FFmpeg
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 
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  if (!q->fdsp) {
366  atrac1_decode_end(avctx);
367  return AVERROR(ENOMEM);
368  }
369 
370  q->bands[0] = q->low;
371  q->bands[1] = q->mid;
372  q->bands[2] = q->high;
373 
374  /* Prepare the mdct overlap buffers */
375  q->SUs[0].spectrum[0] = q->SUs[0].spec1;
376  q->SUs[0].spectrum[1] = q->SUs[0].spec2;
377  q->SUs[1].spectrum[0] = q->SUs[1].spec1;
378  q->SUs[1].spectrum[1] = q->SUs[1].spec2;
379 
380  return 0;
381 }
382 
383 
385  .name = "atrac1",
386  .long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
387  .type = AVMEDIA_TYPE_AUDIO,
388  .id = AV_CODEC_ID_ATRAC1,
389  .priv_data_size = sizeof(AT1Ctx),
391  .close = atrac1_decode_end,
393  .capabilities = AV_CODEC_CAP_DR1,
394  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
396 };
AVCodec
AVCodec.
Definition: codec.h:190
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
mdct_long_nbits
static const uint8_t mdct_long_nbits[3]
Definition: atrac1.c:88
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AT1_MAX_BFU
#define AT1_MAX_BFU
max number of block floating units in a sound unit
Definition: atrac1.c:45
ff_init_ff_sine_windows
void AAC_RENAME() ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
Definition: sinewin_tablegen.h:76
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
AT1_SU_SAMPLES
#define AT1_SU_SAMPLES
number of samples in a sound unit
Definition: atrac1.c:47
bfu_bands_t
static const uint8_t bfu_bands_t[4]
number of BFUs in each QMF band
Definition: atrac1data.h:38
out
FILE * out
Definition: movenc.c:54
FFSWAP
#define FFSWAP(type, a, b)
Definition: common.h:99
atrac1data.h
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:716
AT1_QMF_BANDS
#define AT1_QMF_BANDS
Definition: atrac1.c:52
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
AT1Ctx::spec
float spec[AT1_SU_SAMPLES]
the mdct spectrum buffer
Definition: atrac1.c:76
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:355
data
const char data[16]
Definition: mxf.c:91
AT1SUCtx::num_bfus
int num_bfus
number of Block Floating Units
Definition: atrac1.c:62
at1_imdct
static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec)
Definition: atrac1.c:91
ff_mdct_init
#define ff_mdct_init
Definition: fft.h:169
AT1Ctx
The atrac1 context, holds all needed parameters for decoding.
Definition: atrac1.c:74
init_get_bits
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
AT1SUCtx::spec2
float spec2[AT1_SU_SAMPLES]
mdct buffer
Definition: atrac1.c:65
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
AT1SUCtx
Sound unit struct, one unit is used per channel.
Definition: atrac1.c:60
IDX_HIGH_BAND
#define IDX_HIGH_BAND
Definition: atrac1.c:55
at1_imdct_block
static int at1_imdct_block(AT1SUCtx *su, AT1Ctx *q)
Definition: atrac1.c:106
AT1Ctx::fdsp
AVFloatDSPContext * fdsp
Definition: atrac1.c:83
su
#define su(width, name)
Definition: cbs_av1.c:558
GetBitContext
Definition: get_bits.h:61
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:606
AT1SUCtx::fst_qmf_delay
float fst_qmf_delay[46]
delay line for the 1st stacked QMF filter
Definition: atrac1.c:66
bfu_start_long
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
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold
#define av_cold
Definition: attributes.h:90
atrac1_decode_frame
static int atrac1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: atrac1.c:272
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
atrac.h
bfu_start_short
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
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AT1_SU_MAX_BITS
#define AT1_SU_MAX_BITS
Definition: atrac1.c:49
get_sbits
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:359
channels
channels
Definition: aptx.h:33
get_bits.h
at1_unpack_dequant
static int at1_unpack_dequant(GetBitContext *gb, AT1SUCtx *su, float spec[AT1_SU_SAMPLES])
Definition: atrac1.c:190
bfu_amount_tab1
static const uint8_t bfu_amount_tab1[8]
Definition: atrac1data.h:33
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
AT1Ctx::SUs
AT1SUCtx SUs[AT1_MAX_CHANNELS]
channel sound unit
Definition: atrac1.c:75
ff_atrac_sf_table
float ff_atrac_sf_table[64]
Definition: atrac.c:36
AT1_MAX_CHANNELS
#define AT1_MAX_CHANNELS
Definition: atrac1.c:50
float_dsp.h
FFTContext::imdct_half
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:108
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1854
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
AVPacket::size
int size
Definition: packet.h:356
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:188
AT1Ctx::low
float low[256]
Definition: atrac1.c:78
at1_parse_bsm
static int at1_parse_bsm(GetBitContext *gb, int log2_block_cnt[AT1_QMF_BANDS])
Parse the block size mode byte.
Definition: atrac1.c:167
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
ff_mdct_end
#define ff_mdct_end
Definition: fft.h:170
AT1SUCtx::spec1
float spec1[AT1_SU_SAMPLES]
mdct buffer
Definition: atrac1.c:64
bfu_amount_tab2
static const uint8_t bfu_amount_tab2[4]
Definition: atrac1data.h:34
AVFloatDSPContext
Definition: float_dsp.h:24
AT1SUCtx::log2_block_count
int log2_block_count[AT1_QMF_BANDS]
log2 number of blocks in a band
Definition: atrac1.c:61
AT1SUCtx::spectrum
float * spectrum[2]
Definition: atrac1.c:63
sinewin.h
ff_atrac_generate_tables
av_cold void ff_atrac_generate_tables(void)
Generate common tables.
Definition: atrac.c:48
AT1Ctx::high
float high[512]
Definition: atrac1.c:80
specs_per_bfu
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
bfu_amount_tab3
static const uint8_t bfu_amount_tab3[8]
Definition: atrac1data.h:35
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
DECLARE_ALIGNED
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:112
AV_CODEC_ID_ATRAC1
@ AV_CODEC_ID_ATRAC1
Definition: codec_id.h:456
FFTContext
Definition: fft.h:88
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
AT1SUCtx::snd_qmf_delay
float snd_qmf_delay[46]
delay line for the 2nd stacked QMF filter
Definition: atrac1.c:67
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t
uint8_t
Definition: audio_convert.c:194
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:197
atrac1_decode_init
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
Definition: atrac1.c:333
avcodec.h
atrac1_decode_end
static av_cold int atrac1_decode_end(AVCodecContext *avctx)
Definition: atrac1.c:319
ret
ret
Definition: filter_design.txt:187
AVCodecContext::block_align
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1223
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
AT1SUCtx::last_qmf_delay
float last_qmf_delay[256+39]
delay line for the last stacked QMF filter
Definition: atrac1.c:68
AT1Ctx::mid
float mid[256]
Definition: atrac1.c:79
pos
unsigned int pos
Definition: spdifenc.c:412
ff_atrac1_decoder
AVCodec ff_atrac1_decoder
Definition: atrac1.c:384
fft.h
AVCodecContext
main external API structure.
Definition: avcodec.h:526
temp
else temp
Definition: vf_mcdeint.c:256
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
AVFloatDSPContext::vector_fmul_window
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:119
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
AVPacket
This structure stores compressed data.
Definition: packet.h:332
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
AT1Ctx::mdct_ctx
FFTContext mdct_ctx[3]
Definition: atrac1.c:82
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
at1_subband_synthesis
static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx *su, float *pOut)
Definition: atrac1.c:255
AT1Ctx::bands
float * bands[3]
Definition: atrac1.c:81
ff_atrac_iqmf
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
samples_per_band
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