lavc: add a wrapper for AVCodecContext.get_buffer().
[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 "internal.h"
40 #include "sinewin.h"
41
42 #include "atrac.h"
43 #include "atrac1data.h"
44
45 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
46 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
47 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
48 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
49 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
50 #define AT1_MAX_CHANNELS 2
51
52 #define AT1_QMF_BANDS 3
53 #define IDX_LOW_BAND 0
54 #define IDX_MID_BAND 1
55 #define IDX_HIGH_BAND 2
56
57 /**
58 * Sound unit struct, one unit is used per channel
59 */
60 typedef struct {
61 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
62 int num_bfus; ///< number of Block Floating Units
63 float* spectrum[2];
64 DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
65 DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
66 DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
67 DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
68 DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
69 } AT1SUCtx;
70
71 /**
72 * The atrac1 context, holds all needed parameters for decoding
73 */
74 typedef struct {
75 AVFrame frame;
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 FFTContext mdct_ctx[3];
84 DSPContext dsp;
85 } AT1Ctx;
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
92 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
93 int rev_spec)
94 {
95 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
96 int transf_size = 1 << nbits;
97
98 if (rev_spec) {
99 int i;
100 for (i = 0; i < transf_size / 2; i++)
101 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
102 }
103 mdct_context->imdct_half(mdct_context, out, spec);
104 }
105
106
107 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
108 {
109 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
110 unsigned int start_pos, ref_pos = 0, pos = 0;
111
112 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
113 float *prev_buf;
114 int j;
115
116 band_samples = samples_per_band[band_num];
117 log2_block_count = su->log2_block_count[band_num];
118
119 /* number of mdct blocks in the current QMF band: 1 - for long mode */
120 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
121 num_blocks = 1 << log2_block_count;
122
123 if (num_blocks == 1) {
124 /* mdct block size in samples: 128 (long mode, low & mid bands), */
125 /* 256 (long mode, high band) and 32 (short mode, all bands) */
126 block_size = band_samples >> log2_block_count;
127
128 /* calc transform size in bits according to the block_size_mode */
129 nbits = mdct_long_nbits[band_num] - log2_block_count;
130
131 if (nbits != 5 && nbits != 7 && nbits != 8)
132 return AVERROR_INVALIDDATA;
133 } else {
134 block_size = 32;
135 nbits = 5;
136 }
137
138 start_pos = 0;
139 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
140 for (j=0; j < num_blocks; j++) {
141 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
142
143 /* overlap and window */
144 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
145 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
146
147 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
148 start_pos += block_size;
149 pos += block_size;
150 }
151
152 if (num_blocks == 1)
153 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
154
155 ref_pos += band_samples;
156 }
157
158 /* Swap buffers so the mdct overlap works */
159 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
160
161 return 0;
162 }
163
164 /**
165 * Parse the block size mode byte
166 */
167
168 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
169 {
170 int log2_block_count_tmp, i;
171
172 for (i = 0; i < 2; i++) {
173 /* low and mid band */
174 log2_block_count_tmp = get_bits(gb, 2);
175 if (log2_block_count_tmp & 1)
176 return AVERROR_INVALIDDATA;
177 log2_block_cnt[i] = 2 - log2_block_count_tmp;
178 }
179
180 /* high band */
181 log2_block_count_tmp = get_bits(gb, 2);
182 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
183 return AVERROR_INVALIDDATA;
184 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
185
186 skip_bits(gb, 2);
187 return 0;
188 }
189
190
191 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
192 float spec[AT1_SU_SAMPLES])
193 {
194 int bits_used, band_num, bfu_num, i;
195 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
196 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
197
198 /* parse the info byte (2nd byte) telling how much BFUs were coded */
199 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
200
201 /* calc number of consumed bits:
202 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
203 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
204 bits_used = su->num_bfus * 10 + 32 +
205 bfu_amount_tab2[get_bits(gb, 2)] +
206 (bfu_amount_tab3[get_bits(gb, 3)] << 1);
207
208 /* get word length index (idwl) for each BFU */
209 for (i = 0; i < su->num_bfus; i++)
210 idwls[i] = get_bits(gb, 4);
211
212 /* get scalefactor index (idsf) for each BFU */
213 for (i = 0; i < su->num_bfus; i++)
214 idsfs[i] = get_bits(gb, 6);
215
216 /* zero idwl/idsf for empty BFUs */
217 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
218 idwls[i] = idsfs[i] = 0;
219
220 /* read in the spectral data and reconstruct MDCT spectrum of this channel */
221 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
222 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
223 int pos;
224
225 int num_specs = specs_per_bfu[bfu_num];
226 int word_len = !!idwls[bfu_num] + idwls[bfu_num];
227 float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]];
228 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
229
230 /* check for bitstream overflow */
231 if (bits_used > AT1_SU_MAX_BITS)
232 return AVERROR_INVALIDDATA;
233
234 /* get the position of the 1st spec according to the block size mode */
235 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
236
237 if (word_len) {
238 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
239
240 for (i = 0; i < num_specs; i++) {
241 /* read in a quantized spec and convert it to
242 * signed int and then inverse quantization
243 */
244 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
245 }
246 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
247 memset(&spec[pos], 0, num_specs * sizeof(float));
248 }
249 }
250 }
251
252 return 0;
253 }
254
255
256 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
257 {
258 float temp[256];
259 float iqmf_temp[512 + 46];
260
261 /* combine low and middle bands */
262 ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
263
264 /* delay the signal of the high band by 23 samples */
265 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
266 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
267
268 /* combine (low + middle) and high bands */
269 ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
270 }
271
272
273 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
274 int *got_frame_ptr, AVPacket *avpkt)
275 {
276 const uint8_t *buf = avpkt->data;
277 int buf_size = avpkt->size;
278 AT1Ctx *q = avctx->priv_data;
279 int ch, ret;
280 GetBitContext gb;
281
282
283 if (buf_size < 212 * avctx->channels) {
284 av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
285 return AVERROR_INVALIDDATA;
286 }
287
288 /* get output buffer */
289 q->frame.nb_samples = AT1_SU_SAMPLES;
290 if ((ret = ff_get_buffer(avctx, &q->frame)) < 0) {
291 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
292 return ret;
293 }
294
295 for (ch = 0; ch < avctx->channels; ch++) {
296 AT1SUCtx* su = &q->SUs[ch];
297
298 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
299
300 /* parse block_size_mode, 1st byte */
301 ret = at1_parse_bsm(&gb, su->log2_block_count);
302 if (ret < 0)
303 return ret;
304
305 ret = at1_unpack_dequant(&gb, su, q->spec);
306 if (ret < 0)
307 return ret;
308
309 ret = at1_imdct_block(su, q);
310 if (ret < 0)
311 return ret;
312 at1_subband_synthesis(q, su, (float *)q->frame.extended_data[ch]);
313 }
314
315 *got_frame_ptr = 1;
316 *(AVFrame *)data = q->frame;
317
318 return avctx->block_align;
319 }
320
321
322 static av_cold int atrac1_decode_end(AVCodecContext * avctx)
323 {
324 AT1Ctx *q = avctx->priv_data;
325
326 ff_mdct_end(&q->mdct_ctx[0]);
327 ff_mdct_end(&q->mdct_ctx[1]);
328 ff_mdct_end(&q->mdct_ctx[2]);
329
330 return 0;
331 }
332
333
334 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
335 {
336 AT1Ctx *q = avctx->priv_data;
337 int ret;
338
339 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
340
341 if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
342 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
343 avctx->channels);
344 return AVERROR(EINVAL);
345 }
346
347 /* Init the mdct transforms */
348 if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
349 (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
350 (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
351 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
352 atrac1_decode_end(avctx);
353 return ret;
354 }
355
356 ff_init_ff_sine_windows(5);
357
358 ff_atrac_generate_tables();
359
360 ff_dsputil_init(&q->dsp, avctx);
361
362 q->bands[0] = q->low;
363 q->bands[1] = q->mid;
364 q->bands[2] = q->high;
365
366 /* Prepare the mdct overlap buffers */
367 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
368 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
369 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
370 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
371
372 avcodec_get_frame_defaults(&q->frame);
373 avctx->coded_frame = &q->frame;
374
375 return 0;
376 }
377
378
379 AVCodec ff_atrac1_decoder = {
380 .name = "atrac1",
381 .type = AVMEDIA_TYPE_AUDIO,
382 .id = AV_CODEC_ID_ATRAC1,
383 .priv_data_size = sizeof(AT1Ctx),
384 .init = atrac1_decode_init,
385 .close = atrac1_decode_end,
386 .decode = atrac1_decode_frame,
387 .capabilities = CODEC_CAP_DR1,
388 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
389 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
390 AV_SAMPLE_FMT_NONE },
391 };