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
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! */
42 #include "atrac1data.h"
44 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
45 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
46 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
47 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
48 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
49 #define AT1_MAX_CHANNELS 2
51 #define AT1_QMF_BANDS 3
52 #define IDX_LOW_BAND 0
53 #define IDX_MID_BAND 1
54 #define IDX_HIGH_BAND 2
57 * Sound unit struct, one unit is used per channel
60 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
61 int num_bfus; ///< number of Block Floating Units
63 DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
64 DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
65 DECLARE_ALIGNED(16, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
66 DECLARE_ALIGNED(16, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
67 DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
71 * The atrac1 context, holds all needed parameters for decoding
74 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
75 DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
77 DECLARE_ALIGNED(16, float, low)[256];
78 DECLARE_ALIGNED(16, float, mid)[256];
79 DECLARE_ALIGNED(16, float, high)[512];
81 DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
82 FFTContext mdct_ctx[3];
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};
92 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
95 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
96 int transf_size = 1 << nbits;
100 for (i = 0; i < transf_size / 2; i++)
101 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
103 mdct_context->imdct_half(mdct_context, out, spec);
107 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
109 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
110 unsigned int start_pos, ref_pos = 0, pos = 0;
112 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
116 band_samples = samples_per_band[band_num];
117 log2_block_count = su->log2_block_count[band_num];
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;
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;
128 /* calc transform size in bits according to the block_size_mode */
129 nbits = mdct_long_nbits[band_num] - log2_block_count;
131 if (nbits != 5 && nbits != 7 && nbits != 8)
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);
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);
147 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
148 start_pos += block_size;
153 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
155 ref_pos += band_samples;
158 /* Swap buffers so the mdct overlap works */
159 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
165 * Parse the block size mode byte
168 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
170 int log2_block_count_tmp, i;
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)
177 log2_block_cnt[i] = 2 - log2_block_count_tmp;
181 log2_block_count_tmp = get_bits(gb, 2);
182 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
184 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
191 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
192 float spec[AT1_SU_SAMPLES])
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
198 /* parse the info byte (2nd byte) telling how much BFUs were coded */
199 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
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);
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);
212 /* get scalefactor index (idsf) for each BFU */
213 for (i = 0; i < su->num_bfus; i++)
214 idsfs[i] = get_bits(gb, 6);
216 /* zero idwl/idsf for empty BFUs */
217 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
218 idwls[i] = idsfs[i] = 0;
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++) {
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 */
230 /* check for bitstream overflow */
231 if (bits_used > AT1_SU_MAX_BITS)
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];
238 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
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
244 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
246 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
247 memset(&spec[pos], 0, num_specs * sizeof(float));
256 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
259 float iqmf_temp[512 + 46];
261 /* combine low and middle bands */
262 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
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);
268 /* combine (low + middle) and high bands */
269 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
273 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
274 int *data_size, AVPacket *avpkt)
276 const uint8_t *buf = avpkt->data;
277 int buf_size = avpkt->size;
278 AT1Ctx *q = avctx->priv_data;
281 float* samples = data;
284 if (buf_size < 212 * q->channels) {
285 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
289 for (ch = 0; ch < q->channels; ch++) {
290 AT1SUCtx* su = &q->SUs[ch];
292 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
294 /* parse block_size_mode, 1st byte */
295 ret = at1_parse_bsm(&gb, su->log2_block_count);
299 ret = at1_unpack_dequant(&gb, su, q->spec);
303 ret = at1_imdct_block(su, q);
306 at1_subband_synthesis(q, su, q->out_samples[ch]);
309 /* interleave; FIXME, should create/use a DSP function */
310 if (q->channels == 1) {
312 memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
315 for (i = 0; i < AT1_SU_SAMPLES; i++) {
316 samples[i * 2] = q->out_samples[0][i];
317 samples[i * 2 + 1] = q->out_samples[1][i];
321 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
322 return avctx->block_align;
326 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
328 AT1Ctx *q = avctx->priv_data;
330 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
332 q->channels = avctx->channels;
334 /* Init the mdct transforms */
335 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
336 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
337 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
339 ff_init_ff_sine_windows(5);
341 atrac_generate_tables();
343 dsputil_init(&q->dsp, avctx);
345 q->bands[0] = q->low;
346 q->bands[1] = q->mid;
347 q->bands[2] = q->high;
349 /* Prepare the mdct overlap buffers */
350 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
351 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
352 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
353 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
359 static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
360 AT1Ctx *q = avctx->priv_data;
362 ff_mdct_end(&q->mdct_ctx[0]);
363 ff_mdct_end(&q->mdct_ctx[1]);
364 ff_mdct_end(&q->mdct_ctx[2]);
369 AVCodec ff_atrac1_decoder = {
371 .type = AVMEDIA_TYPE_AUDIO,
372 .id = CODEC_ID_ATRAC1,
373 .priv_data_size = sizeof(AT1Ctx),
374 .init = atrac1_decode_init,
375 .close = atrac1_decode_end,
376 .decode = atrac1_decode_frame,
377 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),