                hist[i + hist_start] = input[(subs * 32 + i) * c->channels + chi];
369
370            hist_start = (hist_start + 32) & 511;
371        }
372    }
373}
374
375static void lfe_downsample(DCAEncContext *c, const int32_t *input)
376{
377    /* FIXME: make 128x LFE downsampling possible */
378    const int lfech = lfe_index[c->channel_config];
379    int i, j, lfes;
380    int32_t hist[512];
381    int32_t accum;
382    int hist_start = 0;
383
384    memcpy(hist, &c->history[c->channels - 1][0], 512 * sizeof(int32_t));
385
386    for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
387        /* Calculate the convolution */
388        accum = 0;
389
390        for (i = hist_start, j = 0; i < 512; i++, j++)
391            accum += mul32(hist[i], c->lfe_fir_64i[j]);
392        for (i = 0; i < hist_start; i++, j++)
393            accum += mul32(hist[i], c->lfe_fir_64i[j]);
394
395        c->downsampled_lfe[lfes] = accum;
396
397        /* Copy in 64 new samples from input */
398        for (i = 0; i < 64; i++)
399            hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + lfech];
400
401        hist_start = (hist_start + 64) & 511;
402    }
403}
404
405static int32_t get_cb(DCAEncContext *c, int32_t in)
406{
407    int i, res = 0;
408    in = FFABS(in);
409
410    for (i = 1024; i > 0; i >>= 1) {
411        if (c->cb_to_level[i + res] >= in)
412            res += i;
413    }
414    return -res;
415}
416
417static int32_t add_cb(DCAEncContext *c, int32_t a, int32_t b)
418{
419    if (a < b)
420        FFSWAP(int32_t, a, b);
421
422    if (a - b >= 256)
423        return a;
424    return a + c->cb_to_add[a - b];
425}
426
427static void calc_power(DCAEncContext *c,
428                       const int32_t in[2 * 256], int32_t power[256])
429{
430    int i;
431    LOCAL_ALIGNED_32(int32_t, data,  [512]);
432    LOCAL_ALIGNED_32(int32_t, coeff, [256]);
433
434    for (i = 0; i < 512; i++)
435        data[i] = norm__(mul32(in[i], 0x3fffffff - (COS_T(4 * i + 2) >> 1)), 4);
436
437    c->mdct.mdct_calc(&c->mdct, coeff, data);
438    for (i = 0; i < 256; i++) {
439        const int32_t cb = get_cb(c, coeff[i]);
440        power[i] = add_cb(c, cb, cb);
441    }
442}
443
444static void adjust_jnd(DCAEncContext *c,
445                       const int32_t in[512], int32_t out_cb[256])
446{
447    int32_t power[256];
448    int32_t out_cb_unnorm[256];
449    int32_t denom;
450    const int32_t ca_cb = -1114;
451    const int32_t cs_cb = 928;
452    const int samplerate_index = c->samplerate_index;
453    int i, j;
454
455    calc_power(c, in, power);
456
457    for (j = 0; j < 256; j++)
458        out_cb_unnorm[j] = -2047; /* and can only grow */
459
460    for (i = 0; i < AUBANDS; i++) {
461        denom = ca_cb; /* and can only grow */
462        for (j = 0; j < 256; j++)
463            denom = add_cb(c, denom, power[j] + c->auf[samplerate_index][i][j]);
464        for (j = 0; j < 256; j++)
465            out_cb_unnorm[j] = add_cb(c, out_cb_unnorm[j],
466                                      -denom + c->auf[samplerate_index][i][j]);
467    }
468
469    for (j = 0; j < 256; j++)
470        out_cb[j] = add_cb(c, out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);
471}
472
473typedef void (*walk_band_t)(DCAEncContext *c, int band1, int band2, int f,
474                            int32_t spectrum1, int32_t spectrum2, int channel,
475                            int32_t * arg);
476
477static void walk_band_low(DCAEncContext *c, int band, int channel,
478                          walk_band_t walk, int32_t *arg)
479{
480    int f;
481
482    if (band == 0) {
483        for (f = 0; f < 4; f++)
484            walk(c, 0, 0, f, 0, -2047, channel, arg);
485    } else {
486        for (f = 0; f < 8; f++)
487            walk(c, band, band - 1, 8 * band - 4 + f,
488                    c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);
489    }
490}
491
492static void walk_band_high(DCAEncContext *c, int band, int channel,
493                           walk_band_t walk, int32_t *arg)
494{
495    int f;
496
497    if (band == 31) {
498        for (f = 0; f < 4; f++)
499            walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);
500    } else {
501        for (f = 0; f < 8; f++)
502            walk(c, band, band + 1, 8 * band + 4 + f,
503                    c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);
504    }
505}
506
507static void update_band_masking(DCAEncContext *c, int band1, int band2,
508                                int f, int32_t spectrum1, int32_t spectrum2,
509                                int channel, int32_t * arg)
510{
511    int32_t value = c->eff_masking_curve_cb[f] - spectrum1;
512
513    if (value < c->band_masking_cb[band1])
514        c->band_masking_cb[band1] = value;
515}
516
517static void calc_masking(DCAEncContext *c, const int32_t *input)
518{
519    int i, k, band, ch, ssf;
520    int32_t data[512];
521
522    for (i = 0; i < 256; i++)
523        for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
524            c->masking_curve_cb[ssf][i] = -2047;
525
526    for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
527        for (ch = 0; ch < c->fullband_channels; ch++) {
528            const int chi = c->channel_order_tab[ch];
529
530            for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
531                data[i] = c->history[ch][k];
532            for (k -= 512; i < 512; i++, k++)
533                data[i] = input[k * c->channels + chi];
534            adjust_jnd(c, data, c->masking_curve_cb[ssf]);
535        }
536    for (i = 0; i < 256; i++) {
537        int32_t m = 2048;
538
539        for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
540            if (c->masking_curve_cb[ssf][i] < m)
541                m = c->masking_curve_cb[ssf][i];
542        c->eff_masking_curve_cb[i] = m;
543    }
544
545    for (band = 0; band < 32; band++) {
546        c->band_masking_cb[band] = 2048;
547        walk_band_low(c, band, 0, update_band_masking, NULL);
548        walk_band_high(c, band, 0, update_band_masking, NULL);
549    }
550}
551
552static inline int32_t find_peak(DCAEncContext *c, const int32_t *in, int len)
553{
554    int sample;
555    int32_t m = 0;
556    for (sample = 0; sample < len; sample++) {
557        int32_t s = abs(in[sample]);
558        if (m < s)
559            m = s;
560    }
561    return get_cb(c, m);
562}
563
564static void find_peaks(DCAEncContext *c)
565{
566    int band, ch;
567
568    for (ch = 0; ch < c->fullband_channels; ch++) {
569        for (band = 0; band < 32; band++)
570            c->peak_cb[ch][band] = find_peak(c, c->subband[ch][band],
571                                             SUBBAND_SAMPLES);
572    }
573
574    if (c->lfe_channel)
575        c->lfe_peak_cb = find_peak(c, c->downsampled_lfe, DCA_LFE_SAMPLES);
576}
577
578static void adpcm_analysis(DCAEncContext *c)
579{
580    int ch, band;
581    int pred_vq_id;
582    int32_t *samples;
583    int32_t estimated_diff[SUBBAND_SAMPLES];
584
585    c->consumed_adpcm_bits = 0;
586    for (ch = 0; ch < c->fullband_channels; ch++) {
587        for (band = 0; band < 32; band++) {
588            samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
589            pred_vq_id = ff_dcaadpcm_subband_analysis(&c->adpcm_ctx, samples,
590                                                      SUBBAND_SAMPLES, estimated_diff);
591            if (pred_vq_id >= 0) {
592                c->prediction_mode[ch][band] = pred_vq_id;
593                c->consumed_adpcm_bits += 12; //12 bits to transmit prediction vq index
594                c->diff_peak_cb[ch][band] = find_peak(c, estimated_diff, 16);
595            } else {
596                c->prediction_mode[ch][band] = -1;
597            }
598        }
599    }
600}
601
602static const int snr_fudge = 128;
603#define USED_1ABITS 1
604#define USED_26ABITS 4
605
606static inline int32_t get_step_size(DCAEncContext *c, int ch, int band)
607{
608    int32_t step_size;
609
610    if (c->bitrate_index == 3)
611        step_size = ff_dca_lossless_quant[c->abits[ch][band]];
612    else
613        step_size = ff_dca_lossy_quant[c->abits[ch][band]];
614
615    return step_size;
616}
617
618static int calc_one_scale(DCAEncContext *c, int32_t peak_cb, int abits,
619                          softfloat *quant)
620{
621    int32_t peak;
622    int our_nscale, try_remove;
623    softfloat our_quant;
624
625    av_assert0(peak_cb <= 0);
626    av_assert0(peak_cb >= -2047);
627
628    our_nscale = 127;
629    peak = c->cb_to_level[-peak_cb];
630
631    for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
632        if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
633            continue;
634        our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
635        our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
636        if ((ff_dca_quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
637            continue;
638        our_nscale -= try_remove;
639    }
640
641    if (our_nscale >= 125)
642        our_nscale = 124;
643
644    quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
645    quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
646    av_assert0((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
647
648    return our_nscale;
649}
650
651static inline void quantize_adpcm_subband(DCAEncContext *c, int ch, int band)
652{
653    int32_t step_size;
654    int32_t diff_peak_cb = c->diff_peak_cb[ch][band];
655    c->scale_factor[ch][band] = calc_one_scale(c, diff_peak_cb,
656                                               c->abits[ch][band],
657                                               &c->quant[ch][band]);
658
659    step_size = get_step_size(c, ch, band);
660    ff_dcaadpcm_do_real(c->prediction_mode[ch][band],
661                        c->quant[ch][band],
662                        ff_dca_scale_factor_quant7[c->scale_factor[ch][band]],
663                        step_size, c->adpcm_history[ch][band], c->subband[ch][band],
664                        c->adpcm_history[ch][band] + 4, c->quantized[ch][band],
665                        SUBBAND_SAMPLES, c->cb_to_level[-diff_peak_cb]);
666}
667
668static void quantize_adpcm(DCAEncContext *c)
669{
670    int band, ch;
671
672    for (ch = 0; ch < c->fullband_channels; ch++)
673        for (band = 0; band < 32; band++)
674            if (c->prediction_mode[ch][band] >= 0)
675                quantize_adpcm_subband(c, ch, band);
676}
677
678static void quantize_pcm(DCAEncContext *c)
679{
680    int sample, band, ch;
681
682    for (ch = 0; ch < c->fullband_channels; ch++) {
683        for (band = 0; band < 32; band++) {
684            if (c->prediction_mode[ch][band] == -1) {
685                for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
686                    int32_t val = quantize_value(c->subband[ch][band][sample],
687                                                 c->quant[ch][band]);
688                    c->quantized[ch][band][sample] = val;
689                }
690            }
691        }
692    }
693}
694
695static void accumulate_huff_bit_consumption(int abits, int32_t *quantized,
696                                            uint32_t *result)
697{
698    uint8_t sel, id = abits - 1;
699    for (sel = 0; sel < ff_dca_quant_index_group_size[id]; sel++)
700        result[sel] += ff_dca_vlc_calc_quant_bits(quantized, SUBBAND_SAMPLES,
701                                                  sel, id);
702}
703
704static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7],
705                              uint32_t clc_bits[DCA_CODE_BOOKS],
706                              int32_t res[DCA_CODE_BOOKS])
707{
708    uint8_t i, sel;
709    uint32_t best_sel_bits[DCA_CODE_BOOKS];
710    int32_t best_sel_id[DCA_CODE_BOOKS];
711    uint32_t t, bits = 0;
712
713    for (i = 0; i < DCA_CODE_BOOKS; i++) {
714
715        av_assert0(!((!!vlc_bits[i][0]) ^ (!!clc_bits[i])));
716        if (vlc_bits[i][0] == 0) {
717            /* do not transmit adjustment index for empty codebooks */
718            res[i] = ff_dca_quant_index_group_size[i];
719            /* and skip it */
720            continue;
721        }
722
723        best_sel_bits[i] = vlc_bits[i][0];
724        best_sel_id[i] = 0;
725        for (sel = 0; sel < ff_dca_quant_index_group_size[i]; sel++) {
726            if (best_sel_bits[i] > vlc_bits[i][sel] && vlc_bits[i][sel]) {
727                best_sel_bits[i] = vlc_bits[i][sel];
728                best_sel_id[i] = sel;
729            }
730        }
731
732        /* 2 bits to transmit scale factor adjustment index */
733        t = best_sel_bits[i] + 2;
734        if (t < clc_bits[i]) {
735            res[i] = best_sel_id[i];
736            bits += t;
737        } else {
738            res[i] = ff_dca_quant_index_group_size[i];
739            bits += clc_bits[i];
740        }
741    }
742    return bits;
743}
744
745static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands,
746                                    int32_t *res)
747{
748    uint8_t i;
749    uint32_t t;
750    int32_t best_sel = 6;
751    int32_t best_bits = bands * 5;
752
753    /* Check do we have subband which cannot be encoded by Huffman tables */
754    for (i = 0; i < bands; i++) {
755        if (abits[i] > 12 || abits[i] == 0) {
756            *res = best_sel;
757            return best_bits;
758        }
759    }
760
761    for (i = 0; i < DCA_BITALLOC_12_COUNT; i++) {
762        t = ff_dca_vlc_calc_alloc_bits(abits, bands, i);
763        if (t < best_bits) {
764            best_bits = t;
765            best_sel = i;
766        }
767    }
768
769    *res = best_sel;
770    return best_bits;
771}
772
773static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero)
774{
775    int ch, band, ret = USED_26ABITS | USED_1ABITS;
776    uint32_t huff_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS][7];
777    uint32_t clc_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS];
778    uint32_t bits_counter = 0;
779
780    c->consumed_bits = 132 + 333 * c->fullband_channels;
781    c->consumed_bits += c->consumed_adpcm_bits;
782    if (c->lfe_channel)
783        c->consumed_bits += 72;
784
785    /* attempt to guess the bit distribution based on the prevoius frame */
786    for (ch = 0; ch < c->fullband_channels; ch++) {
787        for (band = 0; band < 32; band++) {
788            int snr_cb = c->peak_cb[ch][band] - c->band_masking_cb[band] - noise;
789
790            if (snr_cb >= 1312) {
791                c->abits[ch][band] = 26;
792                ret &= ~USED_1ABITS;
793            } else if (snr_cb >= 222) {
794                c->abits[ch][band] = 8 + mul32(snr_cb - 222, 69000000);
795                ret &= ~(USED_26ABITS | USED_1ABITS);
796            } else if (snr_cb >= 0) {
797                c->abits[ch][band] = 2 + mul32(snr_cb, 106000000);
798                ret &= ~(USED_26ABITS | USED_1ABITS);
799            } else if (forbid_zero || snr_cb >= -140) {
800                c->abits[ch][band] = 1;
801                ret &= ~USED_26ABITS;
802            } else {
803                c->abits[ch][band] = 0;
804                ret &= ~(USED_26ABITS | USED_1ABITS);
805            }
806        }
807        c->consumed_bits += set_best_abits_code(c->abits[ch], 32,
808                                                &c->bit_allocation_sel[ch]);
809    }
810
811    /* Recalc scale_factor each time to get bits consumption in case of Huffman coding.
812       It is suboptimal solution */
813    /* TODO: May be cache scaled values */
814    for (ch = 0; ch < c->fullband_channels; ch++) {
815        for (band = 0; band < 32; band++) {
816            if (c->prediction_mode[ch][band] == -1) {
817                c->scale_factor[ch][band] = calc_one_scale(c, c->peak_cb[ch][band],
818                                                           c->abits[ch][band],
819                                                           &c->quant[ch][band]);
820            }
821        }
822    }
823    quantize_adpcm(c);
824    quantize_pcm(c);
825
826    memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t));
827    memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t));
828    for (ch = 0; ch < c->fullband_channels; ch++) {
829        for (band = 0; band < 32; band++) {
830            if (c->abits[ch][band] && c->abits[ch][band] <= DCA_CODE_BOOKS) {
831                accumulate_huff_bit_consumption(c->abits[ch][band],
832                                                c->quantized[ch][band],
833                                                huff_bit_count_accum[ch][c->abits[ch][band] - 1]);
834                clc_bit_count_accum[ch][c->abits[ch][band] - 1] += bit_consumption[c->abits[ch][band]];
835            } else {
836                bits_counter += bit_consumption[c->abits[ch][band]];
837            }
838        }
839    }
840
841    for (ch = 0; ch < c->fullband_channels; ch++) {
842        bits_counter += set_best_code(huff_bit_count_accum[ch],
843                                      clc_bit_count_accum[ch],
844                                      c->quant_index_sel[ch]);
845    }
846
847    c->consumed_bits += bits_counter;
848
849    return ret;
850}
851
852static void assign_bits(DCAEncContext *c)
853{
854    /* Find the bounds where the binary search should work */
855    int low, high, down;
856    int used_abits = 0;
857    int forbid_zero = 1;
858restart:
859    init_quantization_noise(c, c->worst_quantization_noise, forbid_zero);
860    low = high = c->worst_quantization_noise;
861    if (c->consumed_bits > c->frame_bits) {
862        while (c->consumed_bits > c->frame_bits) {
863            if (used_abits == USED_1ABITS && forbid_zero) {
864                forbid_zero = 0;
865                goto restart;
866            }
867            low = high;
868            high += snr_fudge;
869            used_abits = init_quantization_noise(c, high, forbid_zero);
870        }
871    } else {
872        while (c->consumed_bits <= c->frame_bits) {
873            high = low;
874            if (used_abits == USED_26ABITS)
875                goto out; /* The requested bitrate is too high, pad with zeros */
876            low -= snr_fudge;
877            used_abits = init_quantization_noise(c, low, forbid_zero);
878        }
879    }
880
881    /* Now do a binary search between low and high to see what fits */
882    for (down = snr_fudge >> 1; down; down >>= 1) {
883        init_quantization_noise(c, high - down, forbid_zero);
884        if (c->consumed_bits <= c->frame_bits)
885            high -= down;
886    }
887    init_quantization_noise(c, high, forbid_zero);
888out:
889    c->worst_quantization_noise = high;
890    if (high > c->worst_noise_ever)
891        c->worst_noise_ever = high;
892}
893
894static void shift_history(DCAEncContext *c, const int32_t *input)
895{
896    int k, ch;
897
898    for (k = 0; k < 512; k++)
899        for (ch = 0; ch < c->channels; ch++) {
900            const int chi = c->channel_order_tab[ch];
901
902            c->history[ch][k] = input[k * c->channels + chi];
903        }
904}
905
906static void fill_in_adpcm_bufer(DCAEncContext *c)
907{
908     int ch, band;
909     int32_t step_size;
910     /* We fill in ADPCM work buffer for subbands which hasn't been ADPCM coded
911      * in current frame - we need this data if subband of next frame is
912      * ADPCM
913      */
914     for (ch = 0; ch < c->channels; ch++) {
915        for (band = 0; band < 32; band++) {
916            int32_t *samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
917            if (c->prediction_mode[ch][band] == -1) {
918                step_size = get_step_size(c, ch, band);
919
920                ff_dca_core_dequantize(c->adpcm_history[ch][band],
921                                       c->quantized[ch][band]+12, step_size,
922                                       ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], 0, 4);
923            } else {
924                AV_COPY128U(c->adpcm_history[ch][band], c->adpcm_history[ch][band]+4);
925            }
926            /* Copy dequantized values for LPC analysis.
927             * It reduces artifacts in case of extreme quantization,
928             * example: in current frame abits is 1 and has no prediction flag,
929             * but end of this frame is sine like signal. In this case, if LPC analysis uses
930             * original values, likely LPC analysis returns good prediction gain, and sets prediction flag.
931             * But there are no proper value in decoder history, so likely result will be no good.
932             * Bitstream has "Predictor history flag switch", but this flag disables history for all subbands
933             */
934            samples[0] = c->adpcm_history[ch][band][0] * (1 << 7);
935            samples[1] = c->adpcm_history[ch][band][1] * (1 << 7);
936            samples[2] = c->adpcm_history[ch][band][2] * (1 << 7);
937            samples[3] = c->adpcm_history[ch][band][3] * (1 << 7);
938        }
939     }
940}
941
942static void calc_lfe_scales(DCAEncContext *c)
943{
944    if (c->lfe_channel)
945        c->lfe_scale_factor = calc_one_scale(c, c->lfe_peak_cb, 11, &c->lfe_quant);
946}
947
948static void put_frame_header(DCAEncContext *c)
949{
950    /* SYNC */
951    put_bits(&c->pb, 16, 0x7ffe);
952    put_bits(&c->pb, 16, 0x8001);
953
954    /* Frame type: normal */
955    put_bits(&c->pb, 1, 1);
956
957    /* Deficit sample count: none */
958    put_bits(&c->pb, 5, 31);
959
960    /* CRC is not present */
961    put_bits(&c->pb, 1, 0);
962
963    /* Number of PCM sample blocks */
964    put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
965
966    /* Primary frame byte size */
967    put_bits(&c->pb, 14, c->frame_size - 1);
968
969    /* Audio channel arrangement */
970    put_bits(&c->pb, 6, c->channel_config);
971
972    /* Core audio sampling frequency */
973    put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);
974
975    /* Transmission bit rate */
976    put_bits(&c->pb, 5, c->bitrate_index);
977
978    /* Embedded down mix: disabled */
979    put_bits(&c->pb, 1, 0);
980
981    /* Embedded dynamic range flag: not present */
982    put_bits(&c->pb, 1, 0);
983
984    /* Embedded time stamp flag: not present */
985    put_bits(&c->pb, 1, 0);
986
987    /* Auxiliary data flag: not present */
988    put_bits(&c->pb, 1, 0);
989
990    /* HDCD source: no */
991    put_bits(&c->pb, 1, 0);
992
993    /* Extension audio ID: N/A */
994    put_bits(&c->pb, 3, 0);
995
996    /* Extended audio data: not present */
997    put_bits(&c->pb, 1, 0);
998
999    /* Audio sync word insertion flag: after each sub-frame */
1000    put_bits(&c->pb, 1, 0);
1001
1002    /* Low frequency effects flag: not present or 64x subsampling */
1003    put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);
1004
1005    /* Predictor history switch flag: on */
1006    put_bits(&c->pb, 1, 1);
1007
1008    /* No CRC */
1009    /* Multirate interpolator switch: non-perfect reconstruction */
1010    put_bits(&c->pb, 1, 0);
1011
1012    /* Encoder software revision: 7 */
1013    put_bits(&c->pb, 4, 7);
1014
1015    /* Copy history: 0 */
1016    put_bits(&c->pb, 2, 0);
1017
1018    /* Source PCM resolution: 16 bits, not DTS ES */
1019    put_bits(&c->pb, 3, 0);
1020
1021    /* Front sum/difference coding: no */
1022    put_bits(&c->pb, 1, 0);
1023
1024    /* Surrounds sum/difference coding: no */
1025    put_bits(&c->pb, 1, 0);
1026
1027    /* Dialog normalization: 0 dB */
1028    put_bits(&c->pb, 4, 0);
1029}
1030
1031static void put_primary_audio_header(DCAEncContext *c)
1032{
1033    int ch, i;
1034    /* Number of subframes */
1035    put_bits(&c->pb, 4, SUBFRAMES - 1);
1036
1037    /* Number of primary audio channels */
1038    put_bits(&c->pb, 3, c->fullband_channels - 1);
1039
1040    /* Subband activity count */
1041    for (ch = 0; ch < c->fullband_channels; ch++)
1042        put_bits(&c->pb, 5, DCAENC_SUBBANDS - 2);
1043
1044    /* High frequency VQ start subband */
1045    for (ch = 0; ch < c->fullband_channels; ch++)
1046        put_bits(&c->pb, 5, DCAENC_SUBBANDS - 1);
1047
1048    /* Joint intensity coding index: 0, 0 */
1049    for (ch = 0; ch < c->fullband_channels; ch++)
1050        put_bits(&c->pb, 3, 0);
1051
1052    /* Transient mode codebook: A4, A4 (arbitrary) */
1053    for (ch = 0; ch < c->fullband_channels; ch++)
1054        put_bits(&c->pb, 2, 0);
1055
1056    /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
1057    for (ch = 0; ch < c->fullband_channels; ch++)
1058        put_bits(&c->pb, 3, 6);
1059
1060    /* Bit allocation quantizer select: linear 5-bit */
1061    for (ch = 0; ch < c->fullband_channels; ch++)
1062        put_bits(&c->pb, 3, c->bit_allocation_sel[ch]);
1063
1064    /* Quantization index codebook select */
1065    for (i = 0; i < DCA_CODE_BOOKS; i++)
1066        for (ch = 0; ch < c->fullband_channels; ch++)
1067            put_bits(&c->pb, ff_dca_quant_index_sel_nbits[i], c->quant_index_sel[ch][i]);
1068
1069    /* Scale factor adjustment index: transmitted in case of Huffman coding */
1070    for (i = 0; i < DCA_CODE_BOOKS; i++)
1071        for (ch = 0; ch < c->fullband_channels; ch++)
1072            if (c->quant_index_sel[ch][i] < ff_dca_quant_index_group_size[i])
1073                put_bits(&c->pb, 2, 0);
1074
1075    /* Audio header CRC check word: not transmitted */
1076}
1077
1078static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
1079{
1080    int i, j, sum, bits, sel;
1081    if (c->abits[ch][band] <= DCA_CODE_BOOKS) {
1082        av_assert0(c->abits[ch][band] > 0);
1083        sel = c->quant_index_sel[ch][c->abits[ch][band] - 1];
1084        // Huffman codes
1085        if (sel < ff_dca_quant_index_group_size[c->abits[ch][band] - 1]) {
1086            ff_dca_vlc_enc_quant(&c->pb, &c->quantized[ch][band][ss * 8], 8,
1087                                 sel, c->abits[ch][band] - 1);
1088            return;
1089        }
1090
1091        // Block codes
1092        if (c->abits[ch][band] <= 7) {
1093            for (i = 0; i < 8; i += 4) {
1094                sum = 0;
1095                for (j = 3; j >= 0; j--) {
1096                    sum *= ff_dca_quant_levels[c->abits[ch][band]];
1097                    sum += c->quantized[ch][band][ss * 8 + i + j];
1098                    sum += (ff_dca_quant_levels[c->abits[ch][band]] - 1) / 2;
1099                }
1100                put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum);
1101            }
1102            return;
1103        }
1104    }
1105
1106    for (i = 0; i < 8; i++) {
1107        bits = bit_consumption[c->abits[ch][band]] / 16;
1108        put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]);
1109    }
1110}
1111
1112static void put_subframe(DCAEncContext *c, int subframe)
1113{
1114    int i, band, ss, ch;
1115
1116    /* Subsubframes count */
1117    put_bits(&c->pb, 2, SUBSUBFRAMES -1);
1118
1119    /* Partial subsubframe sample count: dummy */
1120    put_bits(&c->pb, 3, 0);
1121
1122    /* Prediction mode: no ADPCM, in each channel and subband */
1123    for (ch = 0; ch < c->fullband_channels; ch++)
1124        for (band = 0; band < DCAENC_SUBBANDS; band++)
1125            put_bits(&c->pb, 1, !(c->prediction_mode[ch][band] == -1));
1126
1127    /* Prediction VQ address */
1128    for (ch = 0; ch < c->fullband_channels; ch++)
1129        for (band = 0; band < DCAENC_SUBBANDS; band++)
1130            if (c->prediction_mode[ch][band] >= 0)
1131                put_bits(&c->pb, 12, c->prediction_mode[ch][band]);
1132
1133    /* Bit allocation index */
1134    for (ch = 0; ch < c->fullband_channels; ch++) {
1135        if (c->bit_allocation_sel[ch] == 6) {
1136            for (band = 0; band < DCAENC_SUBBANDS; band++) {
1137                put_bits(&c->pb, 5, c->abits[ch][band]);
1138            }
1139        } else {
1140            ff_dca_vlc_enc_alloc(&c->pb, c->abits[ch], DCAENC_SUBBANDS,
1141                                 c->bit_allocation_sel[ch]);
1142        }
1143    }
1144
1145    if (SUBSUBFRAMES > 1) {
1146        /* Transition mode: none for each channel and subband */
1147        for (ch = 0; ch < c->fullband_channels; ch++)
1148            for (band = 0; band < DCAENC_SUBBANDS; band++)
1149                if (c->abits[ch][band])
1150                    put_bits(&c->pb, 1, 0); /* codebook A4 */
1151    }
1152
1153    /* Scale factors */
1154    for (ch = 0; ch < c->fullband_channels; ch++)
1155        for (band = 0; band < DCAENC_SUBBANDS; band++)
1156            if (c->abits[ch][band])
1157                put_bits(&c->pb, 7, c->scale_factor[ch][band]);
1158
1159    /* Joint subband scale factor codebook select: not transmitted */
1160    /* Scale factors for joint subband coding: not transmitted */
1161    /* Stereo down-mix coefficients: not transmitted */
1162    /* Dynamic range coefficient: not transmitted */
1163    /* Stde information CRC check word: not transmitted */
1164    /* VQ encoded high frequency subbands: not transmitted */
1165
1166    /* LFE data: 8 samples and scalefactor */
1167    if (c->lfe_channel) {
1168        for (i = 0; i < DCA_LFE_SAMPLES; i++)
1169            put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
1170        put_bits(&c->pb, 8, c->lfe_scale_factor);
1171    }
1172
1173    /* Audio data (subsubframes) */
1174    for (ss = 0; ss < SUBSUBFRAMES ; ss++)
1175        for (ch = 0; ch < c->fullband_channels; ch++)
1176            for (band = 0; band < DCAENC_SUBBANDS; band++)
1177                if (c->abits[ch][band])
1178                    put_subframe_samples(c, ss, band, ch);
1179
1180    /* DSYNC */
1181    put_bits(&c->pb, 16, 0xffff);
1182}
1183
1184static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
1185                        const AVFrame *frame, int *got_packet_ptr)
1186{
1187    DCAEncContext *c = avctx->priv_data;
1188    const int32_t *samples;
1189    int ret, i;
1190
1191    if ((ret = ff_get_encode_buffer(avctx, avpkt, c->frame_size, 0)) < 0)
1192        return ret;
1193
1194    samples = (const int32_t *)frame->data[0];
1195
1196    subband_transform(c, samples);
1197    if (c->lfe_channel)
1198        lfe_downsample(c, samples);
1199
1200    calc_masking(c, samples);
1201    if (c->options.adpcm_mode)
1202        adpcm_analysis(c);
1203    find_peaks(c);
1204    assign_bits(c);
1205    calc_lfe_scales(c);
1206    shift_history(c, samples);
1207
1208    init_put_bits(&c->pb, avpkt->data, avpkt->size);
1209    fill_in_adpcm_bufer(c);
1210    put_frame_header(c);
1211    put_primary_audio_header(c);
1212    for (i = 0; i < SUBFRAMES; i++)
1213        put_subframe(c, i);
1214
1215    flush_put_bits(&c->pb);
1216    memset(put_bits_ptr(&c->pb), 0, put_bytes_left(&c->pb, 0));
1217
1218    avpkt->pts      = frame->pts;
1219    avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
1220    *got_packet_ptr = 1;
1221    return 0;
1222}
1223
1224#define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1225
1226static const AVOption options[] = {
1227    { "dca_adpcm", "Use ADPCM encoding", offsetof(DCAEncContext, options.adpcm_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DCAENC_FLAGS },
1228    { NULL },
1229};
1230
1231static const AVClass dcaenc_class = {
1232    .class_name = "DCA (DTS Coherent Acoustics)",
1233    .item_name = av_default_item_name,
1234    .option = options,
1235    .version = LIBAVUTIL_VERSION_INT,
1236};
1237
1238static const FFCodecDefault defaults[] = {
1239    { "b",          "1411200" },
1240    { NULL },
1241};
1242
1243const FFCodec ff_dca_encoder = {
1244    .p.name                = "dca",
1245    .p.long_name           = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
1246    .p.type                = AVMEDIA_TYPE_AUDIO,
1247    .p.id                  = AV_CODEC_ID_DTS,
1248    .p.capabilities        = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL,
1249    .priv_data_size        = sizeof(DCAEncContext),
1250    .init                  = encode_init,
1251    .close                 = encode_close,
1252    FF_CODEC_ENCODE_CB(encode_frame),
1253    .caps_internal         = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1254    .p.sample_fmts         = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
1255                                                            AV_SAMPLE_FMT_NONE },
1256    .p.supported_samplerates = sample_rates,
1257#if FF_API_OLD_CHANNEL_LAYOUT
1258    .p.channel_layouts     = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
1259                                                  AV_CH_LAYOUT_STEREO,
1260                                                  AV_CH_LAYOUT_2_2,
1261                                                  AV_CH_LAYOUT_5POINT0,
1262                                                  AV_CH_LAYOUT_5POINT1,
1263                                                  0 },
1264#endif
1265    .p.ch_layouts     = (const AVChannelLayout[]){
1266        AV_CHANNEL_LAYOUT_MONO,
1267        AV_CHANNEL_LAYOUT_STEREO,
1268        AV_CHANNEL_LAYOUT_2_2,
1269        AV_CHANNEL_LAYOUT_5POINT0,
1270        AV_CHANNEL_LAYOUT_5POINT1,
1271        { 0 },
1272    },
1273    .defaults              = defaults,
1274    .p.priv_class          = &dcaenc_class,
1275};
1276