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2ba45a60 DM |
1 | /* |
2 | * ATRAC1 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 | * ATRAC1 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 "libavutil/float_dsp.h" | |
36 | #include "avcodec.h" | |
37 | #include "get_bits.h" | |
38 | #include "fft.h" | |
39 | #include "internal.h" | |
40 | #include "sinewin.h" | |
41 | ||
42 | #include "atrac.h" | |
43 | #include "atrac1data.h" | |
44 | ||
45 | #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit | |
46 | #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit | |
47 | #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit | |
48 | #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 | |
49 | #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 | |
50 | #define AT1_MAX_CHANNELS 2 | |
51 | ||
52 | #define AT1_QMF_BANDS 3 | |
53 | #define IDX_LOW_BAND 0 | |
54 | #define IDX_MID_BAND 1 | |
55 | #define IDX_HIGH_BAND 2 | |
56 | ||
57 | /** | |
58 | * Sound unit struct, one unit is used per channel | |
59 | */ | |
60 | typedef struct { | |
61 | int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band | |
62 | int num_bfus; ///< number of Block Floating Units | |
63 | float* spectrum[2]; | |
64 | DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer | |
65 | DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer | |
66 | DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter | |
67 | DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter | |
68 | DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter | |
69 | } AT1SUCtx; | |
70 | ||
71 | /** | |
72 | * The atrac1 context, holds all needed parameters for decoding | |
73 | */ | |
74 | typedef struct { | |
75 | AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit | |
76 | DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer | |
77 | ||
78 | DECLARE_ALIGNED(32, float, low)[256]; | |
79 | DECLARE_ALIGNED(32, float, mid)[256]; | |
80 | DECLARE_ALIGNED(32, float, high)[512]; | |
81 | float* bands[3]; | |
82 | FFTContext mdct_ctx[3]; | |
f6fa7814 | 83 | AVFloatDSPContext *fdsp; |
2ba45a60 DM |
84 | } AT1Ctx; |
85 | ||
86 | /** size of the transform in samples in the long mode for each QMF band */ | |
87 | static const uint16_t samples_per_band[3] = {128, 128, 256}; | |
88 | static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; | |
89 | ||
90 | ||
91 | static 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 | mdct_context->imdct_half(mdct_context, out, spec); | |
103 | } | |
104 | ||
105 | ||
106 | static 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 AVERROR_INVALIDDATA; | |
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 */ | |
f6fa7814 | 143 | q->fdsp->vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, |
2ba45a60 DM |
144 | &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 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 | ||
167 | static 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 AVERROR_INVALIDDATA; | |
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 AVERROR_INVALIDDATA; | |
183 | log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; | |
184 | ||
185 | skip_bits(gb, 2); | |
186 | return 0; | |
187 | } | |
188 | ||
189 | ||
190 | static 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 = ff_atrac_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 AVERROR_INVALIDDATA; | |
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 empty BFU */ | |
246 | memset(&spec[pos], 0, num_specs * sizeof(float)); | |
247 | } | |
248 | } | |
249 | } | |
250 | ||
251 | return 0; | |
252 | } | |
253 | ||
254 | ||
255 | static 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 | ff_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 | ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); | |
269 | } | |
270 | ||
271 | ||
272 | static int atrac1_decode_frame(AVCodecContext *avctx, void *data, | |
273 | int *got_frame_ptr, AVPacket *avpkt) | |
274 | { | |
275 | AVFrame *frame = data; | |
276 | const uint8_t *buf = avpkt->data; | |
277 | int buf_size = avpkt->size; | |
278 | AT1Ctx *q = avctx->priv_data; | |
279 | int ch, ret; | |
280 | GetBitContext gb; | |
281 | ||
282 | ||
283 | if (buf_size < 212 * avctx->channels) { | |
284 | av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n"); | |
285 | return AVERROR_INVALIDDATA; | |
286 | } | |
287 | ||
288 | /* get output buffer */ | |
289 | frame->nb_samples = AT1_SU_SAMPLES; | |
290 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) | |
291 | return ret; | |
292 | ||
293 | for (ch = 0; ch < avctx->channels; ch++) { | |
294 | AT1SUCtx* su = &q->SUs[ch]; | |
295 | ||
296 | init_get_bits(&gb, &buf[212 * ch], 212 * 8); | |
297 | ||
298 | /* parse block_size_mode, 1st byte */ | |
299 | ret = at1_parse_bsm(&gb, su->log2_block_count); | |
300 | if (ret < 0) | |
301 | return ret; | |
302 | ||
303 | ret = at1_unpack_dequant(&gb, su, q->spec); | |
304 | if (ret < 0) | |
305 | return ret; | |
306 | ||
307 | ret = at1_imdct_block(su, q); | |
308 | if (ret < 0) | |
309 | return ret; | |
310 | at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]); | |
311 | } | |
312 | ||
313 | *got_frame_ptr = 1; | |
314 | ||
315 | return avctx->block_align; | |
316 | } | |
317 | ||
318 | ||
319 | static av_cold int atrac1_decode_end(AVCodecContext * avctx) | |
320 | { | |
321 | AT1Ctx *q = avctx->priv_data; | |
322 | ||
323 | ff_mdct_end(&q->mdct_ctx[0]); | |
324 | ff_mdct_end(&q->mdct_ctx[1]); | |
325 | ff_mdct_end(&q->mdct_ctx[2]); | |
326 | ||
f6fa7814 DM |
327 | av_freep(&q->fdsp); |
328 | ||
2ba45a60 DM |
329 | return 0; |
330 | } | |
331 | ||
332 | ||
333 | static av_cold int atrac1_decode_init(AVCodecContext *avctx) | |
334 | { | |
335 | AT1Ctx *q = avctx->priv_data; | |
336 | int ret; | |
337 | ||
338 | avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; | |
339 | ||
340 | if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) { | |
341 | av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", | |
342 | avctx->channels); | |
343 | return AVERROR(EINVAL); | |
344 | } | |
345 | ||
346 | if (avctx->block_align <= 0) { | |
347 | av_log(avctx, AV_LOG_ERROR, "Unsupported block align."); | |
348 | return AVERROR_PATCHWELCOME; | |
349 | } | |
350 | ||
351 | /* Init the mdct transforms */ | |
352 | if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) || | |
353 | (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) || | |
354 | (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) { | |
355 | av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); | |
356 | atrac1_decode_end(avctx); | |
357 | return ret; | |
358 | } | |
359 | ||
360 | ff_init_ff_sine_windows(5); | |
361 | ||
362 | ff_atrac_generate_tables(); | |
363 | ||
f6fa7814 | 364 | q->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); |
2ba45a60 DM |
365 | |
366 | q->bands[0] = q->low; | |
367 | q->bands[1] = q->mid; | |
368 | q->bands[2] = q->high; | |
369 | ||
370 | /* Prepare the mdct overlap buffers */ | |
371 | q->SUs[0].spectrum[0] = q->SUs[0].spec1; | |
372 | q->SUs[0].spectrum[1] = q->SUs[0].spec2; | |
373 | q->SUs[1].spectrum[0] = q->SUs[1].spec1; | |
374 | q->SUs[1].spectrum[1] = q->SUs[1].spec2; | |
375 | ||
376 | return 0; | |
377 | } | |
378 | ||
379 | ||
380 | AVCodec ff_atrac1_decoder = { | |
381 | .name = "atrac1", | |
382 | .long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"), | |
383 | .type = AVMEDIA_TYPE_AUDIO, | |
384 | .id = AV_CODEC_ID_ATRAC1, | |
385 | .priv_data_size = sizeof(AT1Ctx), | |
386 | .init = atrac1_decode_init, | |
387 | .close = atrac1_decode_end, | |
388 | .decode = atrac1_decode_frame, | |
389 | .capabilities = CODEC_CAP_DR1, | |
390 | .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, | |
391 | AV_SAMPLE_FMT_NONE }, | |
392 | }; |