aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/bonk.c
blob: fbea91c750f300c4b558e83b84ac8669cd22b49e (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
/*
 * Bonk audio decoder
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/internal.h"
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "codec_internal.h"
#include "decode.h"
#define BITSTREAM_READER_LE
#include "get_bits.h"
#include "bytestream.h"

typedef struct BitCount {
    uint8_t bit;
    unsigned count;
} BitCount;

typedef struct BonkContext {
    GetBitContext gb;
    int skip;

    uint8_t *bitstream;
    int64_t max_framesize;
    int bitstream_size;
    int bitstream_index;

    uint64_t nb_samples;
    int lossless;
    int mid_side;
    int n_taps;
    int down_sampling;
    int samples_per_packet;

    int state[2][2048], k[2048];
    int *samples[2];
    int *input_samples;
    uint8_t quant[2048];
    BitCount *bits;
} BonkContext;

static av_cold int bonk_close(AVCodecContext *avctx)
{
    BonkContext *s = avctx->priv_data;

    av_freep(&s->bitstream);
    av_freep(&s->input_samples);
    av_freep(&s->samples[0]);
    av_freep(&s->samples[1]);
    av_freep(&s->bits);
    s->bitstream_size = 0;

    return 0;
}

static av_cold int bonk_init(AVCodecContext *avctx)
{
    BonkContext *s = avctx->priv_data;

    avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
    if (avctx->extradata_size < 17)
        return AVERROR(EINVAL);

    if (avctx->extradata[0]) {
        av_log(avctx, AV_LOG_ERROR, "Unsupported version.\n");
        return AVERROR_INVALIDDATA;
    }

    if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2)
        return AVERROR_INVALIDDATA;

    s->nb_samples = AV_RL32(avctx->extradata + 1) / avctx->ch_layout.nb_channels;
    if (!s->nb_samples)
        s->nb_samples = UINT64_MAX;
    s->lossless = avctx->extradata[10] != 0;
    s->mid_side = avctx->extradata[11] != 0;
    s->n_taps = AV_RL16(avctx->extradata + 12);
    if (!s->n_taps || s->n_taps > 2048)
        return AVERROR(EINVAL);

    s->down_sampling = avctx->extradata[14];
    if (!s->down_sampling)
        return AVERROR(EINVAL);

    s->samples_per_packet = AV_RL16(avctx->extradata + 15);
    if (!s->samples_per_packet)
        return AVERROR(EINVAL);

    if (s->down_sampling * s->samples_per_packet < s->n_taps)
        return AVERROR_INVALIDDATA;

    s->max_framesize = s->samples_per_packet * avctx->ch_layout.nb_channels * s->down_sampling * 16LL;
    if (s->max_framesize > (INT32_MAX - AV_INPUT_BUFFER_PADDING_SIZE) / 8)
        return AVERROR_INVALIDDATA;

    s->bitstream = av_calloc(s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));
    if (!s->bitstream)
        return AVERROR(ENOMEM);

    s->input_samples = av_calloc(s->samples_per_packet, sizeof(*s->input_samples));
    if (!s->input_samples)
        return AVERROR(ENOMEM);

    s->samples[0] = av_calloc(s->samples_per_packet * s->down_sampling, sizeof(*s->samples[0]));
    s->samples[1] = av_calloc(s->samples_per_packet * s->down_sampling, sizeof(*s->samples[0]));
    if (!s->samples[0] || !s->samples[1])
        return AVERROR(ENOMEM);

    s->bits = av_calloc(s->max_framesize * 8, sizeof(*s->bits));
    if (!s->bits)
        return AVERROR(ENOMEM);

    for (int i = 0; i < 512; i++) {
        s->quant[i] = sqrt(i + 1);
    }

    return 0;
}

static unsigned read_uint_max(BonkContext *s, uint32_t max)
{
    unsigned value = 0;

    if (max == 0)
        return 0;

    av_assert0(max >> 31 == 0);

    for (unsigned i = 1; i <= max - value; i+=i)
        if (get_bits1(&s->gb))
            value += i;

    return value;
}

static int intlist_read(BonkContext *s, int *buf, int entries, int base_2_part)
{
    int i, low_bits = 0, x = 0, max_x;
    int n_zeros = 0, step = 256, dominant = 0;
    int pos = 0, level = 0;
    BitCount *bits = s->bits;
    int passes = 1;

    memset(buf, 0, entries * sizeof(*buf));
    if (base_2_part) {
        low_bits = get_bits(&s->gb, 4);

        if (low_bits)
            for (i = 0; i < entries; i++)
                buf[i] = get_bits(&s->gb, low_bits);
    }

    while (n_zeros < entries) {
        int steplet = step >> 8;

        if (get_bits_left(&s->gb) <= 0)
            return AVERROR_INVALIDDATA;

        if (!get_bits1(&s->gb)) {
            av_assert0(steplet >= 0);

            if (steplet > 0) {
                bits[x  ].bit   = dominant;
                bits[x++].count = steplet;
            }

            if (!dominant)
                n_zeros += steplet;

            if (step > INT32_MAX*8LL/9 + 1)
                return AVERROR_INVALIDDATA;
            step += step / 8;
        } else if (steplet > 0) {
            int actual_run = read_uint_max(s, steplet - 1);

            av_assert0(actual_run >= 0);

            if (actual_run > 0) {
                bits[x  ].bit   = dominant;
                bits[x++].count = actual_run;
            }

            bits[x  ].bit   = !dominant;
            bits[x++].count = 1;

            if (!dominant)
                n_zeros += actual_run;
            else
                n_zeros++;

            step -= step / 8;
        }

        if (step < 256) {
            step = 65536 / step;
            dominant = !dominant;
        }
    }

    max_x = x;
    x = 0;
    n_zeros = 0;
    for (i = 0; n_zeros < entries; i++) {
        if (x >= max_x)
            return AVERROR_INVALIDDATA;

        if (pos >= entries) {
            pos = 0;
            level += passes << low_bits;
            passes = 1;
            if (bits[x].bit && bits[x].count > entries - n_zeros)
                passes =  bits[x].count / (entries - n_zeros);
        }

        if (level > 1 << 16)
            return AVERROR_INVALIDDATA;

        if (buf[pos] >= level) {
            if (bits[x].bit)
                buf[pos] += passes << low_bits;
            else
                n_zeros++;

            av_assert1(bits[x].count >= passes);
            bits[x].count -= passes;
            x += bits[x].count == 0;
        }

        pos++;
    }

    for (i = 0; i < entries; i++) {
        if (buf[i] && get_bits1(&s->gb)) {
            buf[i] = -buf[i];
        }
    }

    return 0;
}

static inline int shift_down(int a, int b)
{
    return (a >> b) + (a < 0);
}

static inline int shift(int a, int b)
{
    return a + (1 << b - 1) >> b;
}

#define LATTICE_SHIFT 10
#define SAMPLE_SHIFT   4
#define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)

static int predictor_calc_error(int *k, int *state, int order, int error)
{
    int i, x = error - shift_down(k[order-1] * (unsigned)state[order-1], LATTICE_SHIFT);
    int *k_ptr = &(k[order-2]),
        *state_ptr = &(state[order-2]);

    for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--) {
        unsigned k_value = *k_ptr, state_value = *state_ptr;

        x -= shift_down(k_value * (unsigned)state_value, LATTICE_SHIFT);
        state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
    }

    // don't drift too far, to avoid overflows
    x = av_clip(x, -(SAMPLE_FACTOR << 16), SAMPLE_FACTOR << 16);

    state[0] = x;

    return x;
}

static void predictor_init_state(int *k, unsigned *state, int order)
{
    for (int i = order - 2; i >= 0; i--) {
        unsigned x = state[i];

        for (int j = 0, p = i + 1; p < order; j++, p++) {
            int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);

            state[p] += shift_down(k[j] * x, LATTICE_SHIFT);
            x = tmp;
        }
    }
}

static int bonk_decode(AVCodecContext *avctx, AVFrame *frame,
                       int *got_frame_ptr, AVPacket *pkt)
{
    BonkContext *s = avctx->priv_data;
    GetBitContext *gb = &s->gb;
    const uint8_t *buf;
    int quant, n, buf_size, input_buf_size;
    int ret = AVERROR_INVALIDDATA;

    if ((!pkt->size && !s->bitstream_size) || s->nb_samples == 0) {
        *got_frame_ptr = 0;
        return pkt->size;
    }

    buf_size = FFMIN(pkt->size, s->max_framesize - s->bitstream_size);
    input_buf_size = buf_size;
    if (s->bitstream_index + s->bitstream_size + buf_size + AV_INPUT_BUFFER_PADDING_SIZE > s->max_framesize) {
        memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
        s->bitstream_index = 0;
    }
    if (pkt->data)
        memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);
    buf                = &s->bitstream[s->bitstream_index];
    buf_size          += s->bitstream_size;
    s->bitstream_size  = buf_size;
    if (buf_size < s->max_framesize && pkt->data) {
        *got_frame_ptr = 0;
        return input_buf_size;
    }

    frame->nb_samples = FFMIN(s->samples_per_packet * s->down_sampling, s->nb_samples);
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        goto fail;

    if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)
        goto fail;

    skip_bits(gb, s->skip);
    if ((ret = intlist_read(s, s->k, s->n_taps, 0)) < 0)
        goto fail;

    for (int i = 0; i < s->n_taps; i++)
        s->k[i] *= s->quant[i];
    quant = s->lossless ? 1 : get_bits(&s->gb, 16) * SAMPLE_FACTOR;

    for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
        const int samples_per_packet = s->samples_per_packet;
        const int down_sampling = s->down_sampling;
        const int offset = samples_per_packet * down_sampling - 1;
        int *state = s->state[ch];
        int *sample = s->samples[ch];

        predictor_init_state(s->k, state, s->n_taps);
        if ((ret = intlist_read(s, s->input_samples, samples_per_packet, 1)) < 0)
            goto fail;

        for (int i = 0; i < samples_per_packet; i++) {
            for (int j = 0; j < s->down_sampling - 1; j++) {
                sample[0] = predictor_calc_error(s->k, state, s->n_taps, 0);
                sample++;
            }

            sample[0] = predictor_calc_error(s->k, state, s->n_taps, s->input_samples[i] * (unsigned)quant);
            sample++;
        }

        sample = s->samples[ch];
        for (int i = 0; i < s->n_taps; i++)
            state[i] = sample[offset - i];
    }

    if (s->mid_side && avctx->ch_layout.nb_channels == 2) {
        for (int i = 0; i < frame->nb_samples; i++) {
            s->samples[1][i] += shift(s->samples[0][i], 1);
            s->samples[0][i] -= s->samples[1][i];
        }
    }

    if (!s->lossless) {
        for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
            int *samples = s->samples[ch];
            for (int i = 0; i < frame->nb_samples; i++)
                samples[i] = shift(samples[i], 4);
        }
    }

    for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
        int16_t *osamples = (int16_t *)frame->extended_data[ch];
        int *samples = s->samples[ch];
        for (int i = 0; i < frame->nb_samples; i++)
            osamples[i] = av_clip_int16(samples[i]);
    }

    s->nb_samples -= frame->nb_samples;

    s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
    n = get_bits_count(gb) / 8;

    if (n > buf_size) {
fail:
        s->bitstream_size = 0;
        s->bitstream_index = 0;
        return AVERROR_INVALIDDATA;
    }

    *got_frame_ptr = 1;

    if (s->bitstream_size) {
        s->bitstream_index += n;
        s->bitstream_size  -= n;
        return input_buf_size;
    }
    return n;
}

const FFCodec ff_bonk_decoder = {
    .p.name           = "bonk",
    CODEC_LONG_NAME("Bonk audio"),
    .p.type           = AVMEDIA_TYPE_AUDIO,
    .p.id             = AV_CODEC_ID_BONK,
    .priv_data_size   = sizeof(BonkContext),
    .init             = bonk_init,
    FF_CODEC_DECODE_CB(bonk_decode),
    .close            = bonk_close,
    .p.capabilities   = AV_CODEC_CAP_DELAY |
#if FF_API_SUBFRAMES
                        AV_CODEC_CAP_SUBFRAMES |
#endif
                        AV_CODEC_CAP_DR1,
    .caps_internal    = FF_CODEC_CAP_INIT_CLEANUP,
    .p.sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
                                                        AV_SAMPLE_FMT_NONE },
};