FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/atrac1.c
Date: 2022-11-26 13:19:19
Exec Total Coverage
Lines: 131 147 89.1%
Branches: 54 70 77.1%

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