Mercurial > libavcodec.hg
annotate atrac1.c @ 12409:91db982aaaad libavcodec
fixed some return values and deprecated CODEC_TYPE_VIDEO.
dithering (faster) along a linear gradient now.
author | bindhammer |
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date | Tue, 24 Aug 2010 14:02:31 +0000 |
parents | 8b6f3d3b55cb |
children |
rev | line source |
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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 /** | |
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24 * @file |
10157 | 25 * Atrac 1 compatible decoder. |
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26 * This decoder handles raw ATRAC1 data and probably SDDS data. |
10157 | 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" | |
11370 | 38 #include "fft.h" |
10157 | 39 |
40 #include "atrac.h" | |
41 #include "atrac1data.h" | |
42 | |
43 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit | |
44 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit | |
45 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit | |
46 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 | |
47 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 | |
48 #define AT1_MAX_CHANNELS 2 | |
49 | |
50 #define AT1_QMF_BANDS 3 | |
51 #define IDX_LOW_BAND 0 | |
52 #define IDX_MID_BAND 1 | |
53 #define IDX_HIGH_BAND 2 | |
54 | |
55 /** | |
56 * Sound unit struct, one unit is used per channel | |
57 */ | |
58 typedef struct { | |
59 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band | |
60 int num_bfus; ///< number of Block Floating Units | |
61 float* spectrum[2]; | |
11369 | 62 DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer |
63 DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer | |
64 DECLARE_ALIGNED(16, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter | |
65 DECLARE_ALIGNED(16, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter | |
66 DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter | |
10157 | 67 } AT1SUCtx; |
68 | |
69 /** | |
70 * The atrac1 context, holds all needed parameters for decoding | |
71 */ | |
72 typedef struct { | |
73 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit | |
11369 | 74 DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer |
10185 | 75 |
11369 | 76 DECLARE_ALIGNED(16, float, low)[256]; |
77 DECLARE_ALIGNED(16, float, mid)[256]; | |
78 DECLARE_ALIGNED(16, float, high)[512]; | |
10157 | 79 float* bands[3]; |
11369 | 80 DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; |
10199 | 81 FFTContext mdct_ctx[3]; |
10157 | 82 int channels; |
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 | |
10170 | 91 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, |
92 int rev_spec) | |
10157 | 93 { |
10216 | 94 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; |
10157 | 95 int transf_size = 1 << nbits; |
96 | |
97 if (rev_spec) { | |
98 int i; | |
10197 | 99 for (i = 0; i < transf_size / 2; i++) |
10170 | 100 FFSWAP(float, spec[i], spec[transf_size - 1 - i]); |
10157 | 101 } |
10170 | 102 ff_imdct_half(mdct_context, out, spec); |
10157 | 103 } |
104 | |
105 | |
106 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) | |
107 { | |
10197 | 108 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; |
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109 unsigned int start_pos, ref_pos = 0, pos = 0; |
10157 | 110 |
10197 | 111 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
10266 | 112 float *prev_buf; |
113 int j; | |
114 | |
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 | |
10266 | 122 if (num_blocks == 1) { |
10267 | 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; | |
10157 | 126 |
10267 | 127 /* calc transform size in bits according to the block_size_mode */ |
128 nbits = mdct_long_nbits[band_num] - log2_block_count; | |
10157 | 129 |
10267 | 130 if (nbits != 5 && nbits != 7 && nbits != 8) |
131 return -1; | |
10266 | 132 } else { |
133 block_size = 32; | |
134 nbits = 5; | |
135 } | |
10185 | 136 |
10267 | 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); | |
10157 | 141 |
10267 | 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, 0, 16); | |
10185 | 145 |
10267 | 146 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; |
147 start_pos += block_size; | |
148 pos += block_size; | |
149 } | |
10266 | 150 |
151 if (num_blocks == 1) | |
152 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); | |
153 | |
10157 | 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 | |
10170 | 163 /** |
164 * Parse the block size mode byte | |
165 */ | |
10157 | 166 |
10170 | 167 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) |
10157 | 168 { |
169 int log2_block_count_tmp, i; | |
170 | |
10197 | 171 for (i = 0; i < 2; i++) { |
10157 | 172 /* low and mid band */ |
173 log2_block_count_tmp = get_bits(gb, 2); | |
174 if (log2_block_count_tmp & 1) | |
175 return -1; | |
10170 | 176 log2_block_cnt[i] = 2 - log2_block_count_tmp; |
10157 | 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 -1; | |
10170 | 183 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; |
10157 | 184 |
185 skip_bits(gb, 2); | |
186 return 0; | |
187 } | |
188 | |
189 | |
10170 | 190 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, |
191 float spec[AT1_SU_SAMPLES]) | |
10157 | 192 { |
193 int bits_used, band_num, bfu_num, i; | |
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194 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU |
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195 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU |
10157 | 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 */ | |
10197 | 208 for (i = 0; i < su->num_bfus; i++) |
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209 idwls[i] = get_bits(gb, 4); |
10157 | 210 |
211 /* get scalefactor index (idsf) for each BFU */ | |
10197 | 212 for (i = 0; i < su->num_bfus; i++) |
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213 idsfs[i] = get_bits(gb, 6); |
10157 | 214 |
215 /* zero idwl/idsf for empty BFUs */ | |
216 for (i = su->num_bfus; i < AT1_MAX_BFU; i++) | |
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217 idwls[i] = idsfs[i] = 0; |
10157 | 218 |
219 /* read in the spectral data and reconstruct MDCT spectrum of this channel */ | |
10197 | 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++) { | |
10157 | 222 int pos; |
223 | |
224 int num_specs = specs_per_bfu[bfu_num]; | |
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225 int word_len = !!idwls[bfu_num] + idwls[bfu_num]; |
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226 float scale_factor = sf_table[idsfs[bfu_num]]; |
10217 | 227 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ |
10157 | 228 |
229 /* check for bitstream overflow */ | |
230 if (bits_used > AT1_SU_MAX_BITS) | |
231 return -1; | |
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) { | |
10170 | 237 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); |
10157 | 238 |
10197 | 239 for (i = 0; i < num_specs; i++) { |
10157 | 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 */ | |
10197 | 246 memset(&spec[pos], 0, num_specs * sizeof(float)); |
10157 | 247 } |
248 } | |
249 } | |
250 | |
251 return 0; | |
252 } | |
253 | |
254 | |
11374 | 255 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) |
10157 | 256 { |
10197 | 257 float temp[256]; |
258 float iqmf_temp[512 + 46]; | |
10157 | 259 |
260 /* combine low and middle bands */ | |
261 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 */ | |
10197 | 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); | |
10157 | 266 |
267 /* combine (low + middle) and high bands */ | |
268 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); | |
269 } | |
270 | |
271 | |
10170 | 272 static int atrac1_decode_frame(AVCodecContext *avctx, void *data, |
273 int *data_size, AVPacket *avpkt) | |
10157 | 274 { |
275 const uint8_t *buf = avpkt->data; | |
10170 | 276 int buf_size = avpkt->size; |
277 AT1Ctx *q = avctx->priv_data; | |
10157 | 278 int ch, ret, i; |
279 GetBitContext gb; | |
280 float* samples = data; | |
281 | |
282 | |
283 if (buf_size < 212 * q->channels) { | |
284 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n"); | |
285 return -1; | |
286 } | |
287 | |
10197 | 288 for (ch = 0; ch < q->channels; ch++) { |
10157 | 289 AT1SUCtx* su = &q->SUs[ch]; |
290 | |
10197 | 291 init_get_bits(&gb, &buf[212 * ch], 212 * 8); |
10157 | 292 |
293 /* parse block_size_mode, 1st byte */ | |
10170 | 294 ret = at1_parse_bsm(&gb, su->log2_block_count); |
10157 | 295 if (ret < 0) |
296 return ret; | |
297 | |
298 ret = at1_unpack_dequant(&gb, su, q->spec); | |
299 if (ret < 0) | |
300 return ret; | |
301 | |
302 ret = at1_imdct_block(su, q); | |
303 if (ret < 0) | |
304 return ret; | |
305 at1_subband_synthesis(q, su, q->out_samples[ch]); | |
306 } | |
307 | |
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308 /* interleave; FIXME, should create/use a DSP function */ |
10157 | 309 if (q->channels == 1) { |
310 /* mono */ | |
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311 memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4); |
10157 | 312 } else { |
313 /* stereo */ | |
314 for (i = 0; i < AT1_SU_SAMPLES; i++) { | |
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315 samples[i * 2] = q->out_samples[0][i]; |
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316 samples[i * 2 + 1] = q->out_samples[1][i]; |
10157 | 317 } |
318 } | |
319 | |
320 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples); | |
321 return avctx->block_align; | |
322 } | |
323 | |
324 | |
325 static av_cold int atrac1_decode_init(AVCodecContext *avctx) | |
326 { | |
327 AT1Ctx *q = avctx->priv_data; | |
328 | |
329 avctx->sample_fmt = SAMPLE_FMT_FLT; | |
330 | |
331 q->channels = avctx->channels; | |
332 | |
333 /* Init the mdct transforms */ | |
10197 | 334 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15)); |
335 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15)); | |
336 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)); | |
10185 | 337 |
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338 ff_init_ff_sine_windows(5); |
10157 | 339 |
340 atrac_generate_tables(); | |
341 | |
342 dsputil_init(&q->dsp, avctx); | |
343 | |
344 q->bands[0] = q->low; | |
345 q->bands[1] = q->mid; | |
346 q->bands[2] = q->high; | |
347 | |
348 /* Prepare the mdct overlap buffers */ | |
349 q->SUs[0].spectrum[0] = q->SUs[0].spec1; | |
350 q->SUs[0].spectrum[1] = q->SUs[0].spec2; | |
351 q->SUs[1].spectrum[0] = q->SUs[1].spec1; | |
352 q->SUs[1].spectrum[1] = q->SUs[1].spec2; | |
353 | |
354 return 0; | |
355 } | |
356 | |
10218 | 357 |
358 static av_cold int atrac1_decode_end(AVCodecContext * avctx) { | |
359 AT1Ctx *q = avctx->priv_data; | |
360 | |
361 ff_mdct_end(&q->mdct_ctx[0]); | |
362 ff_mdct_end(&q->mdct_ctx[1]); | |
363 ff_mdct_end(&q->mdct_ctx[2]); | |
364 return 0; | |
365 } | |
366 | |
367 | |
10157 | 368 AVCodec atrac1_decoder = { |
369 .name = "atrac1", | |
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370 .type = AVMEDIA_TYPE_AUDIO, |
10157 | 371 .id = CODEC_ID_ATRAC1, |
372 .priv_data_size = sizeof(AT1Ctx), | |
373 .init = atrac1_decode_init, | |
10218 | 374 .close = atrac1_decode_end, |
10157 | 375 .decode = atrac1_decode_frame, |
376 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), | |
377 }; |