1/*
2 * Atrac 1 compatible decoder
3 * Copyright (c) 2009 Maxim Poliakovski
4 * Copyright (c) 2009 Benjamin Larsson
5 *
6 * This file is part of FFmpeg.
7 *
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23/**
24 * @file
25 * Atrac 1 compatible decoder.
26 * This decoder handles raw ATRAC1 data and probably SDDS data.
27 */
28
29/* Many thanks to Tim Craig for all the help! */
30
31#include <math.h>
32#include <stddef.h>
33#include <stdio.h>
34
35#include "avcodec.h"
36#include "get_bits.h"
37#include "dsputil.h"
38#include "fft.h"
39
40#include "atrac.h"
41#include "atrac1data.h"
42
43#define AT1_MAX_BFU      52                 ///< max number of block floating units in a sound unit
44#define AT1_SU_SIZE      212                ///< number of bytes in a sound unit
45#define AT1_SU_SAMPLES   512                ///< number of samples in a sound unit
46#define AT1_FRAME_SIZE   AT1_SU_SIZE * 2
47#define AT1_SU_MAX_BITS  AT1_SU_SIZE * 8
48#define AT1_MAX_CHANNELS 2
49
50#define AT1_QMF_BANDS    3
51#define IDX_LOW_BAND     0
52#define IDX_MID_BAND     1
53#define IDX_HIGH_BAND    2
54
55/**
56 * Sound unit struct, one unit is used per channel
57 */
58typedef struct {
59    int                 log2_block_count[AT1_QMF_BANDS];    ///< log2 number of blocks in a band
60    int                 num_bfus;                           ///< number of Block Floating Units
61    float*              spectrum[2];
62    DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES];     ///< mdct buffer
63    DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES];     ///< mdct buffer
64    DECLARE_ALIGNED(16, float, fst_qmf_delay)[46];         ///< delay line for the 1st stacked QMF filter
65    DECLARE_ALIGNED(16, float, snd_qmf_delay)[46];         ///< delay line for the 2nd stacked QMF filter
66    DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23];    ///< delay line for the last stacked QMF filter
67} AT1SUCtx;
68
69/**
70 * The atrac1 context, holds all needed parameters for decoding
71 */
72typedef struct {
73    AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit
74    DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer
75
76    DECLARE_ALIGNED(16, float,  low)[256];
77    DECLARE_ALIGNED(16, float,  mid)[256];
78    DECLARE_ALIGNED(16, float, high)[512];
79    float*              bands[3];
80    DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
81    FFTContext          mdct_ctx[3];
82    int                 channels;
83    DSPContext          dsp;
84} AT1Ctx;
85
86/** size of the transform in samples in the long mode for each QMF band */
87static const uint16_t samples_per_band[3] = {128, 128, 256};
88static const uint8_t   mdct_long_nbits[3] = {7, 7, 8};
89
90
91static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
92                      int rev_spec)
93{
94    FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
95    int transf_size = 1 << nbits;
96
97    if (rev_spec) {
98        int i;
99        for (i = 0; i < transf_size / 2; i++)
100            FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
101    }
102    ff_imdct_half(mdct_context, out, spec);
103}
104
105
106static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
107{
108    int          band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
109    unsigned int start_pos, ref_pos = 0, pos = 0;
110
111    for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
112        float *prev_buf;
113        int j;
114
115        band_samples = samples_per_band[band_num];
116        log2_block_count = su->log2_block_count[band_num];
117
118        /* number of mdct blocks in the current QMF band: 1 - for long mode */
119        /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
120        num_blocks = 1 << log2_block_count;
121
122        if (num_blocks == 1) {
123            /* mdct block size in samples: 128 (long mode, low & mid bands), */
124            /* 256 (long mode, high band) and 32 (short mode, all bands) */
125            block_size = band_samples >> log2_block_count;
126
127            /* calc transform size in bits according to the block_size_mode */
128            nbits = mdct_long_nbits[band_num] - log2_block_count;
129
130            if (nbits != 5 && nbits != 7 && nbits != 8)
131                return -1;
132        } else {
133            block_size = 32;
134            nbits = 5;
135        }
136
137        start_pos = 0;
138        prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
139        for (j=0; j < num_blocks; j++) {
140            at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
141
142            /* overlap and window */
143            q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
144                                      &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
145
146            prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
147            start_pos += block_size;
148            pos += block_size;
149        }
150
151        if (num_blocks == 1)
152            memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
153
154        ref_pos += band_samples;
155    }
156
157    /* Swap buffers so the mdct overlap works */
158    FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
159
160    return 0;
161}
162
163/**
164 * Parse the block size mode byte
165 */
166
167static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
168{
169    int log2_block_count_tmp, i;
170
171    for (i = 0; i < 2; i++) {
172        /* low and mid band */
173        log2_block_count_tmp = get_bits(gb, 2);
174        if (log2_block_count_tmp & 1)
175            return -1;
176        log2_block_cnt[i] = 2 - log2_block_count_tmp;
177    }
178
179    /* high band */
180    log2_block_count_tmp = get_bits(gb, 2);
181    if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
182        return -1;
183    log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
184
185    skip_bits(gb, 2);
186    return 0;
187}
188
189
190static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
191                              float spec[AT1_SU_SAMPLES])
192{
193    int bits_used, band_num, bfu_num, i;
194    uint8_t idwls[AT1_MAX_BFU];                 ///< the word length indexes for each BFU
195    uint8_t idsfs[AT1_MAX_BFU];                 ///< the scalefactor indexes for each BFU
196
197    /* parse the info byte (2nd byte) telling how much BFUs were coded */
198    su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
199
200    /* calc number of consumed bits:
201        num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
202        + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
203    bits_used = su->num_bfus * 10 + 32 +
204                bfu_amount_tab2[get_bits(gb, 2)] +
205                (bfu_amount_tab3[get_bits(gb, 3)] << 1);
206
207    /* get word length index (idwl) for each BFU */
208    for (i = 0; i < su->num_bfus; i++)
209        idwls[i] = get_bits(gb, 4);
210
211    /* get scalefactor index (idsf) for each BFU */
212    for (i = 0; i < su->num_bfus; i++)
213        idsfs[i] = get_bits(gb, 6);
214
215    /* zero idwl/idsf for empty BFUs */
216    for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
217        idwls[i] = idsfs[i] = 0;
218
219    /* read in the spectral data and reconstruct MDCT spectrum of this channel */
220    for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
221        for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
222            int pos;
223
224            int num_specs = specs_per_bfu[bfu_num];
225            int word_len  = !!idwls[bfu_num] + idwls[bfu_num];
226            float scale_factor = sf_table[idsfs[bfu_num]];
227            bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
228
229            /* check for bitstream overflow */
230            if (bits_used > AT1_SU_MAX_BITS)
231                return -1;
232
233            /* get the position of the 1st spec according to the block size mode */
234            pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
235
236            if (word_len) {
237                float   max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
238
239                for (i = 0; i < num_specs; i++) {
240                    /* read in a quantized spec and convert it to
241                     * signed int and then inverse quantization
242                     */
243                    spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
244                }
245            } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
246                memset(&spec[pos], 0, num_specs * sizeof(float));
247            }
248        }
249    }
250
251    return 0;
252}
253
254
255static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
256{
257    float temp[256];
258    float iqmf_temp[512 + 46];
259
260    /* combine low and middle bands */
261    atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
262
263    /* delay the signal of the high band by 23 samples */
264    memcpy( su->last_qmf_delay,    &su->last_qmf_delay[256], sizeof(float) *  23);
265    memcpy(&su->last_qmf_delay[23], q->bands[2],             sizeof(float) * 256);
266
267    /* combine (low + middle) and high bands */
268    atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
269}
270
271
272static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
273                               int *data_size, AVPacket *avpkt)
274{
275    const uint8_t *buf = avpkt->data;
276    int buf_size       = avpkt->size;
277    AT1Ctx *q          = avctx->priv_data;
278    int ch, ret, i;
279    GetBitContext gb;
280    float* samples = data;
281
282
283    if (buf_size < 212 * q->channels) {
284        av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
285        return -1;
286    }
287
288    for (ch = 0; ch < q->channels; ch++) {
289        AT1SUCtx* su = &q->SUs[ch];
290
291        init_get_bits(&gb, &buf[212 * ch], 212 * 8);
292
293        /* parse block_size_mode, 1st byte */
294        ret = at1_parse_bsm(&gb, su->log2_block_count);
295        if (ret < 0)
296            return ret;
297
298        ret = at1_unpack_dequant(&gb, su, q->spec);
299        if (ret < 0)
300            return ret;
301
302        ret = at1_imdct_block(su, q);
303        if (ret < 0)
304            return ret;
305        at1_subband_synthesis(q, su, q->out_samples[ch]);
306    }
307
308    /* interleave; FIXME, should create/use a DSP function */
309    if (q->channels == 1) {
310        /* mono */
311        memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
312    } else {
313        /* stereo */
314        for (i = 0; i < AT1_SU_SAMPLES; i++) {
315            samples[i * 2]     = q->out_samples[0][i];
316            samples[i * 2 + 1] = q->out_samples[1][i];
317        }
318    }
319
320    *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
321    return avctx->block_align;
322}
323
324
325static av_cold int atrac1_decode_init(AVCodecContext *avctx)
326{
327    AT1Ctx *q = avctx->priv_data;
328
329    avctx->sample_fmt = SAMPLE_FMT_FLT;
330
331    q->channels = avctx->channels;
332
333    /* Init the mdct transforms */
334    ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
335    ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
336    ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
337
338    ff_init_ff_sine_windows(5);
339
340    atrac_generate_tables();
341
342    dsputil_init(&q->dsp, avctx);
343
344    q->bands[0] = q->low;
345    q->bands[1] = q->mid;
346    q->bands[2] = q->high;
347
348    /* Prepare the mdct overlap buffers */
349    q->SUs[0].spectrum[0] = q->SUs[0].spec1;
350    q->SUs[0].spectrum[1] = q->SUs[0].spec2;
351    q->SUs[1].spectrum[0] = q->SUs[1].spec1;
352    q->SUs[1].spectrum[1] = q->SUs[1].spec2;
353
354    return 0;
355}
356
357
358static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
359    AT1Ctx *q = avctx->priv_data;
360
361    ff_mdct_end(&q->mdct_ctx[0]);
362    ff_mdct_end(&q->mdct_ctx[1]);
363    ff_mdct_end(&q->mdct_ctx[2]);
364    return 0;
365}
366
367
368AVCodec atrac1_decoder = {
369    .name = "atrac1",
370    .type = AVMEDIA_TYPE_AUDIO,
371    .id = CODEC_ID_ATRAC1,
372    .priv_data_size = sizeof(AT1Ctx),
373    .init = atrac1_decode_init,
374    .close = atrac1_decode_end,
375    .decode = atrac1_decode_frame,
376    .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
377};
378