b746a54e3b38f910dc5041aaeb085f927bc21462
[libav.git] / libavcodec / atrac1.c
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 Libav.
7 *
8 * Libav 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 * Libav 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 Libav; 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 #include "sinewin.h"
40
41 #include "atrac.h"
42 #include "atrac1data.h"
43
44 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
45 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
46 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
47 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
48 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
49 #define AT1_MAX_CHANNELS 2
50
51 #define AT1_QMF_BANDS 3
52 #define IDX_LOW_BAND 0
53 #define IDX_MID_BAND 1
54 #define IDX_HIGH_BAND 2
55
56 /**
57 * Sound unit struct, one unit is used per channel
58 */
59 typedef struct {
60 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
61 int num_bfus; ///< number of Block Floating Units
62 float* spectrum[2];
63 DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
64 DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
65 DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
66 DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
67 DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
68 } AT1SUCtx;
69
70 /**
71 * The atrac1 context, holds all needed parameters for decoding
72 */
73 typedef struct {
74 AVFrame frame;
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];
83 DSPContext dsp;
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 */
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, 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 emty 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 const uint8_t *buf = avpkt->data;
276 int buf_size = avpkt->size;
277 AT1Ctx *q = avctx->priv_data;
278 int ch, ret;
279 GetBitContext gb;
280
281
282 if (buf_size < 212 * avctx->channels) {
283 av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
284 return AVERROR_INVALIDDATA;
285 }
286
287 /* get output buffer */
288 q->frame.nb_samples = AT1_SU_SAMPLES;
289 if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
290 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
291 return ret;
292 }
293
294 for (ch = 0; ch < avctx->channels; ch++) {
295 AT1SUCtx* su = &q->SUs[ch];
296
297 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
298
299 /* parse block_size_mode, 1st byte */
300 ret = at1_parse_bsm(&gb, su->log2_block_count);
301 if (ret < 0)
302 return ret;
303
304 ret = at1_unpack_dequant(&gb, su, q->spec);
305 if (ret < 0)
306 return ret;
307
308 ret = at1_imdct_block(su, q);
309 if (ret < 0)
310 return ret;
311 at1_subband_synthesis(q, su, (float *)q->frame.extended_data[ch]);
312 }
313
314 *got_frame_ptr = 1;
315 *(AVFrame *)data = q->frame;
316
317 return avctx->block_align;
318 }
319
320
321 static av_cold int atrac1_decode_end(AVCodecContext * avctx)
322 {
323 AT1Ctx *q = avctx->priv_data;
324
325 ff_mdct_end(&q->mdct_ctx[0]);
326 ff_mdct_end(&q->mdct_ctx[1]);
327 ff_mdct_end(&q->mdct_ctx[2]);
328
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 /* Init the mdct transforms */
347 if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
348 (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
349 (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
350 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
351 atrac1_decode_end(avctx);
352 return ret;
353 }
354
355 ff_init_ff_sine_windows(5);
356
357 ff_atrac_generate_tables();
358
359 ff_dsputil_init(&q->dsp, avctx);
360
361 q->bands[0] = q->low;
362 q->bands[1] = q->mid;
363 q->bands[2] = q->high;
364
365 /* Prepare the mdct overlap buffers */
366 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
367 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
368 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
369 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
370
371 avcodec_get_frame_defaults(&q->frame);
372 avctx->coded_frame = &q->frame;
373
374 return 0;
375 }
376
377
378 AVCodec ff_atrac1_decoder = {
379 .name = "atrac1",
380 .type = AVMEDIA_TYPE_AUDIO,
381 .id = AV_CODEC_ID_ATRAC1,
382 .priv_data_size = sizeof(AT1Ctx),
383 .init = atrac1_decode_init,
384 .close = atrac1_decode_end,
385 .decode = atrac1_decode_frame,
386 .capabilities = CODEC_CAP_DR1,
387 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
388 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
389 AV_SAMPLE_FMT_NONE },
390 };