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binkaudio.c
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1 /*
2  * Bink Audio decoder
3  * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
4  * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
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  * Bink Audio decoder
26  *
27  * Technical details here:
28  * http://wiki.multimedia.cx/index.php?title=Bink_Audio
29  */
30 
32 #include "avcodec.h"
33 #define BITSTREAM_READER_LE
34 #include "get_bits.h"
35 #include "dct.h"
36 #include "rdft.h"
37 #include "internal.h"
38 #include "wma_freqs.h"
39 #include "libavutil/intfloat.h"
40 
41 static float quant_table[96];
42 
43 #define MAX_CHANNELS 2
44 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
45 
46 typedef struct BinkAudioContext {
48  int version_b; ///< Bink version 'b'
49  int first;
50  int channels;
51  int frame_len; ///< transform size (samples)
52  int overlap_len; ///< overlap size (samples)
54  int num_bands;
55  unsigned int *bands;
56  float root;
58  float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
60  union {
63  } trans;
65 
66 
68 {
69  BinkAudioContext *s = avctx->priv_data;
70  int sample_rate = avctx->sample_rate;
71  int sample_rate_half;
72  int i;
73  int frame_len_bits;
74 
75  /* determine frame length */
76  if (avctx->sample_rate < 22050) {
77  frame_len_bits = 9;
78  } else if (avctx->sample_rate < 44100) {
79  frame_len_bits = 10;
80  } else {
81  frame_len_bits = 11;
82  }
83 
84  if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
85  av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
86  return AVERROR_INVALIDDATA;
87  }
88  avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
90 
91  s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
92 
93  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
94  // audio is already interleaved for the RDFT format variant
96  if (sample_rate > INT_MAX / avctx->channels)
97  return AVERROR_INVALIDDATA;
98  sample_rate *= avctx->channels;
99  s->channels = 1;
100  if (!s->version_b)
101  frame_len_bits += av_log2(avctx->channels);
102  } else {
103  s->channels = avctx->channels;
105  }
106 
107  s->frame_len = 1 << frame_len_bits;
108  s->overlap_len = s->frame_len / 16;
109  s->block_size = (s->frame_len - s->overlap_len) * s->channels;
110  sample_rate_half = (sample_rate + 1LL) / 2;
111  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
112  s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
113  else
114  s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
115  for (i = 0; i < 96; i++) {
116  /* constant is result of 0.066399999/log10(M_E) */
117  quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
118  }
119 
120  /* calculate number of bands */
121  for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
122  if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
123  break;
124 
125  s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
126  if (!s->bands)
127  return AVERROR(ENOMEM);
128 
129  /* populate bands data */
130  s->bands[0] = 2;
131  for (i = 1; i < s->num_bands; i++)
132  s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
133  s->bands[s->num_bands] = s->frame_len;
134 
135  s->first = 1;
136 
138  ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
140  ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
141  else
142  return -1;
143 
144  return 0;
145 }
146 
147 static float get_float(GetBitContext *gb)
148 {
149  int power = get_bits(gb, 5);
150  float f = ldexpf(get_bits_long(gb, 23), power - 23);
151  if (get_bits1(gb))
152  f = -f;
153  return f;
154 }
155 
156 static const uint8_t rle_length_tab[16] = {
157  2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
158 };
159 
160 /**
161  * Decode Bink Audio block
162  * @param[out] out Output buffer (must contain s->block_size elements)
163  * @return 0 on success, negative error code on failure
164  */
165 static int decode_block(BinkAudioContext *s, float **out, int use_dct)
166 {
167  int ch, i, j, k;
168  float q, quant[25];
169  int width, coeff;
170  GetBitContext *gb = &s->gb;
171 
172  if (use_dct)
173  skip_bits(gb, 2);
174 
175  for (ch = 0; ch < s->channels; ch++) {
176  FFTSample *coeffs = out[ch];
177 
178  if (s->version_b) {
179  if (get_bits_left(gb) < 64)
180  return AVERROR_INVALIDDATA;
181  coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
182  coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
183  } else {
184  if (get_bits_left(gb) < 58)
185  return AVERROR_INVALIDDATA;
186  coeffs[0] = get_float(gb) * s->root;
187  coeffs[1] = get_float(gb) * s->root;
188  }
189 
190  if (get_bits_left(gb) < s->num_bands * 8)
191  return AVERROR_INVALIDDATA;
192  for (i = 0; i < s->num_bands; i++) {
193  int value = get_bits(gb, 8);
194  quant[i] = quant_table[FFMIN(value, 95)];
195  }
196 
197  k = 0;
198  q = quant[0];
199 
200  // parse coefficients
201  i = 2;
202  while (i < s->frame_len) {
203  if (s->version_b) {
204  j = i + 16;
205  } else {
206  int v = get_bits1(gb);
207  if (v) {
208  v = get_bits(gb, 4);
209  j = i + rle_length_tab[v] * 8;
210  } else {
211  j = i + 8;
212  }
213  }
214 
215  j = FFMIN(j, s->frame_len);
216 
217  width = get_bits(gb, 4);
218  if (width == 0) {
219  memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
220  i = j;
221  while (s->bands[k] < i)
222  q = quant[k++];
223  } else {
224  while (i < j) {
225  if (s->bands[k] == i)
226  q = quant[k++];
227  coeff = get_bits(gb, width);
228  if (coeff) {
229  int v;
230  v = get_bits1(gb);
231  if (v)
232  coeffs[i] = -q * coeff;
233  else
234  coeffs[i] = q * coeff;
235  } else {
236  coeffs[i] = 0.0f;
237  }
238  i++;
239  }
240  }
241  }
242 
243  if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
244  coeffs[0] /= 0.5;
245  s->trans.dct.dct_calc(&s->trans.dct, coeffs);
246  }
248  s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
249  }
250 
251  for (ch = 0; ch < s->channels; ch++) {
252  int j;
253  int count = s->overlap_len * s->channels;
254  if (!s->first) {
255  j = ch;
256  for (i = 0; i < s->overlap_len; i++, j += s->channels)
257  out[ch][i] = (s->previous[ch][i] * (count - j) +
258  out[ch][i] * j) / count;
259  }
260  memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
261  s->overlap_len * sizeof(*s->previous[ch]));
262  }
263 
264  s->first = 0;
265 
266  return 0;
267 }
268 
270 {
271  BinkAudioContext * s = avctx->priv_data;
272  av_freep(&s->bands);
273  av_freep(&s->packet_buffer);
275  ff_rdft_end(&s->trans.rdft);
277  ff_dct_end(&s->trans.dct);
278 
279  return 0;
280 }
281 
283 {
284  int n = (-get_bits_count(s)) & 31;
285  if (n) skip_bits(s, n);
286 }
287 
288 static int decode_frame(AVCodecContext *avctx, void *data,
289  int *got_frame_ptr, AVPacket *avpkt)
290 {
291  BinkAudioContext *s = avctx->priv_data;
292  AVFrame *frame = data;
293  GetBitContext *gb = &s->gb;
294  int ret, consumed = 0;
295 
296  if (!get_bits_left(gb)) {
297  uint8_t *buf;
298  /* handle end-of-stream */
299  if (!avpkt->size) {
300  *got_frame_ptr = 0;
301  return 0;
302  }
303  if (avpkt->size < 4) {
304  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
305  return AVERROR_INVALIDDATA;
306  }
308  if (!buf)
309  return AVERROR(ENOMEM);
310  memset(buf + avpkt->size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
311  s->packet_buffer = buf;
312  memcpy(s->packet_buffer, avpkt->data, avpkt->size);
313  if ((ret = init_get_bits8(gb, s->packet_buffer, avpkt->size)) < 0)
314  return ret;
315  consumed = avpkt->size;
316 
317  /* skip reported size */
318  skip_bits_long(gb, 32);
319  }
320 
321  /* get output buffer */
322  frame->nb_samples = s->frame_len;
323  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
324  return ret;
325 
326  if (decode_block(s, (float **)frame->extended_data,
327  avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
328  av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
329  return AVERROR_INVALIDDATA;
330  }
331  get_bits_align32(gb);
332 
333  frame->nb_samples = s->block_size / avctx->channels;
334  *got_frame_ptr = 1;
335 
336  return consumed;
337 }
338 
340  .name = "binkaudio_rdft",
341  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
342  .type = AVMEDIA_TYPE_AUDIO,
344  .priv_data_size = sizeof(BinkAudioContext),
345  .init = decode_init,
346  .close = decode_end,
347  .decode = decode_frame,
348  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
349 };
350 
352  .name = "binkaudio_dct",
353  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
354  .type = AVMEDIA_TYPE_AUDIO,
356  .priv_data_size = sizeof(BinkAudioContext),
357  .init = decode_init,
358  .close = decode_end,
359  .decode = decode_frame,
360  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
361 };
union BinkAudioContext::@38 trans
av_cold void ff_rdft_end(RDFTContext *s)
Definition: rdft.c:132
float, planar
Definition: samplefmt.h:70
const struct AVCodec * codec
Definition: avcodec.h:1521
float v
static float get_float(GetBitContext *gb)
Definition: binkaudio.c:147
const char * s
Definition: avisynth_c.h:631
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define MAX_CHANNELS
Definition: binkaudio.c:43
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
Definition: avfft.h:75
void(* dct_calc)(struct DCTContext *s, FFTSample *data)
Definition: dct.h:37
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
static av_cold int decode_end(AVCodecContext *avctx)
Definition: binkaudio.c:269
Definition: avfft.h:95
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:218
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static av_always_inline float av_int2float(uint32_t i)
Reinterpret a 32-bit integer as a float.
Definition: intfloat.h:40
static const uint8_t rle_length_tab[16]
Definition: binkaudio.c:156
int size
Definition: avcodec.h:1434
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:53
uint8_t * packet_buffer
Definition: binkaudio.c:59
const uint16_t ff_wma_critical_freqs[25]
Definition: wma_freqs.c:23
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:3482
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:882
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2280
uint8_t
#define av_cold
Definition: attributes.h:74
#define av_malloc(s)
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1627
unsigned int * bands
Definition: binkaudio.c:55
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1433
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:213
bitstream reader API header.
float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE/16]
coeffs from previous audio block
Definition: binkaudio.c:58
#define av_log(a,...)
#define expf(x)
Definition: libm.h:72
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:594
enum AVCodecID id
Definition: avcodec.h:3496
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define BINK_BLOCK_MAX_SIZE
Definition: binkaudio.c:44
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:178
static void get_bits_align32(GetBitContext *s)
Definition: binkaudio.c:282
static int decode_block(BinkAudioContext *s, float **out, int use_dct)
Decode Bink Audio block.
Definition: binkaudio.c:165
const char * name
Name of the codec implementation.
Definition: avcodec.h:3489
GLsizei count
Definition: opengl_enc.c:109
float FFTSample
Definition: avfft.h:35
Libavcodec external API header.
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2333
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
Definition: rdft.h:60
GetBitContext gb
Definition: binkaudio.c:47
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:92
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
static float quant_table[96]
Definition: binkaudio.c:41
DCTContext dct
Definition: binkaudio.c:62
Definition: dct.h:31
static av_cold int decode_init(AVCodecContext *avctx)
Definition: binkaudio.c:67
int n
Definition: avisynth_c.h:547
AVCodec ff_binkaudio_rdft_decoder
Definition: binkaudio.c:339
#define av_log2
Definition: intmath.h:100
int overlap_len
overlap size (samples)
Definition: binkaudio.c:52
sample_rate
#define CONFIG_BINKAUDIO_DCT_DECODER
Definition: config.h:862
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: binkaudio.c:288
int sample_rate
samples per second
Definition: avcodec.h:2272
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:446
AVCodec ff_binkaudio_dct_decoder
Definition: binkaudio.c:351
main external API structure.
Definition: avcodec.h:1512
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:1048
#define ldexpf(x, exp)
Definition: libm.h:107
void * buf
Definition: avisynth_c.h:553
int extradata_size
Definition: avcodec.h:1628
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:305
#define CONFIG_BINKAUDIO_RDFT_DECODER
Definition: config.h:863
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:298
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
Set up DCT.
Definition: dct.c:177
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:338
const uint8_t * quant
int frame_len
transform size (samples)
Definition: binkaudio.c:51
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:521
int version_b
Bink version 'b'.
Definition: binkaudio.c:48
common internal api header.
void * av_realloc(void *ptr, size_t size)
Allocate or reallocate a block of memory.
Definition: mem.c:145
RDFTContext rdft
Definition: binkaudio.c:61
FFTSample coeffs[BINK_BLOCK_MAX_SIZE]
Definition: binkaudio.c:57
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:636
void * priv_data
Definition: avcodec.h:1554
static const int16_t coeffs[]
int channels
number of audio channels
Definition: avcodec.h:2273
static const double coeff[2][5]
Definition: vf_owdenoise.c:71
av_cold void ff_dct_end(DCTContext *s)
Definition: dct.c:220
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
#define av_freep(p)
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:215
#define AV_CH_LAYOUT_MONO
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
Definition: rdft.c:99
This structure stores compressed data.
Definition: avcodec.h:1410
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:857
static int width