2 * ATRAC1 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 * ATRAC1 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"
43 #include "atrac1data.h"
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
52 #define AT1_QMF_BANDS 3
53 #define IDX_LOW_BAND 0
54 #define IDX_MID_BAND 1
55 #define IDX_HIGH_BAND 2
58 * Sound unit struct, one unit is used per channel
61 int log2_block_count
[AT1_QMF_BANDS
]; ///< log2 number of blocks in a band
62 int num_bfus
; ///< number of Block Floating Units
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
72 * The atrac1 context, holds all needed parameters for decoding
75 AT1SUCtx SUs
[AT1_MAX_CHANNELS
]; ///< channel sound unit
76 DECLARE_ALIGNED(32, float, spec
)[AT1_SU_SAMPLES
]; ///< the mdct spectrum buffer
78 DECLARE_ALIGNED(32, float, low
)[256];
79 DECLARE_ALIGNED(32, float, mid
)[256];
80 DECLARE_ALIGNED(32, float, high
)[512];
82 FFTContext mdct_ctx
[3];
83 AVFloatDSPContext
*fdsp
;
86 /** size of the transform in samples in the long mode for each QMF band */
87 static const uint16_t samples_per_band
[3] = {128, 128, 256};
88 static const uint8_t mdct_long_nbits
[3] = {7, 7, 8};
91 static void at1_imdct(AT1Ctx
*q
, float *spec
, float *out
, int nbits
,
94 FFTContext
* mdct_context
= &q
->mdct_ctx
[nbits
- 5 - (nbits
> 6)];
95 int transf_size
= 1 << nbits
;
99 for (i
= 0; i
< transf_size
/ 2; i
++)
100 FFSWAP(float, spec
[i
], spec
[transf_size
- 1 - i
]);
102 mdct_context
->imdct_half(mdct_context
, out
, spec
);
106 static int at1_imdct_block(AT1SUCtx
* su
, AT1Ctx
*q
)
108 int band_num
, band_samples
, log2_block_count
, nbits
, num_blocks
, block_size
;
109 unsigned int start_pos
, ref_pos
= 0, pos
= 0;
111 for (band_num
= 0; band_num
< AT1_QMF_BANDS
; band_num
++) {
115 band_samples
= samples_per_band
[band_num
];
116 log2_block_count
= su
->log2_block_count
[band_num
];
118 /* number of mdct blocks in the current QMF band: 1 - for long mode */
119 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
120 num_blocks
= 1 << log2_block_count
;
122 if (num_blocks
== 1) {
123 /* mdct block size in samples: 128 (long mode, low & mid bands), */
124 /* 256 (long mode, high band) and 32 (short mode, all bands) */
125 block_size
= band_samples
>> log2_block_count
;
127 /* calc transform size in bits according to the block_size_mode */
128 nbits
= mdct_long_nbits
[band_num
] - log2_block_count
;
130 if (nbits
!= 5 && nbits
!= 7 && nbits
!= 8)
131 return AVERROR_INVALIDDATA
;
138 prev_buf
= &su
->spectrum
[1][ref_pos
+ band_samples
- 16];
139 for (j
=0; j
< num_blocks
; j
++) {
140 at1_imdct(q
, &q
->spec
[pos
], &su
->spectrum
[0][ref_pos
+ start_pos
], nbits
, band_num
);
142 /* overlap and window */
143 q
->fdsp
->vector_fmul_window(&q
->bands
[band_num
][start_pos
], prev_buf
,
144 &su
->spectrum
[0][ref_pos
+ start_pos
], ff_sine_32
, 16);
146 prev_buf
= &su
->spectrum
[0][ref_pos
+start_pos
+ 16];
147 start_pos
+= block_size
;
152 memcpy(q
->bands
[band_num
] + 32, &su
->spectrum
[0][ref_pos
+ 16], 240 * sizeof(float));
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)
175 return AVERROR_INVALIDDATA
;
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)
182 return AVERROR_INVALIDDATA
;
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
= ff_atrac_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
)
231 return AVERROR_INVALIDDATA
;
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 empty BFU */
246 memset(&spec
[pos
], 0, num_specs
* sizeof(float));
255 static void at1_subband_synthesis(AT1Ctx
*q
, AT1SUCtx
* su
, float *pOut
)
258 float iqmf_temp
[512 + 46];
260 /* combine low and middle bands */
261 ff_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 ff_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 *got_frame_ptr
, AVPacket
*avpkt
)
275 AVFrame
*frame
= data
;
276 const uint8_t *buf
= avpkt
->data
;
277 int buf_size
= avpkt
->size
;
278 AT1Ctx
*q
= avctx
->priv_data
;
283 if (buf_size
< 212 * avctx
->channels
) {
284 av_log(avctx
, AV_LOG_ERROR
, "Not enough data to decode!\n");
285 return AVERROR_INVALIDDATA
;
288 /* get output buffer */
289 frame
->nb_samples
= AT1_SU_SAMPLES
;
290 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0)
293 for (ch
= 0; ch
< avctx
->channels
; ch
++) {
294 AT1SUCtx
* su
= &q
->SUs
[ch
];
296 init_get_bits(&gb
, &buf
[212 * ch
], 212 * 8);
298 /* parse block_size_mode, 1st byte */
299 ret
= at1_parse_bsm(&gb
, su
->log2_block_count
);
303 ret
= at1_unpack_dequant(&gb
, su
, q
->spec
);
307 ret
= at1_imdct_block(su
, q
);
310 at1_subband_synthesis(q
, su
, (float *)frame
->extended_data
[ch
]);
315 return avctx
->block_align
;
319 static av_cold
int atrac1_decode_end(AVCodecContext
* avctx
)
321 AT1Ctx
*q
= avctx
->priv_data
;
323 ff_mdct_end(&q
->mdct_ctx
[0]);
324 ff_mdct_end(&q
->mdct_ctx
[1]);
325 ff_mdct_end(&q
->mdct_ctx
[2]);
333 static av_cold
int atrac1_decode_init(AVCodecContext
*avctx
)
335 AT1Ctx
*q
= avctx
->priv_data
;
338 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLTP
;
340 if (avctx
->channels
< 1 || avctx
->channels
> AT1_MAX_CHANNELS
) {
341 av_log(avctx
, AV_LOG_ERROR
, "Unsupported number of channels: %d\n",
343 return AVERROR(EINVAL
);
346 if (avctx
->block_align
<= 0) {
347 av_log(avctx
, AV_LOG_ERROR
, "Unsupported block align.");
348 return AVERROR_PATCHWELCOME
;
351 /* Init the mdct transforms */
352 if ((ret
= ff_mdct_init(&q
->mdct_ctx
[0], 6, 1, -1.0/ (1 << 15))) ||
353 (ret
= ff_mdct_init(&q
->mdct_ctx
[1], 8, 1, -1.0/ (1 << 15))) ||
354 (ret
= ff_mdct_init(&q
->mdct_ctx
[2], 9, 1, -1.0/ (1 << 15)))) {
355 av_log(avctx
, AV_LOG_ERROR
, "Error initializing MDCT\n");
356 atrac1_decode_end(avctx
);
360 ff_init_ff_sine_windows(5);
362 ff_atrac_generate_tables();
364 q
->fdsp
= avpriv_float_dsp_alloc(avctx
->flags
& CODEC_FLAG_BITEXACT
);
366 q
->bands
[0] = q
->low
;
367 q
->bands
[1] = q
->mid
;
368 q
->bands
[2] = q
->high
;
370 /* Prepare the mdct overlap buffers */
371 q
->SUs
[0].spectrum
[0] = q
->SUs
[0].spec1
;
372 q
->SUs
[0].spectrum
[1] = q
->SUs
[0].spec2
;
373 q
->SUs
[1].spectrum
[0] = q
->SUs
[1].spec1
;
374 q
->SUs
[1].spectrum
[1] = q
->SUs
[1].spec2
;
380 AVCodec ff_atrac1_decoder
= {
382 .long_name
= NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
383 .type
= AVMEDIA_TYPE_AUDIO
,
384 .id
= AV_CODEC_ID_ATRAC1
,
385 .priv_data_size
= sizeof(AT1Ctx
),
386 .init
= atrac1_decode_init
,
387 .close
= atrac1_decode_end
,
388 .decode
= atrac1_decode_frame
,
389 .capabilities
= CODEC_CAP_DR1
,
390 .sample_fmts
= (const enum AVSampleFormat
[]) { AV_SAMPLE_FMT_FLTP
,
391 AV_SAMPLE_FMT_NONE
},