Factorize duplicated code in at1_imdct_block()
[libav.git] / libavcodec / atrac1.c
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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 libavcodec/atrac1.c
25 * Atrac 1 compatible decoder.
e704b012 26 * This decoder handles raw ATRAC1 data and probably SDDS data.
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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
39#include "atrac.h"
40#include "atrac1data.h"
41
42#define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
43#define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
44#define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
45#define AT1_FRAME_SIZE AT1_SU_SIZE * 2
46#define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
47#define AT1_MAX_CHANNELS 2
48
49#define AT1_QMF_BANDS 3
50#define IDX_LOW_BAND 0
51#define IDX_MID_BAND 1
52#define IDX_HIGH_BAND 2
53
54/**
55 * Sound unit struct, one unit is used per channel
56 */
57typedef struct {
58 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
59 int num_bfus; ///< number of Block Floating Units
dbb0f96f 60 float* spectrum[2];
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61 DECLARE_ALIGNED_16(float, spec1[AT1_SU_SAMPLES]); ///< mdct buffer
62 DECLARE_ALIGNED_16(float, spec2[AT1_SU_SAMPLES]); ///< mdct buffer
63 DECLARE_ALIGNED_16(float, fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter
64 DECLARE_ALIGNED_16(float, snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter
65 DECLARE_ALIGNED_16(float, last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter
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66} AT1SUCtx;
67
68/**
69 * The atrac1 context, holds all needed parameters for decoding
70 */
71typedef struct {
72 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
1e1898c0 73 DECLARE_ALIGNED_16(float, spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
10634c03 74
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75 DECLARE_ALIGNED_16(float, low[256]);
76 DECLARE_ALIGNED_16(float, mid[256]);
77 DECLARE_ALIGNED_16(float, high[512]);
dbb0f96f 78 float* bands[3];
1e1898c0 79 DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
01b22147 80 FFTContext mdct_ctx[3];
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81 int channels;
82 DSPContext dsp;
83} AT1Ctx;
84
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85/** size of the transform in samples in the long mode for each QMF band */
86static const uint16_t samples_per_band[3] = {128, 128, 256};
87static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
88
89
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90static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
91 int rev_spec)
dbb0f96f 92{
a872e5c1 93 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
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94 int transf_size = 1 << nbits;
95
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96 if (rev_spec) {
97 int i;
1e1898c0 98 for (i = 0; i < transf_size / 2; i++)
04a6d1b0 99 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
dbb0f96f 100 }
04a6d1b0 101 ff_imdct_half(mdct_context, out, spec);
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102}
103
104
105static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
106{
1e1898c0 107 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
ec129499 108 unsigned int start_pos, ref_pos = 0, pos = 0;
dbb0f96f 109
1e1898c0 110 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
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111 float *prev_buf;
112 int j;
113
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114 band_samples = samples_per_band[band_num];
115 log2_block_count = su->log2_block_count[band_num];
116
117 /* number of mdct blocks in the current QMF band: 1 - for long mode */
118 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
119 num_blocks = 1 << log2_block_count;
120
b11d40d1 121 if (num_blocks == 1) {
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122 /* mdct block size in samples: 128 (long mode, low & mid bands), */
123 /* 256 (long mode, high band) and 32 (short mode, all bands) */
124 block_size = band_samples >> log2_block_count;
125
126 /* calc transform size in bits according to the block_size_mode */
127 nbits = mdct_long_nbits[band_num] - log2_block_count;
128
1e1898c0 129 if (nbits != 5 && nbits != 7 && nbits != 8)
dbb0f96f 130 return -1;
dbb0f96f 131 } else {
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132 block_size = 32;
133 nbits = 5;
134 }
135
0105f497 136 start_pos = 0;
1e1898c0 137 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
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138 for (j=0; j < num_blocks; j++) {
139 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
dbb0f96f 140
b11d40d1 141 /* overlap and window */
0105f497 142 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
edd897b1 143 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
10634c03 144
1e1898c0 145 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
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146 start_pos += block_size;
147 pos += block_size;
dbb0f96f 148 }
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149
150 if (num_blocks == 1)
151 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
152
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153 ref_pos += band_samples;
154 }
155
156 /* Swap buffers so the mdct overlap works */
157 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
158
159 return 0;
160}
161
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162/**
163 * Parse the block size mode byte
164 */
dbb0f96f 165
04a6d1b0 166static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
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167{
168 int log2_block_count_tmp, i;
169
1e1898c0 170 for (i = 0; i < 2; i++) {
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171 /* low and mid band */
172 log2_block_count_tmp = get_bits(gb, 2);
173 if (log2_block_count_tmp & 1)
174 return -1;
04a6d1b0 175 log2_block_cnt[i] = 2 - log2_block_count_tmp;
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176 }
177
178 /* high band */
179 log2_block_count_tmp = get_bits(gb, 2);
180 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
181 return -1;
04a6d1b0 182 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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183
184 skip_bits(gb, 2);
185 return 0;
186}
187
188
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189static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
190 float spec[AT1_SU_SAMPLES])
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191{
192 int bits_used, band_num, bfu_num, i;
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193 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
194 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
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195
196 /* parse the info byte (2nd byte) telling how much BFUs were coded */
197 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
198
199 /* calc number of consumed bits:
200 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
201 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
202 bits_used = su->num_bfus * 10 + 32 +
203 bfu_amount_tab2[get_bits(gb, 2)] +
204 (bfu_amount_tab3[get_bits(gb, 3)] << 1);
205
206 /* get word length index (idwl) for each BFU */
1e1898c0 207 for (i = 0; i < su->num_bfus; i++)
b6a23702 208 idwls[i] = get_bits(gb, 4);
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209
210 /* get scalefactor index (idsf) for each BFU */
1e1898c0 211 for (i = 0; i < su->num_bfus; i++)
b6a23702 212 idsfs[i] = get_bits(gb, 6);
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213
214 /* zero idwl/idsf for empty BFUs */
215 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
b6a23702 216 idwls[i] = idsfs[i] = 0;
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217
218 /* read in the spectral data and reconstruct MDCT spectrum of this channel */
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219 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
220 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
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221 int pos;
222
223 int num_specs = specs_per_bfu[bfu_num];
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224 int word_len = !!idwls[bfu_num] + idwls[bfu_num];
225 float scale_factor = sf_table[idsfs[bfu_num]];
78b3a12d 226 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
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227
228 /* check for bitstream overflow */
229 if (bits_used > AT1_SU_MAX_BITS)
230 return -1;
231
232 /* get the position of the 1st spec according to the block size mode */
233 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
234
235 if (word_len) {
04a6d1b0 236 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
dbb0f96f 237
1e1898c0 238 for (i = 0; i < num_specs; i++) {
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239 /* read in a quantized spec and convert it to
240 * signed int and then inverse quantization
241 */
242 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
243 }
244 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
1e1898c0 245 memset(&spec[pos], 0, num_specs * sizeof(float));
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246 }
247 }
248 }
249
250 return 0;
251}
252
253
254void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
255{
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256 float temp[256];
257 float iqmf_temp[512 + 46];
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258
259 /* combine low and middle bands */
260 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
261
262 /* delay the signal of the high band by 23 samples */
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263 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
264 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
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265
266 /* combine (low + middle) and high bands */
267 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
268}
269
270
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271static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
272 int *data_size, AVPacket *avpkt)
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273{
274 const uint8_t *buf = avpkt->data;
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275 int buf_size = avpkt->size;
276 AT1Ctx *q = avctx->priv_data;
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277 int ch, ret, i;
278 GetBitContext gb;
279 float* samples = data;
280
281
282 if (buf_size < 212 * q->channels) {
283 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
284 return -1;
285 }
286
1e1898c0 287 for (ch = 0; ch < q->channels; ch++) {
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288 AT1SUCtx* su = &q->SUs[ch];
289
1e1898c0 290 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
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291
292 /* parse block_size_mode, 1st byte */
04a6d1b0 293 ret = at1_parse_bsm(&gb, su->log2_block_count);
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294 if (ret < 0)
295 return ret;
296
297 ret = at1_unpack_dequant(&gb, su, q->spec);
298 if (ret < 0)
299 return ret;
300
301 ret = at1_imdct_block(su, q);
302 if (ret < 0)
303 return ret;
304 at1_subband_synthesis(q, su, q->out_samples[ch]);
305 }
306
307 /* round, convert to 16bit and interleave */
308 if (q->channels == 1) {
309 /* mono */
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310 q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
311 32700.0 / (1 << 15), AT1_SU_SAMPLES);
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312 } else {
313 /* stereo */
314 for (i = 0; i < AT1_SU_SAMPLES; i++) {
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315 samples[i * 2] = av_clipf(q->out_samples[0][i],
316 -32700.0 / (1 << 15),
317 32700.0 / (1 << 15));
318 samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
319 -32700.0 / (1 << 15),
320 32700.0 / (1 << 15));
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321 }
322 }
323
324 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
325 return avctx->block_align;
326}
327
328
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329static av_cold int atrac1_decode_init(AVCodecContext *avctx)
330{
331 AT1Ctx *q = avctx->priv_data;
332
333 avctx->sample_fmt = SAMPLE_FMT_FLT;
334
335 q->channels = avctx->channels;
336
337 /* Init the mdct transforms */
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338 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
339 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
340 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
10634c03 341
edd897b1 342 ff_sine_window_init(ff_sine_32, 32);
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343
344 atrac_generate_tables();
345
346 dsputil_init(&q->dsp, avctx);
347
348 q->bands[0] = q->low;
349 q->bands[1] = q->mid;
350 q->bands[2] = q->high;
351
352 /* Prepare the mdct overlap buffers */
353 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
354 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
355 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
356 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
357
358 return 0;
359}
360
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361
362static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
363 AT1Ctx *q = avctx->priv_data;
364
365 ff_mdct_end(&q->mdct_ctx[0]);
366 ff_mdct_end(&q->mdct_ctx[1]);
367 ff_mdct_end(&q->mdct_ctx[2]);
368 return 0;
369}
370
371
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372AVCodec atrac1_decoder = {
373 .name = "atrac1",
374 .type = CODEC_TYPE_AUDIO,
375 .id = CODEC_ID_ATRAC1,
376 .priv_data_size = sizeof(AT1Ctx),
377 .init = atrac1_decode_init,
9caab878 378 .close = atrac1_decode_end,
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379 .decode = atrac1_decode_frame,
380 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
381};