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! */
35 #include "libavutil/float_dsp.h"
44 #include "atrac1data.h"
46 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
47 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
48 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
49 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
50 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
51 #define AT1_MAX_CHANNELS 2
53 #define AT1_QMF_BANDS 3
54 #define IDX_LOW_BAND 0
55 #define IDX_MID_BAND 1
56 #define IDX_HIGH_BAND 2
59 * Sound unit struct, one unit is used per channel
62 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
63 int num_bfus; ///< number of Block Floating Units
65 DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
66 DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
67 DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
68 DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
69 DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
73 * The atrac1 context, holds all needed parameters for decoding
77 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
78 DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
80 DECLARE_ALIGNED(32, float, low)[256];
81 DECLARE_ALIGNED(32, float, mid)[256];
82 DECLARE_ALIGNED(32, float, high)[512];
84 FFTContext mdct_ctx[3];
85 AVFloatDSPContext fdsp;
88 /** size of the transform in samples in the long mode for each QMF band */
89 static const uint16_t samples_per_band[3] = {128, 128, 256};
90 static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
93 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
96 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
97 int transf_size = 1 << nbits;
101 for (i = 0; i < transf_size / 2; i++)
102 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
104 mdct_context->imdct_half(mdct_context, out, spec);
108 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
110 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
111 unsigned int start_pos, ref_pos = 0, pos = 0;
113 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
117 band_samples = samples_per_band[band_num];
118 log2_block_count = su->log2_block_count[band_num];
120 /* number of mdct blocks in the current QMF band: 1 - for long mode */
121 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
122 num_blocks = 1 << log2_block_count;
124 if (num_blocks == 1) {
125 /* mdct block size in samples: 128 (long mode, low & mid bands), */
126 /* 256 (long mode, high band) and 32 (short mode, all bands) */
127 block_size = band_samples >> log2_block_count;
129 /* calc transform size in bits according to the block_size_mode */
130 nbits = mdct_long_nbits[band_num] - log2_block_count;
132 if (nbits != 5 && nbits != 7 && nbits != 8)
133 return AVERROR_INVALIDDATA;
140 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
141 for (j=0; j < num_blocks; j++) {
142 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
144 /* overlap and window */
145 q->fdsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
146 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
148 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
149 start_pos += block_size;
154 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
156 ref_pos += band_samples;
159 /* Swap buffers so the mdct overlap works */
160 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
166 * Parse the block size mode byte
169 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
171 int log2_block_count_tmp, i;
173 for (i = 0; i < 2; i++) {
174 /* low and mid band */
175 log2_block_count_tmp = get_bits(gb, 2);
176 if (log2_block_count_tmp & 1)
177 return AVERROR_INVALIDDATA;
178 log2_block_cnt[i] = 2 - log2_block_count_tmp;
182 log2_block_count_tmp = get_bits(gb, 2);
183 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
184 return AVERROR_INVALIDDATA;
185 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
192 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
193 float spec[AT1_SU_SAMPLES])
195 int bits_used, band_num, bfu_num, i;
196 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
197 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
199 /* parse the info byte (2nd byte) telling how much BFUs were coded */
200 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
202 /* calc number of consumed bits:
203 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
204 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
205 bits_used = su->num_bfus * 10 + 32 +
206 bfu_amount_tab2[get_bits(gb, 2)] +
207 (bfu_amount_tab3[get_bits(gb, 3)] << 1);
209 /* get word length index (idwl) for each BFU */
210 for (i = 0; i < su->num_bfus; i++)
211 idwls[i] = get_bits(gb, 4);
213 /* get scalefactor index (idsf) for each BFU */
214 for (i = 0; i < su->num_bfus; i++)
215 idsfs[i] = get_bits(gb, 6);
217 /* zero idwl/idsf for empty BFUs */
218 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
219 idwls[i] = idsfs[i] = 0;
221 /* read in the spectral data and reconstruct MDCT spectrum of this channel */
222 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
223 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
226 int num_specs = specs_per_bfu[bfu_num];
227 int word_len = !!idwls[bfu_num] + idwls[bfu_num];
228 float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]];
229 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
231 /* check for bitstream overflow */
232 if (bits_used > AT1_SU_MAX_BITS)
233 return AVERROR_INVALIDDATA;
235 /* get the position of the 1st spec according to the block size mode */
236 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
239 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
241 for (i = 0; i < num_specs; i++) {
242 /* read in a quantized spec and convert it to
243 * signed int and then inverse quantization
245 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
247 } else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */
248 memset(&spec[pos], 0, num_specs * sizeof(float));
257 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
260 float iqmf_temp[512 + 46];
262 /* combine low and middle bands */
263 ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
265 /* delay the signal of the high band by 23 samples */
266 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
267 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
269 /* combine (low + middle) and high bands */
270 ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
274 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
275 int *got_frame_ptr, AVPacket *avpkt)
277 const uint8_t *buf = avpkt->data;
278 int buf_size = avpkt->size;
279 AT1Ctx *q = avctx->priv_data;
284 if (buf_size < 212 * avctx->channels) {
285 av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
286 return AVERROR_INVALIDDATA;
289 /* get output buffer */
290 q->frame.nb_samples = AT1_SU_SAMPLES;
291 if ((ret = ff_get_buffer(avctx, &q->frame)) < 0) {
292 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
296 for (ch = 0; ch < avctx->channels; ch++) {
297 AT1SUCtx* su = &q->SUs[ch];
299 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
301 /* parse block_size_mode, 1st byte */
302 ret = at1_parse_bsm(&gb, su->log2_block_count);
306 ret = at1_unpack_dequant(&gb, su, q->spec);
310 ret = at1_imdct_block(su, q);
313 at1_subband_synthesis(q, su, (float *)q->frame.extended_data[ch]);
317 *(AVFrame *)data = q->frame;
319 return avctx->block_align;
323 static av_cold int atrac1_decode_end(AVCodecContext * avctx)
325 AT1Ctx *q = avctx->priv_data;
327 ff_mdct_end(&q->mdct_ctx[0]);
328 ff_mdct_end(&q->mdct_ctx[1]);
329 ff_mdct_end(&q->mdct_ctx[2]);
335 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
337 AT1Ctx *q = avctx->priv_data;
340 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
342 if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
343 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
345 return AVERROR(EINVAL);
348 if (avctx->block_align <= 0) {
349 av_log_ask_for_sample(avctx, "unsupported block align\n");
350 return AVERROR_PATCHWELCOME;
353 /* Init the mdct transforms */
354 if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
355 (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
356 (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
357 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
358 atrac1_decode_end(avctx);
362 ff_init_ff_sine_windows(5);
364 ff_atrac_generate_tables();
366 avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
368 q->bands[0] = q->low;
369 q->bands[1] = q->mid;
370 q->bands[2] = q->high;
372 /* Prepare the mdct overlap buffers */
373 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
374 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
375 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
376 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
378 avcodec_get_frame_defaults(&q->frame);
379 avctx->coded_frame = &q->frame;
385 AVCodec ff_atrac1_decoder = {
387 .type = AVMEDIA_TYPE_AUDIO,
388 .id = AV_CODEC_ID_ATRAC1,
389 .priv_data_size = sizeof(AT1Ctx),
390 .init = atrac1_decode_init,
391 .close = atrac1_decode_end,
392 .decode = atrac1_decode_frame,
393 .capabilities = CODEC_CAP_DR1,
394 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
395 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
396 AV_SAMPLE_FMT_NONE },