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
++) {
114 band_samples
= samples_per_band
[band_num
];
115 log2_block_count
= su
->log2_block_count
[band_num
];
117 /* number of mdct blocks in the current QMF band: 1 - for long mode */
118 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
119 num_blocks
= 1 << log2_block_count
;
121 if (num_blocks
== 1) {
122 /* mdct block size in samples: 128 (long mode, low & mid bands), */
123 /* 256 (long mode, high band) and 32 (short mode, all bands) */
124 block_size
= band_samples
>> log2_block_count
;
126 /* calc transform size in bits according to the block_size_mode */
127 nbits
= mdct_long_nbits
[band_num
] - log2_block_count
;
129 if (nbits
!= 5 && nbits
!= 7 && nbits
!= 8)
137 prev_buf
= &su
->spectrum
[1][ref_pos
+ band_samples
- 16];
138 for (j
=0; j
< num_blocks
; j
++) {
139 at1_imdct(q
, &q
->spec
[pos
], &su
->spectrum
[0][ref_pos
+ start_pos
], nbits
, band_num
);
141 /* overlap and window */
142 q
->dsp
.vector_fmul_window(&q
->bands
[band_num
][start_pos
], prev_buf
,
143 &su
->spectrum
[0][ref_pos
+ start_pos
], ff_sine_32
, 0, 16);
145 prev_buf
= &su
->spectrum
[0][ref_pos
+start_pos
+ 16];
146 start_pos
+= block_size
;
151 memcpy(q
->bands
[band_num
] + 32, &su
->spectrum
[0][ref_pos
+ 16], 240 * sizeof(float));
153 ref_pos
+= band_samples
;
156 /* Swap buffers so the mdct overlap works */
157 FFSWAP(float*, su
->spectrum
[0], su
->spectrum
[1]);
163 * Parse the block size mode byte
166 static int at1_parse_bsm(GetBitContext
* gb
, int log2_block_cnt
[AT1_QMF_BANDS
])
168 int log2_block_count_tmp
, i
;
170 for (i
= 0; i
< 2; i
++) {
171 /* low and mid band */
172 log2_block_count_tmp
= get_bits(gb
, 2);
173 if (log2_block_count_tmp
& 1)
175 log2_block_cnt
[i
] = 2 - log2_block_count_tmp
;
179 log2_block_count_tmp
= get_bits(gb
, 2);
180 if (log2_block_count_tmp
!= 0 && log2_block_count_tmp
!= 3)
182 log2_block_cnt
[IDX_HIGH_BAND
] = 3 - log2_block_count_tmp
;
189 static int at1_unpack_dequant(GetBitContext
* gb
, AT1SUCtx
* su
,
190 float spec
[AT1_SU_SAMPLES
])
192 int bits_used
, band_num
, bfu_num
, i
;
193 uint8_t idwls
[AT1_MAX_BFU
]; ///< the word length indexes for each BFU
194 uint8_t idsfs
[AT1_MAX_BFU
]; ///< the scalefactor indexes for each BFU
196 /* parse the info byte (2nd byte) telling how much BFUs were coded */
197 su
->num_bfus
= bfu_amount_tab1
[get_bits(gb
, 3)];
199 /* calc number of consumed bits:
200 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
201 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
202 bits_used
= su
->num_bfus
* 10 + 32 +
203 bfu_amount_tab2
[get_bits(gb
, 2)] +
204 (bfu_amount_tab3
[get_bits(gb
, 3)] << 1);
206 /* get word length index (idwl) for each BFU */
207 for (i
= 0; i
< su
->num_bfus
; i
++)
208 idwls
[i
] = get_bits(gb
, 4);
210 /* get scalefactor index (idsf) for each BFU */
211 for (i
= 0; i
< su
->num_bfus
; i
++)
212 idsfs
[i
] = get_bits(gb
, 6);
214 /* zero idwl/idsf for empty BFUs */
215 for (i
= su
->num_bfus
; i
< AT1_MAX_BFU
; i
++)
216 idwls
[i
] = idsfs
[i
] = 0;
218 /* read in the spectral data and reconstruct MDCT spectrum of this channel */
219 for (band_num
= 0; band_num
< AT1_QMF_BANDS
; band_num
++) {
220 for (bfu_num
= bfu_bands_t
[band_num
]; bfu_num
< bfu_bands_t
[band_num
+1]; bfu_num
++) {
223 int num_specs
= specs_per_bfu
[bfu_num
];
224 int word_len
= !!idwls
[bfu_num
] + idwls
[bfu_num
];
225 float scale_factor
= sf_table
[idsfs
[bfu_num
]];
226 bits_used
+= word_len
* num_specs
; /* add number of bits consumed by current BFU */
228 /* check for bitstream overflow */
229 if (bits_used
> AT1_SU_MAX_BITS
)
232 /* get the position of the 1st spec according to the block size mode */
233 pos
= su
->log2_block_count
[band_num
] ? bfu_start_short
[bfu_num
] : bfu_start_long
[bfu_num
];
236 float max_quant
= 1.0 / (float)((1 << (word_len
- 1)) - 1);
238 for (i
= 0; i
< num_specs
; i
++) {
239 /* read in a quantized spec and convert it to
240 * signed int and then inverse quantization
242 spec
[pos
+i
] = get_sbits(gb
, word_len
) * scale_factor
* max_quant
;
244 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
245 memset(&spec
[pos
], 0, num_specs
* sizeof(float));
254 void at1_subband_synthesis(AT1Ctx
*q
, AT1SUCtx
* su
, float *pOut
)
257 float iqmf_temp
[512 + 46];
259 /* combine low and middle bands */
260 atrac_iqmf(q
->bands
[0], q
->bands
[1], 128, temp
, su
->fst_qmf_delay
, iqmf_temp
);
262 /* delay the signal of the high band by 23 samples */
263 memcpy( su
->last_qmf_delay
, &su
->last_qmf_delay
[256], sizeof(float) * 23);
264 memcpy(&su
->last_qmf_delay
[23], q
->bands
[2], sizeof(float) * 256);
266 /* combine (low + middle) and high bands */
267 atrac_iqmf(temp
, su
->last_qmf_delay
, 256, pOut
, su
->snd_qmf_delay
, iqmf_temp
);
271 static int atrac1_decode_frame(AVCodecContext
*avctx
, void *data
,
272 int *data_size
, AVPacket
*avpkt
)
274 const uint8_t *buf
= avpkt
->data
;
275 int buf_size
= avpkt
->size
;
276 AT1Ctx
*q
= avctx
->priv_data
;
279 float* samples
= data
;
282 if (buf_size
< 212 * q
->channels
) {
283 av_log(q
,AV_LOG_ERROR
,"Not enought data to decode!\n");
287 for (ch
= 0; ch
< q
->channels
; ch
++) {
288 AT1SUCtx
* su
= &q
->SUs
[ch
];
290 init_get_bits(&gb
, &buf
[212 * ch
], 212 * 8);
292 /* parse block_size_mode, 1st byte */
293 ret
= at1_parse_bsm(&gb
, su
->log2_block_count
);
297 ret
= at1_unpack_dequant(&gb
, su
, q
->spec
);
301 ret
= at1_imdct_block(su
, q
);
304 at1_subband_synthesis(q
, su
, q
->out_samples
[ch
]);
307 /* round, convert to 16bit and interleave */
308 if (q
->channels
== 1) {
310 q
->dsp
.vector_clipf(samples
, q
->out_samples
[0], -32700.0 / (1 << 15),
311 32700.0 / (1 << 15), AT1_SU_SAMPLES
);
314 for (i
= 0; i
< AT1_SU_SAMPLES
; i
++) {
315 samples
[i
* 2] = av_clipf(q
->out_samples
[0][i
],
316 -32700.0 / (1 << 15),
317 32700.0 / (1 << 15));
318 samples
[i
* 2 + 1] = av_clipf(q
->out_samples
[1][i
],
319 -32700.0 / (1 << 15),
320 32700.0 / (1 << 15));
324 *data_size
= q
->channels
* AT1_SU_SAMPLES
* sizeof(*samples
);
325 return avctx
->block_align
;
329 static av_cold
int atrac1_decode_init(AVCodecContext
*avctx
)
331 AT1Ctx
*q
= avctx
->priv_data
;
333 avctx
->sample_fmt
= SAMPLE_FMT_FLT
;
335 q
->channels
= avctx
->channels
;
337 /* Init the mdct transforms */
338 ff_mdct_init(&q
->mdct_ctx
[0], 6, 1, -1.0/ (1 << 15));
339 ff_mdct_init(&q
->mdct_ctx
[1], 8, 1, -1.0/ (1 << 15));
340 ff_mdct_init(&q
->mdct_ctx
[2], 9, 1, -1.0/ (1 << 15));
342 ff_sine_window_init(ff_sine_32
, 32);
344 atrac_generate_tables();
346 dsputil_init(&q
->dsp
, avctx
);
348 q
->bands
[0] = q
->low
;
349 q
->bands
[1] = q
->mid
;
350 q
->bands
[2] = q
->high
;
352 /* Prepare the mdct overlap buffers */
353 q
->SUs
[0].spectrum
[0] = q
->SUs
[0].spec1
;
354 q
->SUs
[0].spectrum
[1] = q
->SUs
[0].spec2
;
355 q
->SUs
[1].spectrum
[0] = q
->SUs
[1].spec1
;
356 q
->SUs
[1].spectrum
[1] = q
->SUs
[1].spec2
;
362 static av_cold
int atrac1_decode_end(AVCodecContext
* avctx
) {
363 AT1Ctx
*q
= avctx
->priv_data
;
365 ff_mdct_end(&q
->mdct_ctx
[0]);
366 ff_mdct_end(&q
->mdct_ctx
[1]);
367 ff_mdct_end(&q
->mdct_ctx
[2]);
372 AVCodec atrac1_decoder
= {
374 .type
= CODEC_TYPE_AUDIO
,
375 .id
= CODEC_ID_ATRAC1
,
376 .priv_data_size
= sizeof(AT1Ctx
),
377 .init
= atrac1_decode_init
,
378 .close
= atrac1_decode_end
,
379 .decode
= atrac1_decode_frame
,
380 .long_name
= NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),