FFmpeg  4.4.4
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 "libavutil/mem_internal.h"
37 
38 #include "avcodec.h"
39 #include "get_bits.h"
40 #include "fft.h"
41 #include "internal.h"
42 #include "sinewin.h"
43 
44 #include "atrac.h"
45 #include "atrac1data.h"
46 
47 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
48 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
49 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
50 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
51 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
52 #define AT1_MAX_CHANNELS 2
53 
54 #define AT1_QMF_BANDS 3
55 #define IDX_LOW_BAND 0
56 #define IDX_MID_BAND 1
57 #define IDX_HIGH_BAND 2
58 
59 /**
60  * Sound unit struct, one unit is used per channel
61  */
62 typedef struct AT1SUCtx {
63  int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
64  int num_bfus; ///< number of Block Floating Units
65  float* spectrum[2];
66  DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
67  DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
68  DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
69  DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
70  DECLARE_ALIGNED(32, float, last_qmf_delay)[256+39]; ///< delay line for the last stacked QMF filter
71 } AT1SUCtx;
72 
73 /**
74  * The atrac1 context, holds all needed parameters for decoding
75  */
76 typedef struct AT1Ctx {
77  AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
78  DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
79 
80  DECLARE_ALIGNED(32, float, low)[256];
81  DECLARE_ALIGNED(32, float, mid)[256];
82  DECLARE_ALIGNED(32, float, high)[512];
83  float* bands[3];
85  void (*vector_fmul_window)(float *dst, const float *src0,
86  const float *src1, const float *win, int len);
87 } AT1Ctx;
88 
89 /** size of the transform in samples in the long mode for each QMF band */
90 static const uint16_t samples_per_band[3] = {128, 128, 256};
91 static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
92 
93 
94 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
95  int rev_spec)
96 {
97  FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
98  int transf_size = 1 << nbits;
99 
100  if (rev_spec) {
101  int i;
102  for (i = 0; i < transf_size / 2; i++)
103  FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
104  }
105  mdct_context->imdct_half(mdct_context, out, spec);
106 }
107 
108 
110 {
111  int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
112  unsigned int start_pos, ref_pos = 0, pos = 0;
113 
114  for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
115  float *prev_buf;
116  int j;
117 
118  band_samples = samples_per_band[band_num];
119  log2_block_count = su->log2_block_count[band_num];
120 
121  /* number of mdct blocks in the current QMF band: 1 - for long mode */
122  /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
123  num_blocks = 1 << log2_block_count;
124 
125  if (num_blocks == 1) {
126  /* mdct block size in samples: 128 (long mode, low & mid bands), */
127  /* 256 (long mode, high band) and 32 (short mode, all bands) */
128  block_size = band_samples >> log2_block_count;
129 
130  /* calc transform size in bits according to the block_size_mode */
131  nbits = mdct_long_nbits[band_num] - log2_block_count;
132 
133  if (nbits != 5 && nbits != 7 && nbits != 8)
134  return AVERROR_INVALIDDATA;
135  } else {
136  block_size = 32;
137  nbits = 5;
138  }
139 
140  start_pos = 0;
141  prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
142  for (j=0; j < num_blocks; j++) {
143  at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
144 
145  /* overlap and window */
146  q->vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
147  &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
148 
149  prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
150  start_pos += block_size;
151  pos += block_size;
152  }
153 
154  if (num_blocks == 1)
155  memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
156 
157  ref_pos += band_samples;
158  }
159 
160  /* Swap buffers so the mdct overlap works */
161  FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
162 
163  return 0;
164 }
165 
166 /**
167  * Parse the block size mode byte
168  */
169 
170 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
171 {
172  int log2_block_count_tmp, i;
173 
174  for (i = 0; i < 2; i++) {
175  /* low and mid band */
176  log2_block_count_tmp = get_bits(gb, 2);
177  if (log2_block_count_tmp & 1)
178  return AVERROR_INVALIDDATA;
179  log2_block_cnt[i] = 2 - log2_block_count_tmp;
180  }
181 
182  /* high band */
183  log2_block_count_tmp = get_bits(gb, 2);
184  if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
185  return AVERROR_INVALIDDATA;
186  log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
187 
188  skip_bits(gb, 2);
189  return 0;
190 }
191 
192 
194  float spec[AT1_SU_SAMPLES])
195 {
196  int bits_used, band_num, bfu_num, i;
197  uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
198  uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
199 
200  /* parse the info byte (2nd byte) telling how much BFUs were coded */
201  su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
202 
203  /* calc number of consumed bits:
204  num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
205  + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
206  bits_used = su->num_bfus * 10 + 32 +
207  bfu_amount_tab2[get_bits(gb, 2)] +
208  (bfu_amount_tab3[get_bits(gb, 3)] << 1);
209 
210  /* get word length index (idwl) for each BFU */
211  for (i = 0; i < su->num_bfus; i++)
212  idwls[i] = get_bits(gb, 4);
213 
214  /* get scalefactor index (idsf) for each BFU */
215  for (i = 0; i < su->num_bfus; i++)
216  idsfs[i] = get_bits(gb, 6);
217 
218  /* zero idwl/idsf for empty BFUs */
219  for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
220  idwls[i] = idsfs[i] = 0;
221 
222  /* read in the spectral data and reconstruct MDCT spectrum of this channel */
223  for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
224  for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
225  int pos;
226 
227  int num_specs = specs_per_bfu[bfu_num];
228  int word_len = !!idwls[bfu_num] + idwls[bfu_num];
229  float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]];
230  bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
231 
232  /* check for bitstream overflow */
233  if (bits_used > AT1_SU_MAX_BITS)
234  return AVERROR_INVALIDDATA;
235 
236  /* get the position of the 1st spec according to the block size mode */
237  pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
238 
239  if (word_len) {
240  float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
241 
242  for (i = 0; i < num_specs; i++) {
243  /* read in a quantized spec and convert it to
244  * signed int and then inverse quantization
245  */
246  spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
247  }
248  } else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */
249  memset(&spec[pos], 0, num_specs * sizeof(float));
250  }
251  }
252  }
253 
254  return 0;
255 }
256 
257 
258 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
259 {
260  float temp[256];
261  float iqmf_temp[512 + 46];
262 
263  /* combine low and middle bands */
264  ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
265 
266  /* delay the signal of the high band by 39 samples */
267  memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 39);
268  memcpy(&su->last_qmf_delay[39], q->bands[2], sizeof(float) * 256);
269 
270  /* combine (low + middle) and high bands */
271  ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
272 }
273 
274 
275 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
276  int *got_frame_ptr, AVPacket *avpkt)
277 {
278  AVFrame *frame = data;
279  const uint8_t *buf = avpkt->data;
280  int buf_size = avpkt->size;
281  AT1Ctx *q = avctx->priv_data;
282  int ch, ret;
283  GetBitContext gb;
284 
285 
286  if (buf_size < 212 * avctx->channels) {
287  av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
288  return AVERROR_INVALIDDATA;
289  }
290 
291  /* get output buffer */
293  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
294  return ret;
295 
296  for (ch = 0; ch < avctx->channels; ch++) {
297  AT1SUCtx* su = &q->SUs[ch];
298 
299  init_get_bits(&gb, &buf[212 * ch], 212 * 8);
300 
301  /* parse block_size_mode, 1st byte */
302  ret = at1_parse_bsm(&gb, su->log2_block_count);
303  if (ret < 0)
304  return ret;
305 
306  ret = at1_unpack_dequant(&gb, su, q->spec);
307  if (ret < 0)
308  return ret;
309 
310  ret = at1_imdct_block(su, q);
311  if (ret < 0)
312  return ret;
313  at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]);
314  }
315 
316  *got_frame_ptr = 1;
317 
318  return avctx->block_align;
319 }
320 
321 
323 {
324  AT1Ctx *q = avctx->priv_data;
325 
326  ff_mdct_end(&q->mdct_ctx[0]);
327  ff_mdct_end(&q->mdct_ctx[1]);
328  ff_mdct_end(&q->mdct_ctx[2]);
329 
330  return 0;
331 }
332 
333 
335 {
336  AT1Ctx *q = avctx->priv_data;
337  AVFloatDSPContext *fdsp;
338  int ret;
339 
341 
342  if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
343  av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
344  avctx->channels);
345  return AVERROR(EINVAL);
346  }
347 
348  if (avctx->block_align <= 0) {
349  av_log(avctx, AV_LOG_ERROR, "Unsupported block align.");
350  return AVERROR_PATCHWELCOME;
351  }
352 
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  return ret;
359  }
360 
362 
364 
366  if (!fdsp)
367  return AVERROR(ENOMEM);
369  av_free(fdsp);
370 
371  q->bands[0] = q->low;
372  q->bands[1] = q->mid;
373  q->bands[2] = q->high;
374 
375  /* Prepare the mdct overlap buffers */
376  q->SUs[0].spectrum[0] = q->SUs[0].spec1;
377  q->SUs[0].spectrum[1] = q->SUs[0].spec2;
378  q->SUs[1].spectrum[0] = q->SUs[1].spec1;
379  q->SUs[1].spectrum[1] = q->SUs[1].spec2;
380 
381  return 0;
382 }
383 
384 
386  .name = "atrac1",
387  .long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
388  .type = AVMEDIA_TYPE_AUDIO,
389  .id = AV_CODEC_ID_ATRAC1,
390  .priv_data_size = sizeof(AT1Ctx),
392  .close = atrac1_decode_end,
394  .capabilities = AV_CODEC_CAP_DR1,
395  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
398 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
static float win(SuperEqualizerContext *s, float n, int N)
channels
Definition: aptx.h:33
static av_cold int atrac1_decode_end(AVCodecContext *avctx)
Definition: atrac1.c:322
static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec)
Definition: atrac1.c:94
#define AT1_MAX_CHANNELS
Definition: atrac1.c:52
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:90
static int atrac1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: atrac1.c:275
static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx *su, float *pOut)
Definition: atrac1.c:258
static int at1_imdct_block(AT1SUCtx *su, AT1Ctx *q)
Definition: atrac1.c:109
static int at1_parse_bsm(GetBitContext *gb, int log2_block_cnt[AT1_QMF_BANDS])
Parse the block size mode byte.
Definition: atrac1.c:170
static int at1_unpack_dequant(GetBitContext *gb, AT1SUCtx *su, float spec[AT1_SU_SAMPLES])
Definition: atrac1.c:193
static const uint8_t mdct_long_nbits[3]
Definition: atrac1.c:91
#define IDX_HIGH_BAND
Definition: atrac1.c:57
#define AT1_MAX_BFU
max number of block floating units in a sound unit
Definition: atrac1.c:47
#define AT1_SU_MAX_BITS
Definition: atrac1.c:51
AVCodec ff_atrac1_decoder
Definition: atrac1.c:385
#define AT1_QMF_BANDS
Definition: atrac1.c:54
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
Definition: atrac1.c:334
#define AT1_SU_SAMPLES
number of samples in a sound unit
Definition: atrac1.c:49
ATRAC1 compatible decoder data.
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_tab2[4]
Definition: atrac1data.h:34
static const uint8_t bfu_amount_tab1[8]
Definition: atrac1data.h:33
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
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
static const uint8_t bfu_amount_tab3[8]
Definition: atrac1data.h:35
static const uint8_t bfu_bands_t[4]
number of BFUs in each QMF band
Definition: atrac1data.h:38
float ff_atrac_sf_table[64]
Definition: atrac.c:38
av_cold void ff_atrac_generate_tables(void)
Generate common tables.
Definition: atrac.c:63
void ff_atrac_iqmf(float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
Quadrature mirror synthesis filter.
Definition: atrac.c:130
ATRAC common header.
#define av_cold
Definition: attributes.h:88
uint8_t
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
#define su(width, name)
Definition: cbs_av1.c:554
#define FFSWAP(type, a, b)
Definition: common.h:108
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static AVFrame * frame
#define ff_mdct_init
Definition: fft.h:161
#define ff_mdct_end
Definition: fft.h:162
bitstream reader API header.
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:359
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
@ AV_CODEC_ID_ATRAC1
Definition: codec_id.h:470
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:117
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
int i
Definition: input.c:407
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:41
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:49
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
const char data[16]
Definition: mxf.c:142
typedef void(RENAME(mix_any_func_type))
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
unsigned int pos
Definition: spdifenc.c:412
The atrac1 context, holds all needed parameters for decoding.
Definition: atrac1.c:76
AT1SUCtx SUs[AT1_MAX_CHANNELS]
channel sound unit
Definition: atrac1.c:77
FFTContext mdct_ctx[3]
Definition: atrac1.c:84
float * bands[3]
Definition: atrac1.c:83
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Definition: atrac1.c:85
float spec[AT1_SU_SAMPLES]
the mdct spectrum buffer
Definition: atrac1.c:78
float mid[256]
Definition: atrac1.c:81
float low[256]
Definition: atrac1.c:80
float high[512]
Definition: atrac1.c:82
Sound unit struct, one unit is used per channel.
Definition: atrac1.c:62
float fst_qmf_delay[46]
delay line for the 1st stacked QMF filter
Definition: atrac1.c:68
float spec1[AT1_SU_SAMPLES]
mdct buffer
Definition: atrac1.c:66
int num_bfus
number of Block Floating Units
Definition: atrac1.c:64
float snd_qmf_delay[46]
delay line for the 2nd stacked QMF filter
Definition: atrac1.c:69
float spec2[AT1_SU_SAMPLES]
mdct buffer
Definition: atrac1.c:67
int log2_block_count[AT1_QMF_BANDS]
log2 number of blocks in a band
Definition: atrac1.c:63
float last_qmf_delay[256+39]
delay line for the last stacked QMF filter
Definition: atrac1.c:70
float * spectrum[2]
Definition: atrac1.c:65
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:616
int channels
number of audio channels
Definition: avcodec.h:1197
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1233
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
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
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
Definition: fft.h:83
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:103
#define av_free(p)
#define av_log(a,...)
#define src1
Definition: h264pred.c:140
#define src0
Definition: h264pred.c:139
FILE * out
Definition: movenc.c:54
else temp
Definition: vf_mcdeint.c:259
int len