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atrac1.c
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1 /*
2  * ATRAC1 compatible decoder
3  * Copyright (c) 2009 Maxim Poliakovski
4  * Copyright (c) 2009 Benjamin Larsson
5  *
6  * This file is part of FFmpeg.
7  *
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.
12  *
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.
17  *
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
21  */
22 
23 /**
24  * @file
25  * ATRAC1 compatible decoder.
26  * This decoder handles raw ATRAC1 data and probably SDDS data.
27  */
28 
29 /* Many thanks to Tim Craig for all the help! */
30 
31 #include <math.h>
32 #include <stddef.h>
33 #include <stdio.h>
34 
35 #include "libavutil/float_dsp.h"
36 #include "avcodec.h"
37 #include "get_bits.h"
38 #include "fft.h"
39 #include "internal.h"
40 #include "sinewin.h"
41 
42 #include "atrac.h"
43 #include "atrac1data.h"
44 
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
51 
52 #define AT1_QMF_BANDS 3
53 #define IDX_LOW_BAND 0
54 #define IDX_MID_BAND 1
55 #define IDX_HIGH_BAND 2
56 
57 /**
58  * Sound unit struct, one unit is used per channel
59  */
60 typedef struct AT1SUCtx {
61  int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
62  int num_bfus; ///< number of Block Floating Units
63  float* spectrum[2];
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+39]; ///< delay line for the last stacked QMF filter
69 } AT1SUCtx;
70 
71 /**
72  * The atrac1 context, holds all needed parameters for decoding
73  */
74 typedef struct AT1Ctx {
75  AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
76  DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
77 
78  DECLARE_ALIGNED(32, float, low)[256];
79  DECLARE_ALIGNED(32, float, mid)[256];
80  DECLARE_ALIGNED(32, float, high)[512];
81  float* bands[3];
84 } AT1Ctx;
85 
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};
89 
90 
91 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
92  int rev_spec)
93 {
94  FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
95  int transf_size = 1 << nbits;
96 
97  if (rev_spec) {
98  int i;
99  for (i = 0; i < transf_size / 2; i++)
100  FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
101  }
102  mdct_context->imdct_half(mdct_context, out, spec);
103 }
104 
105 
106 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
107 {
108  int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
109  unsigned int start_pos, ref_pos = 0, pos = 0;
110 
111  for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
112  float *prev_buf;
113  int j;
114 
115  band_samples = samples_per_band[band_num];
116  log2_block_count = su->log2_block_count[band_num];
117 
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;
121 
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;
126 
127  /* calc transform size in bits according to the block_size_mode */
128  nbits = mdct_long_nbits[band_num] - log2_block_count;
129 
130  if (nbits != 5 && nbits != 7 && nbits != 8)
131  return AVERROR_INVALIDDATA;
132  } else {
133  block_size = 32;
134  nbits = 5;
135  }
136 
137  start_pos = 0;
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);
141 
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);
145 
146  prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
147  start_pos += block_size;
148  pos += block_size;
149  }
150 
151  if (num_blocks == 1)
152  memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
153 
154  ref_pos += band_samples;
155  }
156 
157  /* Swap buffers so the mdct overlap works */
158  FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
159 
160  return 0;
161 }
162 
163 /**
164  * Parse the block size mode byte
165  */
166 
167 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
168 {
169  int log2_block_count_tmp, i;
170 
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;
177  }
178 
179  /* high band */
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;
184 
185  skip_bits(gb, 2);
186  return 0;
187 }
188 
189 
191  float spec[AT1_SU_SAMPLES])
192 {
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
196 
197  /* parse the info byte (2nd byte) telling how much BFUs were coded */
198  su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
199 
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);
206 
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);
210 
211  /* get scalefactor index (idsf) for each BFU */
212  for (i = 0; i < su->num_bfus; i++)
213  idsfs[i] = get_bits(gb, 6);
214 
215  /* zero idwl/idsf for empty BFUs */
216  for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
217  idwls[i] = idsfs[i] = 0;
218 
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++) {
222  int pos;
223 
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 */
228 
229  /* check for bitstream overflow */
230  if (bits_used > AT1_SU_MAX_BITS)
231  return AVERROR_INVALIDDATA;
232 
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];
235 
236  if (word_len) {
237  float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
238 
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
242  */
243  spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
244  }
245  } else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */
246  memset(&spec[pos], 0, num_specs * sizeof(float));
247  }
248  }
249  }
250 
251  return 0;
252 }
253 
254 
255 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
256 {
257  float temp[256];
258  float iqmf_temp[512 + 46];
259 
260  /* combine low and middle bands */
261  ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
262 
263  /* delay the signal of the high band by 39 samples */
264  memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 39);
265  memcpy(&su->last_qmf_delay[39], q->bands[2], sizeof(float) * 256);
266 
267  /* combine (low + middle) and high bands */
268  ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
269 }
270 
271 
272 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
273  int *got_frame_ptr, AVPacket *avpkt)
274 {
275  AVFrame *frame = data;
276  const uint8_t *buf = avpkt->data;
277  int buf_size = avpkt->size;
278  AT1Ctx *q = avctx->priv_data;
279  int ch, ret;
280  GetBitContext gb;
281 
282 
283  if (buf_size < 212 * avctx->channels) {
284  av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
285  return AVERROR_INVALIDDATA;
286  }
287 
288  /* get output buffer */
289  frame->nb_samples = AT1_SU_SAMPLES;
290  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
291  return ret;
292 
293  for (ch = 0; ch < avctx->channels; ch++) {
294  AT1SUCtx* su = &q->SUs[ch];
295 
296  init_get_bits(&gb, &buf[212 * ch], 212 * 8);
297 
298  /* parse block_size_mode, 1st byte */
299  ret = at1_parse_bsm(&gb, su->log2_block_count);
300  if (ret < 0)
301  return ret;
302 
303  ret = at1_unpack_dequant(&gb, su, q->spec);
304  if (ret < 0)
305  return ret;
306 
307  ret = at1_imdct_block(su, q);
308  if (ret < 0)
309  return ret;
310  at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]);
311  }
312 
313  *got_frame_ptr = 1;
314 
315  return avctx->block_align;
316 }
317 
318 
320 {
321  AT1Ctx *q = avctx->priv_data;
322 
323  ff_mdct_end(&q->mdct_ctx[0]);
324  ff_mdct_end(&q->mdct_ctx[1]);
325  ff_mdct_end(&q->mdct_ctx[2]);
326 
327  av_freep(&q->fdsp);
328 
329  return 0;
330 }
331 
332 
334 {
335  AT1Ctx *q = avctx->priv_data;
336  int ret;
337 
339 
340  if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
341  av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
342  avctx->channels);
343  return AVERROR(EINVAL);
344  }
345 
346  if (avctx->block_align <= 0) {
347  av_log(avctx, AV_LOG_ERROR, "Unsupported block align.");
348  return AVERROR_PATCHWELCOME;
349  }
350 
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);
357  return ret;
358  }
359 
361 
363 
365 
366  q->bands[0] = q->low;
367  q->bands[1] = q->mid;
368  q->bands[2] = q->high;
369 
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;
375 
376  return 0;
377 }
378 
379 
381  .name = "atrac1",
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),
387  .close = atrac1_decode_end,
389  .capabilities = AV_CODEC_CAP_DR1,
390  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
392 };
float, planar
Definition: samplefmt.h:69
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
float snd_qmf_delay[46]
delay line for the 2nd stacked QMF filter
Definition: atrac1.c:67
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
AVCodec ff_atrac1_decoder
Definition: atrac1.c:380
Sound unit struct, one unit is used per channel.
Definition: atrac1.c:60
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:247
static int at1_unpack_dequant(GetBitContext *gb, AT1SUCtx *su, float spec[AT1_SU_SAMPLES])
Definition: atrac1.c:190
else temp
Definition: vf_mcdeint.c:259
AT1SUCtx SUs[AT1_MAX_CHANNELS]
channel sound unit
Definition: atrac1.c:75
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
Definition: atrac1.c:333
static av_cold int atrac1_decode_end(AVCodecContext *avctx)
Definition: atrac1.c:319
int size
Definition: avcodec.h:1602
static const uint8_t bfu_amount_tab2[4]
Definition: atrac1data.h:34
void ff_atrac_iqmf(float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
Quadrature mirror synthesis filter.
Definition: atrac.c:127
#define AT1_MAX_CHANNELS
Definition: atrac1.c:50
AVCodec.
Definition: avcodec.h:3601
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2475
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:232
static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx *su, float *pOut)
Definition: atrac1.c:255
static const uint8_t specs_per_bfu[52]
number of spectral lines in each BFU block floating unit = group of spectral frequencies having the s...
Definition: atrac1data.h:44
float ff_atrac_sf_table[64]
Definition: atrac.c:36
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
Definition: float_dsp.h:103
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2446
uint8_t
#define av_cold
Definition: attributes.h:82
static const uint16_t samples_per_band[3]
size of the transform in samples in the long mode for each QMF band
Definition: atrac1.c:87
The atrac1 context, holds all needed parameters for decoding.
Definition: atrac1.c:74
static AVFrame * frame
float spec2[AT1_SU_SAMPLES]
mdct buffer
Definition: atrac1.c:65
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:101
uint8_t * data
Definition: avcodec.h:1601
#define AT1_SU_SAMPLES
number of samples in a sound unit
Definition: atrac1.c:47
ATRAC common header.
bitstream reader API header.
static int at1_parse_bsm(GetBitContext *gb, int log2_block_cnt[AT1_QMF_BANDS])
Parse the block size mode byte.
Definition: atrac1.c:167
#define av_log(a,...)
float mid[256]
Definition: atrac1.c:79
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec)
Definition: atrac1.c:91
#define AVERROR(e)
Definition: error.h:43
static const uint16_t bfu_start_short[52]
start position of each BFU in the MDCT spectrum for the short mode
Definition: atrac1data.h:58
static const uint8_t bfu_amount_tab1[8]
Definition: atrac1data.h:33
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1771
const char * name
Name of the codec implementation.
Definition: avcodec.h:3608
#define ff_mdct_init
Definition: fft.h:169
float spec1[AT1_SU_SAMPLES]
mdct buffer
Definition: atrac1.c:64
AVFloatDSPContext * fdsp
Definition: atrac1.c:83
#define AT1_QMF_BANDS
Definition: atrac1.c:52
int log2_block_count[AT1_QMF_BANDS]
log2 number of blocks in a band
Definition: atrac1.c:61
Definition: fft.h:88
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:886
#define IDX_HIGH_BAND
Definition: atrac1.c:55
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
#define AT1_SU_MAX_BITS
Definition: atrac1.c:49
float fst_qmf_delay[46]
delay line for the 1st stacked QMF filter
Definition: atrac1.c:66
main external API structure.
Definition: avcodec.h:1676
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:947
void * buf
Definition: avisynth_c.h:690
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:108
static int atrac1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: atrac1.c:272
float high[512]
Definition: atrac1.c:80
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:292
FFTContext mdct_ctx[3]
Definition: atrac1.c:82
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:410
float * spectrum[2]
Definition: atrac1.c:63
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:119
static const uint8_t mdct_long_nbits[3]
Definition: atrac1.c:88
#define AT1_MAX_BFU
max number of block floating units in a sound unit
Definition: atrac1.c:45
static const uint8_t bfu_bands_t[4]
number of BFUs in each QMF band
Definition: atrac1data.h:38
float spec[AT1_SU_SAMPLES]
the mdct spectrum buffer
Definition: atrac1.c:76
common internal api header.
#define ff_mdct_end
Definition: fft.h:170
float * bands[3]
Definition: atrac1.c:81
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
ATRAC1 compatible decoder data.
void * priv_data
Definition: avcodec.h:1718
float last_qmf_delay[256+39]
delay line for the last stacked QMF filter
Definition: atrac1.c:68
int channels
number of audio channels
Definition: avcodec.h:2439
float low[256]
Definition: atrac1.c:78
av_cold void ff_atrac_generate_tables(void)
Generate common tables.
Definition: atrac.c:48
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
#define av_freep(p)
static int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt)
Definition: ffmpeg.c:2045
#define FFSWAP(type, a, b)
Definition: common.h:99
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:231
This structure stores compressed data.
Definition: avcodec.h:1578
static const uint8_t bfu_amount_tab3[8]
Definition: atrac1data.h:35
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:959
int num_bfus
number of Block Floating Units
Definition: atrac1.c:62
static int at1_imdct_block(AT1SUCtx *su, AT1Ctx *q)
Definition: atrac1.c:106
void AAC_RENAME() ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
static const uint16_t bfu_start_long[52]
start position of each BFU in the MDCT spectrum for the long mode
Definition: atrac1data.h:51