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1
/*
2
 * AAC encoder
3
 * Copyright (C) 2008 Konstantin Shishkov
4
 *
5
 * This file is part of FFmpeg.
6
 *
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 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
9
 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
11
 *
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 * FFmpeg is distributed in the hope that it will be useful,
13
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15
 * Lesser General Public License for more details.
16
 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20
 */
21

    
22
/**
23
 * @file libavcodec/aacenc.c
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 * AAC encoder
25
 */
26

    
27
/***********************************
28
 *              TODOs:
29
 * add sane pulse detection
30
 * add temporal noise shaping
31
 ***********************************/
32

    
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#include "avcodec.h"
34
#include "put_bits.h"
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#include "dsputil.h"
36
#include "mpeg4audio.h"
37

    
38
#include "aac.h"
39
#include "aactab.h"
40
#include "aacenc.h"
41

    
42
#include "psymodel.h"
43

    
44
static const uint8_t swb_size_1024_96[] = {
45
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
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    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
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    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
48
};
49

    
50
static const uint8_t swb_size_1024_64[] = {
51
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
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    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
53
    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
54
};
55

    
56
static const uint8_t swb_size_1024_48[] = {
57
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
59
    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
60
    96
61
};
62

    
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static const uint8_t swb_size_1024_32[] = {
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    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
67
};
68

    
69
static const uint8_t swb_size_1024_24[] = {
70
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
71
    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
72
    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
73
};
74

    
75
static const uint8_t swb_size_1024_16[] = {
76
    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
77
    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
78
    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
79
};
80

    
81
static const uint8_t swb_size_1024_8[] = {
82
    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
83
    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
84
    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
85
};
86

    
87
static const uint8_t *swb_size_1024[] = {
88
    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
89
    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
90
    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
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    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
92
};
93

    
94
static const uint8_t swb_size_128_96[] = {
95
    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
96
};
97

    
98
static const uint8_t swb_size_128_48[] = {
99
    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
100
};
101

    
102
static const uint8_t swb_size_128_24[] = {
103
    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
104
};
105

    
106
static const uint8_t swb_size_128_16[] = {
107
    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
108
};
109

    
110
static const uint8_t swb_size_128_8[] = {
111
    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
112
};
113

    
114
static const uint8_t *swb_size_128[] = {
115
    /* the last entry on the following row is swb_size_128_64 but is a
116
       duplicate of swb_size_128_96 */
117
    swb_size_128_96, swb_size_128_96, swb_size_128_96,
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    swb_size_128_48, swb_size_128_48, swb_size_128_48,
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    swb_size_128_24, swb_size_128_24, swb_size_128_16,
120
    swb_size_128_16, swb_size_128_16, swb_size_128_8
121
};
122

    
123
/** default channel configurations */
124
static const uint8_t aac_chan_configs[6][5] = {
125
 {1, TYPE_SCE},                               // 1 channel  - single channel element
126
 {1, TYPE_CPE},                               // 2 channels - channel pair
127
 {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
128
 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
129
 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
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 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
131
};
132

    
133
/**
134
 * Make AAC audio config object.
135
 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
136
 */
137
static void put_audio_specific_config(AVCodecContext *avctx)
138
{
139
    PutBitContext pb;
140
    AACEncContext *s = avctx->priv_data;
141

    
142
    init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
143
    put_bits(&pb, 5, 2); //object type - AAC-LC
144
    put_bits(&pb, 4, s->samplerate_index); //sample rate index
145
    put_bits(&pb, 4, avctx->channels);
146
    //GASpecificConfig
147
    put_bits(&pb, 1, 0); //frame length - 1024 samples
148
    put_bits(&pb, 1, 0); //does not depend on core coder
149
    put_bits(&pb, 1, 0); //is not extension
150
    flush_put_bits(&pb);
151
}
152

    
153
static av_cold int aac_encode_init(AVCodecContext *avctx)
154
{
155
    AACEncContext *s = avctx->priv_data;
156
    int i;
157
    const uint8_t *sizes[2];
158
    int lengths[2];
159

    
160
    avctx->frame_size = 1024;
161

    
162
    for(i = 0; i < 16; i++)
163
        if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
164
            break;
165
    if(i == 16){
166
        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
167
        return -1;
168
    }
169
    if(avctx->channels > 6){
170
        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
171
        return -1;
172
    }
173
    s->samplerate_index = i;
174

    
175
    dsputil_init(&s->dsp, avctx);
176
    ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
177
    ff_mdct_init(&s->mdct128,   8, 0, 1.0);
178
    // window init
179
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
180
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
181
    ff_sine_window_init(ff_sine_1024, 1024);
182
    ff_sine_window_init(ff_sine_128, 128);
183

    
184
    s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
185
    s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
186
    avctx->extradata = av_malloc(2);
187
    avctx->extradata_size = 2;
188
    put_audio_specific_config(avctx);
189

    
190
    sizes[0] = swb_size_1024[i];
191
    sizes[1] = swb_size_128[i];
192
    lengths[0] = ff_aac_num_swb_1024[i];
193
    lengths[1] = ff_aac_num_swb_128[i];
194
    ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
195
    s->psypp = ff_psy_preprocess_init(avctx);
196
    s->coder = &ff_aac_coders[0];
197

    
198
    s->lambda = avctx->global_quality ? avctx->global_quality : 120;
199
#if !CONFIG_HARDCODED_TABLES
200
    for (i = 0; i < 428; i++)
201
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
202
#endif /* CONFIG_HARDCODED_TABLES */
203

    
204
    if (avctx->channels > 5)
205
        av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
206
               "The output will most likely be an illegal bitstream.\n");
207

    
208
    return 0;
209
}
210

    
211
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
212
                                  SingleChannelElement *sce, short *audio, int channel)
213
{
214
    int i, j, k;
215
    const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
216
    const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
217
    const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
218

    
219
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
220
        memcpy(s->output, sce->saved, sizeof(float)*1024);
221
        if(sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE){
222
            memset(s->output, 0, sizeof(s->output[0]) * 448);
223
            for(i = 448; i < 576; i++)
224
                s->output[i] = sce->saved[i] * pwindow[i - 448];
225
            for(i = 576; i < 704; i++)
226
                s->output[i] = sce->saved[i];
227
        }
228
        if(sce->ics.window_sequence[0] != LONG_START_SEQUENCE){
229
            j = channel;
230
            for (i = 0; i < 1024; i++, j += avctx->channels){
231
                s->output[i+1024]         = audio[j] * lwindow[1024 - i - 1];
232
                sce->saved[i] = audio[j] * lwindow[i];
233
            }
234
        }else{
235
            j = channel;
236
            for(i = 0; i < 448; i++, j += avctx->channels)
237
                s->output[i+1024]         = audio[j];
238
            for(i = 448; i < 576; i++, j += avctx->channels)
239
                s->output[i+1024]         = audio[j] * swindow[576 - i - 1];
240
            memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
241
            j = channel;
242
            for(i = 0; i < 1024; i++, j += avctx->channels)
243
                sce->saved[i] = audio[j];
244
        }
245
        ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
246
    }else{
247
        j = channel;
248
        for (k = 0; k < 1024; k += 128) {
249
            for(i = 448 + k; i < 448 + k + 256; i++)
250
                s->output[i - 448 - k] = (i < 1024)
251
                                         ? sce->saved[i]
252
                                         : audio[channel + (i-1024)*avctx->channels];
253
            s->dsp.vector_fmul        (s->output,     k ?  swindow : pwindow, 128);
254
            s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
255
            ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
256
        }
257
        j = channel;
258
        for(i = 0; i < 1024; i++, j += avctx->channels)
259
            sce->saved[i] = audio[j];
260
    }
261
}
262

    
263
/**
264
 * Encode ics_info element.
265
 * @see Table 4.6 (syntax of ics_info)
266
 */
267
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
268
{
269
    int w;
270

    
271
    put_bits(&s->pb, 1, 0);                // ics_reserved bit
272
    put_bits(&s->pb, 2, info->window_sequence[0]);
273
    put_bits(&s->pb, 1, info->use_kb_window[0]);
274
    if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
275
        put_bits(&s->pb, 6, info->max_sfb);
276
        put_bits(&s->pb, 1, 0);            // no prediction
277
    }else{
278
        put_bits(&s->pb, 4, info->max_sfb);
279
        for(w = 1; w < 8; w++){
280
            put_bits(&s->pb, 1, !info->group_len[w]);
281
        }
282
    }
283
}
284

    
285
/**
286
 * Encode MS data.
287
 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
288
 */
289
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
290
{
291
    int i, w;
292

    
293
    put_bits(pb, 2, cpe->ms_mode);
294
    if(cpe->ms_mode == 1){
295
        for(w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w]){
296
            for(i = 0; i < cpe->ch[0].ics.max_sfb; i++)
297
                put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
298
        }
299
    }
300
}
301

    
302
/**
303
 * Produce integer coefficients from scalefactors provided by the model.
304
 */
305
static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
306
{
307
    int i, w, w2, g, ch;
308
    int start, sum, maxsfb, cmaxsfb;
309

    
310
    for(ch = 0; ch < chans; ch++){
311
        IndividualChannelStream *ics = &cpe->ch[ch].ics;
312
        start = 0;
313
        maxsfb = 0;
314
        cpe->ch[ch].pulse.num_pulse = 0;
315
        for(w = 0; w < ics->num_windows*16; w += 16){
316
            for(g = 0; g < ics->num_swb; g++){
317
                sum = 0;
318
                //apply M/S
319
                if(!ch && cpe->ms_mask[w + g]){
320
                    for(i = 0; i < ics->swb_sizes[g]; i++){
321
                        cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
322
                        cpe->ch[1].coeffs[start+i] =  cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
323
                    }
324
                }
325
                start += ics->swb_sizes[g];
326
            }
327
            for(cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--);
328
            maxsfb = FFMAX(maxsfb, cmaxsfb);
329
        }
330
        ics->max_sfb = maxsfb;
331

    
332
        //adjust zero bands for window groups
333
        for(w = 0; w < ics->num_windows; w += ics->group_len[w]){
334
            for(g = 0; g < ics->max_sfb; g++){
335
                i = 1;
336
                for(w2 = w; w2 < w + ics->group_len[w]; w2++){
337
                    if(!cpe->ch[ch].zeroes[w2*16 + g]){
338
                        i = 0;
339
                        break;
340
                    }
341
                }
342
                cpe->ch[ch].zeroes[w*16 + g] = i;
343
            }
344
        }
345
    }
346

    
347
    if(chans > 1 && cpe->common_window){
348
        IndividualChannelStream *ics0 = &cpe->ch[0].ics;
349
        IndividualChannelStream *ics1 = &cpe->ch[1].ics;
350
        int msc = 0;
351
        ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
352
        ics1->max_sfb = ics0->max_sfb;
353
        for(w = 0; w < ics0->num_windows*16; w += 16)
354
            for(i = 0; i < ics0->max_sfb; i++)
355
                if(cpe->ms_mask[w+i]) msc++;
356
        if(msc == 0 || ics0->max_sfb == 0) cpe->ms_mode = 0;
357
        else cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
358
    }
359
}
360

    
361
/**
362
 * Encode scalefactor band coding type.
363
 */
364
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
365
{
366
    int w;
367

    
368
    for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
369
        s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
370
    }
371
}
372

    
373
/**
374
 * Encode scalefactors.
375
 */
376
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
377
{
378
    int off = sce->sf_idx[0], diff;
379
    int i, w;
380

    
381
    for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
382
        for(i = 0; i < sce->ics.max_sfb; i++){
383
            if(!sce->zeroes[w*16 + i]){
384
                diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
385
                if(diff < 0 || diff > 120) av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
386
                off = sce->sf_idx[w*16 + i];
387
                put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
388
            }
389
        }
390
    }
391
}
392

    
393
/**
394
 * Encode pulse data.
395
 */
396
static void encode_pulses(AACEncContext *s, Pulse *pulse)
397
{
398
    int i;
399

    
400
    put_bits(&s->pb, 1, !!pulse->num_pulse);
401
    if(!pulse->num_pulse) return;
402

    
403
    put_bits(&s->pb, 2, pulse->num_pulse - 1);
404
    put_bits(&s->pb, 6, pulse->start);
405
    for(i = 0; i < pulse->num_pulse; i++){
406
        put_bits(&s->pb, 5, pulse->pos[i]);
407
        put_bits(&s->pb, 4, pulse->amp[i]);
408
    }
409
}
410

    
411
/**
412
 * Encode spectral coefficients processed by psychoacoustic model.
413
 */
414
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
415
{
416
    int start, i, w, w2;
417

    
418
    for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
419
        start = 0;
420
        for(i = 0; i < sce->ics.max_sfb; i++){
421
            if(sce->zeroes[w*16 + i]){
422
                start += sce->ics.swb_sizes[i];
423
                continue;
424
            }
425
            for(w2 = w; w2 < w + sce->ics.group_len[w]; w2++){
426
                s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
427
                                         sce->ics.swb_sizes[i],
428
                                         sce->sf_idx[w*16 + i],
429
                                         sce->band_type[w*16 + i],
430
                                         s->lambda);
431
            }
432
            start += sce->ics.swb_sizes[i];
433
        }
434
    }
435
}
436

    
437
/**
438
 * Encode one channel of audio data.
439
 */
440
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
441
{
442
    put_bits(&s->pb, 8, sce->sf_idx[0]);
443
    if(!common_window) put_ics_info(s, &sce->ics);
444
    encode_band_info(s, sce);
445
    encode_scale_factors(avctx, s, sce);
446
    encode_pulses(s, &sce->pulse);
447
    put_bits(&s->pb, 1, 0); //tns
448
    put_bits(&s->pb, 1, 0); //ssr
449
    encode_spectral_coeffs(s, sce);
450
    return 0;
451
}
452

    
453
/**
454
 * Write some auxiliary information about the created AAC file.
455
 */
456
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
457
{
458
    int i, namelen, padbits;
459

    
460
    namelen = strlen(name) + 2;
461
    put_bits(&s->pb, 3, TYPE_FIL);
462
    put_bits(&s->pb, 4, FFMIN(namelen, 15));
463
    if(namelen >= 15)
464
        put_bits(&s->pb, 8, namelen - 16);
465
    put_bits(&s->pb, 4, 0); //extension type - filler
466
    padbits = 8 - (put_bits_count(&s->pb) & 7);
467
    align_put_bits(&s->pb);
468
    for(i = 0; i < namelen - 2; i++)
469
        put_bits(&s->pb, 8, name[i]);
470
    put_bits(&s->pb, 12 - padbits, 0);
471
}
472

    
473
static int aac_encode_frame(AVCodecContext *avctx,
474
                            uint8_t *frame, int buf_size, void *data)
475
{
476
    AACEncContext *s = avctx->priv_data;
477
    int16_t *samples = s->samples, *samples2, *la;
478
    ChannelElement *cpe;
479
    int i, j, chans, tag, start_ch;
480
    const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
481
    int chan_el_counter[4];
482

    
483
    if(s->last_frame)
484
        return 0;
485
    if(data){
486
        if(!s->psypp){
487
            memcpy(s->samples + 1024 * avctx->channels, data, 1024 * avctx->channels * sizeof(s->samples[0]));
488
        }else{
489
            start_ch = 0;
490
            samples2 = s->samples + 1024 * avctx->channels;
491
            for(i = 0; i < chan_map[0]; i++){
492
                tag = chan_map[i+1];
493
                chans = tag == TYPE_CPE ? 2 : 1;
494
                ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, samples2 + start_ch, start_ch, chans);
495
                start_ch += chans;
496
            }
497
        }
498
    }
499
    if(!avctx->frame_number){
500
        memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
501
        return 0;
502
    }
503

    
504
    init_put_bits(&s->pb, frame, buf_size*8);
505
    if((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)){
506
        put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
507
    }
508
    start_ch = 0;
509
    memset(chan_el_counter, 0, sizeof(chan_el_counter));
510
    for(i = 0; i < chan_map[0]; i++){
511
        FFPsyWindowInfo wi[2];
512
        tag = chan_map[i+1];
513
        chans = tag == TYPE_CPE ? 2 : 1;
514
        cpe = &s->cpe[i];
515
        samples2 = samples + start_ch;
516
        la = samples2 + 1024 * avctx->channels + start_ch;
517
        if(!data) la = NULL;
518
        for(j = 0; j < chans; j++){
519
            IndividualChannelStream *ics = &cpe->ch[j].ics;
520
            int k;
521
            wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
522
            ics->window_sequence[1] = ics->window_sequence[0];
523
            ics->window_sequence[0] = wi[j].window_type[0];
524
            ics->use_kb_window[1]   = ics->use_kb_window[0];
525
            ics->use_kb_window[0]   = wi[j].window_shape;
526
            ics->num_windows        = wi[j].num_windows;
527
            ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
528
            ics->num_swb            = s->psy.num_bands[ics->num_windows == 8];
529
            for(k = 0; k < ics->num_windows; k++)
530
                ics->group_len[k] = wi[j].grouping[k];
531

    
532
            s->cur_channel = start_ch + j;
533
            apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
534
            s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
535
        }
536
        cpe->common_window = 0;
537
        if(chans > 1
538
            && wi[0].window_type[0] == wi[1].window_type[0]
539
            && wi[0].window_shape   == wi[1].window_shape){
540

    
541
            cpe->common_window = 1;
542
            for(j = 0; j < wi[0].num_windows; j++){
543
                if(wi[0].grouping[j] != wi[1].grouping[j]){
544
                    cpe->common_window = 0;
545
                    break;
546
                }
547
            }
548
        }
549
        if(cpe->common_window && s->coder->search_for_ms)
550
            s->coder->search_for_ms(s, cpe, s->lambda);
551
        adjust_frame_information(s, cpe, chans);
552
        put_bits(&s->pb, 3, tag);
553
        put_bits(&s->pb, 4, chan_el_counter[tag]++);
554
        if(chans == 2){
555
            put_bits(&s->pb, 1, cpe->common_window);
556
            if(cpe->common_window){
557
                put_ics_info(s, &cpe->ch[0].ics);
558
                encode_ms_info(&s->pb, cpe);
559
            }
560
        }
561
        for(j = 0; j < chans; j++){
562
            s->cur_channel = start_ch + j;
563
            ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
564
            encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
565
        }
566
        start_ch += chans;
567
    }
568

    
569
    put_bits(&s->pb, 3, TYPE_END);
570
    flush_put_bits(&s->pb);
571
    avctx->frame_bits = put_bits_count(&s->pb);
572

    
573
    // rate control stuff
574
    if(!(avctx->flags & CODEC_FLAG_QSCALE)){
575
        float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
576
        s->lambda *= ratio;
577
    }
578

    
579
    if (avctx->frame_bits > 6144*avctx->channels) {
580
        av_log(avctx, AV_LOG_ERROR, "input buffer violation %d > %d.\n", avctx->frame_bits, 6144*avctx->channels);
581
    }
582

    
583
    if(!data)
584
        s->last_frame = 1;
585
    memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
586
    return put_bits_count(&s->pb)>>3;
587
}
588

    
589
static av_cold int aac_encode_end(AVCodecContext *avctx)
590
{
591
    AACEncContext *s = avctx->priv_data;
592

    
593
    ff_mdct_end(&s->mdct1024);
594
    ff_mdct_end(&s->mdct128);
595
    ff_psy_end(&s->psy);
596
    ff_psy_preprocess_end(s->psypp);
597
    av_freep(&s->samples);
598
    av_freep(&s->cpe);
599
    return 0;
600
}
601

    
602
AVCodec aac_encoder = {
603
    "aac",
604
    CODEC_TYPE_AUDIO,
605
    CODEC_ID_AAC,
606
    sizeof(AACEncContext),
607
    aac_encode_init,
608
    aac_encode_frame,
609
    aac_encode_end,
610
    .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
611
    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
612
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
613
};