2 * Atrac 1 compatible decoder
3 * Copyright (c) 2009 Maxim Poliakovski
4 * Copyright (c) 2009 Benjamin Larsson
6 * This file is part of FFmpeg.
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.
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.
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
24 * @file libavcodec/atrac1.c
25 * Atrac 1 compatible decoder.
26 * This decoder handles raw ATRAC1 data and probably SDDS data.
29 /* Many thanks to Tim Craig for all the help! */
40 #include "atrac1data.h"
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
49 #define AT1_QMF_BANDS 3
50 #define IDX_LOW_BAND 0
51 #define IDX_MID_BAND 1
52 #define IDX_HIGH_BAND 2
55 * Sound unit struct, one unit is used per channel
58 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
59 int num_bfus; ///< number of Block Floating Units
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
69 * The atrac1 context, holds all needed parameters for decoding
72 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
73 DECLARE_ALIGNED_16(float, spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
75 DECLARE_ALIGNED_16(float, low[256]);
76 DECLARE_ALIGNED_16(float, mid[256]);
77 DECLARE_ALIGNED_16(float, high[512]);
79 DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
80 FFTContext mdct_ctx[3];
85 /** size of the transform in samples in the long mode for each QMF band */
86 static const uint16_t samples_per_band[3] = {128, 128, 256};
87 static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
90 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
93 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
94 int transf_size = 1 << nbits;
98 for (i = 0; i < transf_size / 2; i++)
99 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
101 ff_imdct_half(mdct_context, out, spec);
105 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
107 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
108 unsigned int start_pos, ref_pos = 0, pos = 0;
110 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
111 band_samples = samples_per_band[band_num];
112 log2_block_count = su->log2_block_count[band_num];
114 /* number of mdct blocks in the current QMF band: 1 - for long mode */
115 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
116 num_blocks = 1 << log2_block_count;
118 /* mdct block size in samples: 128 (long mode, low & mid bands), */
119 /* 256 (long mode, high band) and 32 (short mode, all bands) */
120 block_size = band_samples >> log2_block_count;
122 /* calc transform size in bits according to the block_size_mode */
123 nbits = mdct_long_nbits[band_num] - log2_block_count;
125 if (nbits != 5 && nbits != 7 && nbits != 8)
128 if (num_blocks == 1) {
130 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num);
131 pos += block_size; // move to the next mdct block in the spectrum
133 /* overlap and window long blocks */
134 q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos + band_samples - 16],
135 &su->spectrum[0][ref_pos], ff_sine_32, 0, 16);
136 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
141 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
142 for (; num_blocks != 0; num_blocks--) {
143 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], 5, band_num);
145 /* overlap and window between short blocks */
146 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
147 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
149 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
150 start_pos += 32; // use hardcoded block_size
154 ref_pos += band_samples;
157 /* Swap buffers so the mdct overlap works */
158 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
164 * Parse the block size mode byte
167 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
169 int log2_block_count_tmp, i;
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)
176 log2_block_cnt[i] = 2 - log2_block_count_tmp;
180 log2_block_count_tmp = get_bits(gb, 2);
181 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
183 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
190 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
191 float spec[AT1_SU_SAMPLES])
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
197 /* parse the info byte (2nd byte) telling how much BFUs were coded */
198 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
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);
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);
211 /* get scalefactor index (idsf) for each BFU */
212 for (i = 0; i < su->num_bfus; i++)
213 idsfs[i] = get_bits(gb, 6);
215 /* zero idwl/idsf for empty BFUs */
216 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
217 idwls[i] = idsfs[i] = 0;
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++) {
224 int num_specs = specs_per_bfu[bfu_num];
225 int word_len = !!idwls[bfu_num] + idwls[bfu_num];
226 float scale_factor = sf_table[idsfs[bfu_num]];
227 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
229 /* check for bitstream overflow */
230 if (bits_used > AT1_SU_MAX_BITS)
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];
237 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
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
243 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
245 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
246 memset(&spec[pos], 0, num_specs * sizeof(float));
255 void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
258 float iqmf_temp[512 + 46];
260 /* combine low and middle bands */
261 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
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);
267 /* combine (low + middle) and high bands */
268 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
272 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
273 int *data_size, AVPacket *avpkt)
275 const uint8_t *buf = avpkt->data;
276 int buf_size = avpkt->size;
277 AT1Ctx *q = avctx->priv_data;
280 float* samples = data;
283 if (buf_size < 212 * q->channels) {
284 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
288 for (ch = 0; ch < q->channels; ch++) {
289 AT1SUCtx* su = &q->SUs[ch];
291 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
293 /* parse block_size_mode, 1st byte */
294 ret = at1_parse_bsm(&gb, su->log2_block_count);
298 ret = at1_unpack_dequant(&gb, su, q->spec);
302 ret = at1_imdct_block(su, q);
305 at1_subband_synthesis(q, su, q->out_samples[ch]);
308 /* round, convert to 16bit and interleave */
309 if (q->channels == 1) {
311 q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
312 32700.0 / (1 << 15), AT1_SU_SAMPLES);
315 for (i = 0; i < AT1_SU_SAMPLES; i++) {
316 samples[i * 2] = av_clipf(q->out_samples[0][i],
317 -32700.0 / (1 << 15),
318 32700.0 / (1 << 15));
319 samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
320 -32700.0 / (1 << 15),
321 32700.0 / (1 << 15));
325 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
326 return avctx->block_align;
330 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
332 AT1Ctx *q = avctx->priv_data;
334 avctx->sample_fmt = SAMPLE_FMT_FLT;
336 q->channels = avctx->channels;
338 /* Init the mdct transforms */
339 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
340 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
341 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
343 ff_sine_window_init(ff_sine_32, 32);
345 atrac_generate_tables();
347 dsputil_init(&q->dsp, avctx);
349 q->bands[0] = q->low;
350 q->bands[1] = q->mid;
351 q->bands[2] = q->high;
353 /* Prepare the mdct overlap buffers */
354 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
355 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
356 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
357 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
363 static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
364 AT1Ctx *q = avctx->priv_data;
366 ff_mdct_end(&q->mdct_ctx[0]);
367 ff_mdct_end(&q->mdct_ctx[1]);
368 ff_mdct_end(&q->mdct_ctx[2]);
373 AVCodec atrac1_decoder = {
375 .type = CODEC_TYPE_AUDIO,
376 .id = CODEC_ID_ATRAC1,
377 .priv_data_size = sizeof(AT1Ctx),
378 .init = atrac1_decode_init,
379 .close = atrac1_decode_end,
380 .decode = atrac1_decode_frame,
381 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),