FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavcodec/atrac1.c
Date: 2026-01-14 03:33:33
Exec Total Coverage
Lines: 130 146 89.0%
Functions: 8 8 100.0%
Branches: 56 72 77.8%
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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
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/mem.h"
35 #include "libavutil/mem_internal.h"
36 #include "libavutil/tx.h"
37
38 #include "avcodec.h"
39 #include "codec_internal.h"
40 #include "decode.h"
41 #include "get_bits.h"
42 #include "sinewin.h"
43
44 #include "atrac.h"
45 #include "atrac1data.h"
46
47 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
48 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
49 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
50 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
51 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
52 #define AT1_MAX_CHANNELS 8
53
54 #define AT1_QMF_BANDS 3
55 #define IDX_LOW_BAND 0
56 #define IDX_MID_BAND 1
57 #define IDX_HIGH_BAND 2
58
59 /**
60 * Sound unit struct, one unit is used per channel
61 */
62 typedef struct AT1SUCtx {
63 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
64 int num_bfus; ///< number of Block Floating Units
65 float* spectrum[2];
66 DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
67 DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
68 DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
69 DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
70 DECLARE_ALIGNED(32, float, last_qmf_delay)[256+39]; ///< delay line for the last stacked QMF filter
71 } AT1SUCtx;
72
73 /**
74 * The atrac1 context, holds all needed parameters for decoding
75 */
76 typedef struct AT1Ctx {
77 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
78 DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
79
80 DECLARE_ALIGNED(32, float, low)[256];
81 DECLARE_ALIGNED(32, float, mid)[256];
82 DECLARE_ALIGNED(32, float, high)[512];
83 float* bands[3];
84 AVTXContext *mdct_ctx[3];
85 av_tx_fn mdct_fn[3];
86 void (*vector_fmul_window)(float *dst, const float *src0,
87 const float *src1, const float *win, int len);
88 } AT1Ctx;
89
90 /** size of the transform in samples in the long mode for each QMF band */
91 static const uint16_t samples_per_band[3] = {128, 128, 256};
92 static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
93
94
95 7157 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
96 int rev_spec)
97 {
98 7157 AVTXContext *mdct_context = q->mdct_ctx[nbits - 5 - (nbits > 6)];
99 7157 av_tx_fn mdct_fn = q->mdct_fn[nbits - 5 - (nbits > 6)];
100 7157 int transf_size = 1 << nbits;
101
102
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 7157 if (rev_spec) {
103 int i;
104
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 459048 for (i = 0; i < transf_size / 2; i++)
105 454272 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
106 }
107 7157 mdct_fn(mdct_context, out, spec, sizeof(float));
108 7157 }
109
110
111 2366 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
112 {
113 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
114 2366 unsigned int start_pos, ref_pos = 0, pos = 0;
115
116
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 9464 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
117 float *prev_buf;
118 int j;
119
120 7098 band_samples = samples_per_band[band_num];
121 7098 log2_block_count = su->log2_block_count[band_num];
122
123 /* number of mdct blocks in the current QMF band: 1 - for long mode */
124 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
125 7098 num_blocks = 1 << log2_block_count;
126
127
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 7098 if (num_blocks == 1) {
128 /* mdct block size in samples: 128 (long mode, low & mid bands), */
129 /* 256 (long mode, high band) and 32 (short mode, all bands) */
130 7085 block_size = band_samples >> log2_block_count;
131
132 /* calc transform size in bits according to the block_size_mode */
133 7085 nbits = mdct_long_nbits[band_num] - log2_block_count;
134
135
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 7085 if (nbits != 5 && nbits != 7 && nbits != 8)
136 ✗ return AVERROR_INVALIDDATA;
137 } else {
138 13 block_size = 32;
139 13 nbits = 5;
140 }
141
142 7098 start_pos = 0;
143 7098 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
144
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 14255 for (j=0; j < num_blocks; j++) {
145 7157 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
146
147 /* overlap and window */
148 7157 q->vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
149 7157 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
150
151 7157 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
152 7157 start_pos += block_size;
153 7157 pos += block_size;
154 }
155
156
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 7098 if (num_blocks == 1)
157 7085 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
158
159 7098 ref_pos += band_samples;
160 }
161
162 /* Swap buffers so the mdct overlap works */
163 2366 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
164
165 2366 return 0;
166 }
167
168 /**
169 * Parse the block size mode byte
170 */
171
172 2366 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
173 {
174 int log2_block_count_tmp, i;
175
176
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 7098 for (i = 0; i < 2; i++) {
177 /* low and mid band */
178 4732 log2_block_count_tmp = get_bits(gb, 2);
179
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 4732 if (log2_block_count_tmp & 1)
180 ✗ return AVERROR_INVALIDDATA;
181 4732 log2_block_cnt[i] = 2 - log2_block_count_tmp;
182 }
183
184 /* high band */
185 2366 log2_block_count_tmp = get_bits(gb, 2);
186
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 2366 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
187 ✗ return AVERROR_INVALIDDATA;
188 2366 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
189
190 2366 skip_bits(gb, 2);
191 2366 return 0;
192 }
193
194
195 2366 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
196 float spec[AT1_SU_SAMPLES])
197 {
198 int bits_used, band_num, bfu_num, i;
199 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
200 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
201
202 /* parse the info byte (2nd byte) telling how much BFUs were coded */
203 2366 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
204
205 /* calc number of consumed bits:
206 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
207 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
208 7098 bits_used = su->num_bfus * 10 + 32 +
209 2366 bfu_amount_tab2[get_bits(gb, 2)] +
210 2366 (bfu_amount_tab3[get_bits(gb, 3)] << 1);
211
212 /* get word length index (idwl) for each BFU */
213
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 107674 for (i = 0; i < su->num_bfus; i++)
214 105308 idwls[i] = get_bits(gb, 4);
215
216 /* get scalefactor index (idsf) for each BFU */
217
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 107674 for (i = 0; i < su->num_bfus; i++)
218 105308 idsfs[i] = get_bits(gb, 6);
219
220 /* zero idwl/idsf for empty BFUs */
221
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 20090 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
222 17724 idwls[i] = idsfs[i] = 0;
223
224 /* read in the spectral data and reconstruct MDCT spectrum of this channel */
225
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 9464 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
226
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 130130 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
227 int pos;
228
229 123032 int num_specs = specs_per_bfu[bfu_num];
230 123032 int word_len = !!idwls[bfu_num] + idwls[bfu_num];
231 123032 float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]];
232 123032 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
233
234 /* check for bitstream overflow */
235
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 123032 if (bits_used > AT1_SU_MAX_BITS)
236 ✗ return AVERROR_INVALIDDATA;
237
238 /* get the position of the 1st spec according to the block size mode */
239
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 123032 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
240
241
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 123032 if (word_len) {
242 57252 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
243
244
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 555280 for (i = 0; i < num_specs; i++) {
245 /* read in a quantized spec and convert it to
246 * signed int and then inverse quantization
247 */
248 498028 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
249 }
250 } else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */
251 65780 memset(&spec[pos], 0, num_specs * sizeof(float));
252 }
253 }
254 }
255
256 2366 return 0;
257 }
258
259
260 2366 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
261 {
262 float temp[256];
263 float iqmf_temp[512 + 46];
264
265 /* combine low and middle bands */
266 2366 ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
267
268 /* delay the signal of the high band by 39 samples */
269 2366 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 39);
270 2366 memcpy(&su->last_qmf_delay[39], q->bands[2], sizeof(float) * 256);
271
272 /* combine (low + middle) and high bands */
273 2366 ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
274 2366 }
275
276
277 1184 static int atrac1_decode_frame(AVCodecContext *avctx, AVFrame *frame,
278 int *got_frame_ptr, AVPacket *avpkt)
279 {
280 1184 const uint8_t *buf = avpkt->data;
281 1184 int buf_size = avpkt->size;
282 1184 AT1Ctx *q = avctx->priv_data;
283 1184 int channels = avctx->ch_layout.nb_channels;
284 int ch, ret;
285 GetBitContext gb;
286
287
288
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 1184 if (buf_size < 212 * channels) {
289 1 av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
290 1 return AVERROR_INVALIDDATA;
291 }
292
293 /* get output buffer */
294 1183 frame->nb_samples = AT1_SU_SAMPLES;
295
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 1183 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
296 ✗ return ret;
297
298
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 3549 for (ch = 0; ch < channels; ch++) {
299 2366 AT1SUCtx* su = &q->SUs[ch];
300
301 2366 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
302
303 /* parse block_size_mode, 1st byte */
304 2366 ret = at1_parse_bsm(&gb, su->log2_block_count);
305
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 2366 if (ret < 0)
306 ✗ return ret;
307
308 2366 ret = at1_unpack_dequant(&gb, su, q->spec);
309
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 2366 if (ret < 0)
310 ✗ return ret;
311
312 2366 ret = at1_imdct_block(su, q);
313
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 2366 if (ret < 0)
314 ✗ return ret;
315 2366 at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]);
316 }
317
318 1183 *got_frame_ptr = 1;
319
320 1183 return avctx->block_align;
321 }
322
323
324 9 static av_cold int atrac1_decode_end(AVCodecContext * avctx)
325 {
326 9 AT1Ctx *q = avctx->priv_data;
327
328 9 av_tx_uninit(&q->mdct_ctx[0]);
329 9 av_tx_uninit(&q->mdct_ctx[1]);
330 9 av_tx_uninit(&q->mdct_ctx[2]);
331
332 9 return 0;
333 }
334
335
336 9 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
337 {
338 9 AT1Ctx *q = avctx->priv_data;
339 AVFloatDSPContext *fdsp;
340 9 int channels = avctx->ch_layout.nb_channels;
341 9 float scale = -1.0 / (1 << 15);
342 int ret, ch;
343
344 9 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
345
346
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 9 if (channels < 1 || channels > AT1_MAX_CHANNELS) {
347 ✗ av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
348 channels);
349 ✗ return AVERROR(EINVAL);
350 }
351
352
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 9 if (avctx->block_align <= 0) {
353 ✗ av_log(avctx, AV_LOG_ERROR, "Unsupported block align.");
354 ✗ return AVERROR_PATCHWELCOME;
355 }
356
357 /* Init the mdct transforms */
358 9 if ((ret = av_tx_init(&q->mdct_ctx[0], &q->mdct_fn[0], AV_TX_FLOAT_MDCT,
359
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 9 1, 32, &scale, 0) < 0))
360 ✗ return ret;
361 9 if ((ret = av_tx_init(&q->mdct_ctx[1], &q->mdct_fn[1], AV_TX_FLOAT_MDCT,
362
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 9 1, 128, &scale, 0) < 0))
363 ✗ return ret;
364 9 if ((ret = av_tx_init(&q->mdct_ctx[2], &q->mdct_fn[2], AV_TX_FLOAT_MDCT,
365
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 9 1, 256, &scale, 0) < 0))
366 ✗ return ret;
367
368 9 ff_init_ff_sine_windows(5);
369
370 9 ff_atrac_generate_tables();
371
372 9 fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
373
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 9 if (!fdsp)
374 ✗ return AVERROR(ENOMEM);
375 9 q->vector_fmul_window = fdsp->vector_fmul_window;
376 9 av_free(fdsp);
377
378 9 q->bands[0] = q->low;
379 9 q->bands[1] = q->mid;
380 9 q->bands[2] = q->high;
381
382 /* Prepare the mdct overlap buffers */
383
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 81 for (ch = 0; ch < AT1_MAX_CHANNELS; ch++) {
384 72 q->SUs[ch].spectrum[0] = q->SUs[ch].spec1;
385 72 q->SUs[ch].spectrum[1] = q->SUs[ch].spec2;
386 }
387
388 9 return 0;
389 }
390
391
392 const FFCodec ff_atrac1_decoder = {
393 .p.name = "atrac1",
394 CODEC_LONG_NAME("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
395 .p.type = AVMEDIA_TYPE_AUDIO,
396 .p.id = AV_CODEC_ID_ATRAC1,
397 .priv_data_size = sizeof(AT1Ctx),
398 .init = atrac1_decode_init,
399 .close = atrac1_decode_end,
400 FF_CODEC_DECODE_CB(atrac1_decode_frame),
401 .p.capabilities = AV_CODEC_CAP_DR1,
402 CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP),
403 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
404 };
405