Mercurial > libavcodec.hg
annotate atrac1.c @ 10215:a43faa684a20 libavcodec
Move the scale factor and word length indexes to the stack.
author | banan |
---|---|
date | Mon, 21 Sep 2009 20:49:08 +0000 |
parents | 38ab367d4231 |
children | 5906eb8b992b |
rev | line source |
---|---|
10157 | 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. | |
26 * This decoder handles raw ATRAC1 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 | |
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 */ | |
57 typedef 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 | |
60 float* spectrum[2]; | |
10197 | 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 | |
10157 | 66 } AT1SUCtx; |
67 | |
68 /** | |
69 * The atrac1 context, holds all needed parameters for decoding | |
70 */ | |
71 typedef struct { | |
72 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit | |
10197 | 73 DECLARE_ALIGNED_16(float, spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer |
10185 | 74 |
10197 | 75 DECLARE_ALIGNED_16(float, low[256]); |
76 DECLARE_ALIGNED_16(float, mid[256]); | |
77 DECLARE_ALIGNED_16(float, high[512]); | |
10157 | 78 float* bands[3]; |
10197 | 79 DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]); |
10199 | 80 FFTContext mdct_ctx[3]; |
10157 | 81 int channels; |
82 DSPContext dsp; | |
83 } AT1Ctx; | |
84 | |
10185 | 85 DECLARE_ALIGNED_16(static float, short_window[32]); |
10157 | 86 |
87 /** size of the transform in samples in the long mode for each QMF band */ | |
88 static const uint16_t samples_per_band[3] = {128, 128, 256}; | |
89 static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; | |
90 | |
91 | |
10170 | 92 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, |
93 int rev_spec) | |
10157 | 94 { |
10199 | 95 FFTContext* mdct_context; |
10157 | 96 int transf_size = 1 << nbits; |
97 | |
10197 | 98 mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; |
10157 | 99 |
100 if (rev_spec) { | |
101 int i; | |
10197 | 102 for (i = 0; i < transf_size / 2; i++) |
10170 | 103 FFSWAP(float, spec[i], spec[transf_size - 1 - i]); |
10157 | 104 } |
10170 | 105 ff_imdct_half(mdct_context, out, spec); |
10157 | 106 } |
107 | |
108 | |
109 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) | |
110 { | |
10197 | 111 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; |
10198
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112 unsigned int start_pos, ref_pos = 0, pos = 0; |
10157 | 113 |
10197 | 114 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
10157 | 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 /* 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 | |
10197 | 129 if (nbits != 5 && nbits != 7 && nbits != 8) |
10157 | 130 return -1; |
131 | |
132 if (num_blocks == 1) { | |
10189 | 133 /* long blocks */ |
10157 | 134 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num); |
135 pos += block_size; // move to the next mdct block in the spectrum | |
10185 | 136 |
137 /* overlap and window long blocks */ | |
10197 | 138 q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos + band_samples - 16], |
139 &su->spectrum[0][ref_pos], short_window, 0, 16); | |
140 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); | |
10157 | 141 } else { |
10189 | 142 /* short blocks */ |
10185 | 143 float *prev_buf; |
10189 | 144 start_pos = 0; |
10197 | 145 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; |
146 for (; num_blocks != 0; num_blocks--) { | |
147 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], 5, band_num); | |
10157 | 148 |
149 /* overlap and window between short blocks */ | |
10189 | 150 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, |
10197 | 151 &su->spectrum[0][ref_pos + start_pos], short_window, 0, 16); |
10185 | 152 |
10197 | 153 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; |
10157 | 154 start_pos += 32; // use hardcoded block_size |
155 pos += 32; | |
156 } | |
157 } | |
158 ref_pos += band_samples; | |
159 } | |
160 | |
161 /* Swap buffers so the mdct overlap works */ | |
162 FFSWAP(float*, su->spectrum[0], su->spectrum[1]); | |
163 | |
164 return 0; | |
165 } | |
166 | |
10170 | 167 /** |
168 * Parse the block size mode byte | |
169 */ | |
10157 | 170 |
10170 | 171 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) |
10157 | 172 { |
173 int log2_block_count_tmp, i; | |
174 | |
10197 | 175 for (i = 0; i < 2; i++) { |
10157 | 176 /* low and mid band */ |
177 log2_block_count_tmp = get_bits(gb, 2); | |
178 if (log2_block_count_tmp & 1) | |
179 return -1; | |
10170 | 180 log2_block_cnt[i] = 2 - log2_block_count_tmp; |
10157 | 181 } |
182 | |
183 /* high band */ | |
184 log2_block_count_tmp = get_bits(gb, 2); | |
185 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) | |
186 return -1; | |
10170 | 187 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; |
10157 | 188 |
189 skip_bits(gb, 2); | |
190 return 0; | |
191 } | |
192 | |
193 | |
10170 | 194 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, |
195 float spec[AT1_SU_SAMPLES]) | |
10157 | 196 { |
197 int bits_used, band_num, bfu_num, i; | |
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198 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU |
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199 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU |
10157 | 200 |
201 /* parse the info byte (2nd byte) telling how much BFUs were coded */ | |
202 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 bits_used = su->num_bfus * 10 + 32 + | |
208 bfu_amount_tab2[get_bits(gb, 2)] + | |
209 (bfu_amount_tab3[get_bits(gb, 3)] << 1); | |
210 | |
211 /* get word length index (idwl) for each BFU */ | |
10197 | 212 for (i = 0; i < su->num_bfus; i++) |
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213 idwls[i] = get_bits(gb, 4); |
10157 | 214 |
215 /* get scalefactor index (idsf) for each BFU */ | |
10197 | 216 for (i = 0; i < su->num_bfus; i++) |
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217 idsfs[i] = get_bits(gb, 6); |
10157 | 218 |
219 /* zero idwl/idsf for empty BFUs */ | |
220 for (i = su->num_bfus; i < AT1_MAX_BFU; i++) | |
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221 idwls[i] = idsfs[i] = 0; |
10157 | 222 |
223 /* read in the spectral data and reconstruct MDCT spectrum of this channel */ | |
10197 | 224 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
225 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) { | |
10157 | 226 int pos; |
227 | |
228 int num_specs = specs_per_bfu[bfu_num]; | |
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229 int word_len = !!idwls[bfu_num] + idwls[bfu_num]; |
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230 float scale_factor = sf_table[idsfs[bfu_num]]; |
10157 | 231 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ |
232 | |
233 /* check for bitstream overflow */ | |
234 if (bits_used > AT1_SU_MAX_BITS) | |
235 return -1; | |
236 | |
237 /* get the position of the 1st spec according to the block size mode */ | |
238 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; | |
239 | |
240 if (word_len) { | |
10170 | 241 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); |
10157 | 242 |
10197 | 243 for (i = 0; i < num_specs; i++) { |
10157 | 244 /* read in a quantized spec and convert it to |
245 * signed int and then inverse quantization | |
246 */ | |
247 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 emty BFU */ | |
10197 | 250 memset(&spec[pos], 0, num_specs * sizeof(float)); |
10157 | 251 } |
252 } | |
253 } | |
254 | |
255 return 0; | |
256 } | |
257 | |
258 | |
259 void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) | |
260 { | |
10197 | 261 float temp[256]; |
262 float iqmf_temp[512 + 46]; | |
10157 | 263 |
264 /* combine low and middle bands */ | |
265 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 23 samples */ | |
10197 | 268 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23); |
269 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256); | |
10157 | 270 |
271 /* combine (low + middle) and high bands */ | |
272 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); | |
273 } | |
274 | |
275 | |
10170 | 276 static int atrac1_decode_frame(AVCodecContext *avctx, void *data, |
277 int *data_size, AVPacket *avpkt) | |
10157 | 278 { |
279 const uint8_t *buf = avpkt->data; | |
10170 | 280 int buf_size = avpkt->size; |
281 AT1Ctx *q = avctx->priv_data; | |
10157 | 282 int ch, ret, i; |
283 GetBitContext gb; | |
284 float* samples = data; | |
285 | |
286 | |
287 if (buf_size < 212 * q->channels) { | |
288 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n"); | |
289 return -1; | |
290 } | |
291 | |
10197 | 292 for (ch = 0; ch < q->channels; ch++) { |
10157 | 293 AT1SUCtx* su = &q->SUs[ch]; |
294 | |
10197 | 295 init_get_bits(&gb, &buf[212 * ch], 212 * 8); |
10157 | 296 |
297 /* parse block_size_mode, 1st byte */ | |
10170 | 298 ret = at1_parse_bsm(&gb, su->log2_block_count); |
10157 | 299 if (ret < 0) |
300 return ret; | |
301 | |
302 ret = at1_unpack_dequant(&gb, su, q->spec); | |
303 if (ret < 0) | |
304 return ret; | |
305 | |
306 ret = at1_imdct_block(su, q); | |
307 if (ret < 0) | |
308 return ret; | |
309 at1_subband_synthesis(q, su, q->out_samples[ch]); | |
310 } | |
311 | |
312 /* round, convert to 16bit and interleave */ | |
313 if (q->channels == 1) { | |
314 /* mono */ | |
10197 | 315 q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15), |
316 32700.0 / (1 << 15), AT1_SU_SAMPLES); | |
10157 | 317 } else { |
318 /* stereo */ | |
319 for (i = 0; i < AT1_SU_SAMPLES; i++) { | |
10197 | 320 samples[i * 2] = av_clipf(q->out_samples[0][i], |
321 -32700.0 / (1 << 15), | |
322 32700.0 / (1 << 15)); | |
323 samples[i * 2 + 1] = av_clipf(q->out_samples[1][i], | |
324 -32700.0 / (1 << 15), | |
325 32700.0 / (1 << 15)); | |
10157 | 326 } |
327 } | |
328 | |
329 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples); | |
330 return avctx->block_align; | |
331 } | |
332 | |
333 | |
334 static av_cold int atrac1_decode_init(AVCodecContext *avctx) | |
335 { | |
336 AT1Ctx *q = avctx->priv_data; | |
337 | |
338 avctx->sample_fmt = SAMPLE_FMT_FLT; | |
339 | |
340 q->channels = avctx->channels; | |
341 | |
342 /* Init the mdct transforms */ | |
10197 | 343 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15)); |
344 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15)); | |
345 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)); | |
10185 | 346 |
347 ff_sine_window_init(short_window, 32); | |
10157 | 348 |
349 atrac_generate_tables(); | |
350 | |
351 dsputil_init(&q->dsp, avctx); | |
352 | |
353 q->bands[0] = q->low; | |
354 q->bands[1] = q->mid; | |
355 q->bands[2] = q->high; | |
356 | |
357 /* Prepare the mdct overlap buffers */ | |
358 q->SUs[0].spectrum[0] = q->SUs[0].spec1; | |
359 q->SUs[0].spectrum[1] = q->SUs[0].spec2; | |
360 q->SUs[1].spectrum[0] = q->SUs[1].spec1; | |
361 q->SUs[1].spectrum[1] = q->SUs[1].spec2; | |
362 | |
363 return 0; | |
364 } | |
365 | |
366 AVCodec atrac1_decoder = { | |
367 .name = "atrac1", | |
368 .type = CODEC_TYPE_AUDIO, | |
369 .id = CODEC_ID_ATRAC1, | |
370 .priv_data_size = sizeof(AT1Ctx), | |
371 .init = atrac1_decode_init, | |
372 .close = NULL, | |
373 .decode = atrac1_decode_frame, | |
374 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), | |
375 }; |