FFmpeg
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 
31 #include "config_components.h"
32 
34 #include "libavutil/intfloat.h"
35 #include "libavutil/mem_internal.h"
36 
37 #define BITSTREAM_READER_LE
38 #include "avcodec.h"
39 #include "dct.h"
40 #include "decode.h"
41 #include "get_bits.h"
42 #include "codec_internal.h"
43 #include "internal.h"
44 #include "rdft.h"
45 #include "wma_freqs.h"
46 
47 #define MAX_DCT_CHANNELS 6
48 #define MAX_CHANNELS 2
49 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
50 
51 typedef struct BinkAudioContext {
53  int version_b; ///< Bink version 'b'
54  int first;
55  int channels;
56  int ch_offset;
57  int frame_len; ///< transform size (samples)
58  int overlap_len; ///< overlap size (samples)
60  int num_bands;
61  float root;
62  unsigned int bands[26];
63  float previous[MAX_DCT_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
64  float quant_table[96];
66  union {
69  } trans;
71 
72 
74 {
75  BinkAudioContext *s = avctx->priv_data;
76  int sample_rate = avctx->sample_rate;
77  int sample_rate_half;
78  int i, ret;
79  int frame_len_bits;
80  int max_channels = avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT ? MAX_CHANNELS : MAX_DCT_CHANNELS;
81  int channels = avctx->ch_layout.nb_channels;
82 
83  /* determine frame length */
84  if (avctx->sample_rate < 22050) {
85  frame_len_bits = 9;
86  } else if (avctx->sample_rate < 44100) {
87  frame_len_bits = 10;
88  } else {
89  frame_len_bits = 11;
90  }
91 
92  if (channels < 1 || channels > max_channels) {
93  av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", channels);
94  return AVERROR_INVALIDDATA;
95  }
98 
99  s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
100 
101  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
102  // audio is already interleaved for the RDFT format variant
103  avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
104  if (sample_rate > INT_MAX / channels)
105  return AVERROR_INVALIDDATA;
107  s->channels = 1;
108  if (!s->version_b)
109  frame_len_bits += av_log2(channels);
110  } else {
111  s->channels = channels;
113  }
114 
115  s->frame_len = 1 << frame_len_bits;
116  s->overlap_len = s->frame_len / 16;
117  s->block_size = (s->frame_len - s->overlap_len) * FFMIN(MAX_CHANNELS, s->channels);
118  sample_rate_half = (sample_rate + 1LL) / 2;
119  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
120  s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
121  else
122  s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
123  for (i = 0; i < 96; i++) {
124  /* constant is result of 0.066399999/log10(M_E) */
125  s->quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
126  }
127 
128  /* calculate number of bands */
129  for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
130  if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
131  break;
132 
133  /* populate bands data */
134  s->bands[0] = 2;
135  for (i = 1; i < s->num_bands; i++)
136  s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
137  s->bands[s->num_bands] = s->frame_len;
138 
139  s->first = 1;
140 
141  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
142  ret = ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
143  else if (CONFIG_BINKAUDIO_DCT_DECODER)
144  ret = ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
145  else
146  av_assert0(0);
147  if (ret < 0)
148  return ret;
149 
150  s->pkt = avctx->internal->in_pkt;
151 
152  return 0;
153 }
154 
155 static float get_float(GetBitContext *gb)
156 {
157  int power = get_bits(gb, 5);
158  float f = ldexpf(get_bits(gb, 23), power - 23);
159  if (get_bits1(gb))
160  f = -f;
161  return f;
162 }
163 
164 static const uint8_t rle_length_tab[16] = {
165  2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
166 };
167 
168 /**
169  * Decode Bink Audio block
170  * @param[out] out Output buffer (must contain s->block_size elements)
171  * @return 0 on success, negative error code on failure
172  */
173 static int decode_block(BinkAudioContext *s, float **out, int use_dct,
174  int channels, int ch_offset)
175 {
176  int ch, i, j, k;
177  float q, quant[25];
178  int width, coeff;
179  GetBitContext *gb = &s->gb;
180 
181  if (use_dct)
182  skip_bits(gb, 2);
183 
184  for (ch = 0; ch < channels; ch++) {
185  FFTSample *coeffs = out[ch + ch_offset];
186 
187  if (s->version_b) {
188  if (get_bits_left(gb) < 64)
189  return AVERROR_INVALIDDATA;
190  coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
191  coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
192  } else {
193  if (get_bits_left(gb) < 58)
194  return AVERROR_INVALIDDATA;
195  coeffs[0] = get_float(gb) * s->root;
196  coeffs[1] = get_float(gb) * s->root;
197  }
198 
199  if (get_bits_left(gb) < s->num_bands * 8)
200  return AVERROR_INVALIDDATA;
201  for (i = 0; i < s->num_bands; i++) {
202  int value = get_bits(gb, 8);
203  quant[i] = s->quant_table[FFMIN(value, 95)];
204  }
205 
206  k = 0;
207  q = quant[0];
208 
209  // parse coefficients
210  i = 2;
211  while (i < s->frame_len) {
212  if (s->version_b) {
213  j = i + 16;
214  } else {
215  int v = get_bits1(gb);
216  if (v) {
217  v = get_bits(gb, 4);
218  j = i + rle_length_tab[v] * 8;
219  } else {
220  j = i + 8;
221  }
222  }
223 
224  j = FFMIN(j, s->frame_len);
225 
226  width = get_bits(gb, 4);
227  if (width == 0) {
228  memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
229  i = j;
230  while (s->bands[k] < i)
231  q = quant[k++];
232  } else {
233  while (i < j) {
234  if (s->bands[k] == i)
235  q = quant[k++];
236  coeff = get_bits(gb, width);
237  if (coeff) {
238  int v;
239  v = get_bits1(gb);
240  if (v)
241  coeffs[i] = -q * coeff;
242  else
243  coeffs[i] = q * coeff;
244  } else {
245  coeffs[i] = 0.0f;
246  }
247  i++;
248  }
249  }
250  }
251 
252  if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
253  coeffs[0] /= 0.5;
254  s->trans.dct.dct_calc(&s->trans.dct, coeffs);
255  }
256  else if (CONFIG_BINKAUDIO_RDFT_DECODER)
257  s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
258  }
259 
260  for (ch = 0; ch < channels; ch++) {
261  int j;
262  int count = s->overlap_len * channels;
263  if (!s->first) {
264  j = ch;
265  for (i = 0; i < s->overlap_len; i++, j += channels)
266  out[ch + ch_offset][i] = (s->previous[ch + ch_offset][i] * (count - j) +
267  out[ch + ch_offset][i] * j) / count;
268  }
269  memcpy(s->previous[ch + ch_offset], &out[ch + ch_offset][s->frame_len - s->overlap_len],
270  s->overlap_len * sizeof(*s->previous[ch + ch_offset]));
271  }
272 
273  s->first = 0;
274 
275  return 0;
276 }
277 
279 {
280  BinkAudioContext * s = avctx->priv_data;
281  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
282  ff_rdft_end(&s->trans.rdft);
283  else if (CONFIG_BINKAUDIO_DCT_DECODER)
284  ff_dct_end(&s->trans.dct);
285 
286  return 0;
287 }
288 
290 {
291  int n = (-get_bits_count(s)) & 31;
292  if (n) skip_bits(s, n);
293 }
294 
296 {
297  BinkAudioContext *s = avctx->priv_data;
298  GetBitContext *gb = &s->gb;
299  int ret;
300 
301 again:
302  if (!s->pkt->data) {
303  ret = ff_decode_get_packet(avctx, s->pkt);
304  if (ret < 0) {
305  s->ch_offset = 0;
306  return ret;
307  }
308 
309  if (s->pkt->size < 4) {
310  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
312  goto fail;
313  }
314 
315  ret = init_get_bits8(gb, s->pkt->data, s->pkt->size);
316  if (ret < 0)
317  goto fail;
318 
319  /* skip reported size */
320  skip_bits_long(gb, 32);
321  }
322 
323  /* get output buffer */
324  if (s->ch_offset == 0) {
325  frame->nb_samples = s->frame_len;
326  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
327  return ret;
328  }
329 
330  if (decode_block(s, (float **)frame->extended_data,
332  FFMIN(MAX_CHANNELS, s->channels - s->ch_offset), s->ch_offset)) {
333  av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
334  s->ch_offset = 0;
335  return AVERROR_INVALIDDATA;
336  }
337  s->ch_offset += MAX_CHANNELS;
338  get_bits_align32(gb);
339  if (!get_bits_left(gb)) {
340  memset(gb, 0, sizeof(*gb));
341  av_packet_unref(s->pkt);
342  }
343  if (s->ch_offset >= s->channels) {
344  s->ch_offset = 0;
345  } else {
346  goto again;
347  }
348 
349  frame->nb_samples = s->block_size / FFMIN(avctx->ch_layout.nb_channels, MAX_CHANNELS);
350 
351  return 0;
352 fail:
353  s->ch_offset = 0;
354  av_packet_unref(s->pkt);
355  return ret;
356 }
357 
358 static void decode_flush(AVCodecContext *avctx)
359 {
360  BinkAudioContext *const s = avctx->priv_data;
361 
362  /* s->pkt coincides with avctx->internal->in_pkt
363  * and is unreferenced generically when flushing. */
364  s->first = 1;
365  s->ch_offset = 0;
366 }
367 
369  .p.name = "binkaudio_rdft",
370  CODEC_LONG_NAME("Bink Audio (RDFT)"),
371  .p.type = AVMEDIA_TYPE_AUDIO,
373  .priv_data_size = sizeof(BinkAudioContext),
374  .init = decode_init,
375  .flush = decode_flush,
376  .close = decode_end,
378  .p.capabilities = AV_CODEC_CAP_DR1,
379  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
380 };
381 
383  .p.name = "binkaudio_dct",
384  CODEC_LONG_NAME("Bink Audio (DCT)"),
385  .p.type = AVMEDIA_TYPE_AUDIO,
387  .priv_data_size = sizeof(BinkAudioContext),
388  .init = decode_init,
389  .flush = decode_flush,
390  .close = decode_end,
392  .p.capabilities = AV_CODEC_CAP_DR1,
393  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
394 };
av_packet_unref
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:422
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
ff_decode_get_packet
int ff_decode_get_packet(AVCodecContext *avctx, AVPacket *pkt)
Called by decoders to get the next packet for decoding.
Definition: decode.c:205
BinkAudioContext::first
int first
Definition: binkaudio.c:54
BinkAudioContext::version_b
int version_b
Bink version 'b'.
Definition: binkaudio.c:53
FF_CODEC_CAP_INIT_CLEANUP
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: codec_internal.h:42
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:839
mem_internal.h
out
FILE * out
Definition: movenc.c:54
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1007
wma_freqs.h
BinkAudioContext::channels
int channels
Definition: binkaudio.c:55
rdft.h
get_bits_long
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:546
get_float
static float get_float(GetBitContext *gb)
Definition: binkaudio.c:155
BINK_BLOCK_MAX_SIZE
#define BINK_BLOCK_MAX_SIZE
Definition: binkaudio.c:49
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
DFT_C2R
@ DFT_C2R
Definition: avfft.h:75
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
internal.h
expf
#define expf(x)
Definition: libm.h:283
FFCodec
Definition: codec_internal.h:119
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:303
intfloat.h
sample_rate
sample_rate
Definition: ffmpeg_filter.c:156
BinkAudioContext::gb
GetBitContext gb
Definition: binkaudio.c:52
init
static int init
Definition: av_tx.c:47
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:123
DCT_III
@ DCT_III
Definition: avfft.h:95
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:407
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2059
fail
#define fail()
Definition: checkasm.h:133
av_int2float
static av_always_inline float av_int2float(uint32_t i)
Reinterpret a 32-bit integer as a float.
Definition: intfloat.h:40
GetBitContext
Definition: get_bits.h:61
ff_rdft_end
av_cold void ff_rdft_end(RDFTContext *s)
Definition: rdft.c:117
quant
static int quant(float coef, const float Q, const float rounding)
Quantize one coefficient.
Definition: aacenc_utils.h:59
BinkAudioContext
Definition: binkaudio.c:51
BinkAudioContext::quant_table
float quant_table[96]
Definition: binkaudio.c:64
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
av_cold
#define av_cold
Definition: attributes.h:90
dct.h
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:667
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:500
width
#define width
s
#define s(width, name)
Definition: cbs_vp9.c:256
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
decode_end
static av_cold int decode_end(AVCodecContext *avctx)
Definition: binkaudio.c:278
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
binkaudio_receive_frame
static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Definition: binkaudio.c:295
AV_CODEC_ID_BINKAUDIO_DCT
@ AV_CODEC_ID_BINKAUDIO_DCT
Definition: codec_id.h:481
channels
channels
Definition: aptx.h:31
decode.h
get_bits.h
ff_wma_critical_freqs
const uint16_t ff_wma_critical_freqs[25]
Definition: wma_freqs.c:23
BinkAudioContext::ch_offset
int ch_offset
Definition: binkaudio.c:56
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:264
ldexpf
#define ldexpf(x, exp)
Definition: libm.h:389
flush
static void flush(AVCodecContext *avctx)
Definition: aacdec_template.c:607
BinkAudioContext::overlap_len
int overlap_len
overlap size (samples)
Definition: binkaudio.c:58
AVCodecContext::internal
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:433
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
MAX_DCT_CHANNELS
#define MAX_DCT_CHANNELS
Definition: binkaudio.c:47
av_channel_layout_default
void av_channel_layout_default(AVChannelLayout *ch_layout, int nb_channels)
Get the default channel layout for a given number of channels.
Definition: channel_layout.c:960
FFTSample
float FFTSample
Definition: avfft.h:35
decode_block
static int decode_block(BinkAudioContext *s, float **out, int use_dct, int channels, int ch_offset)
Decode Bink Audio block.
Definition: binkaudio.c:173
BinkAudioContext::pkt
AVPacket * pkt
Definition: binkaudio.c:65
ff_binkaudio_dct_decoder
const FFCodec ff_binkaudio_dct_decoder
Definition: binkaudio.c:382
av_channel_layout_uninit
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
Definition: channel_layout.c:630
f
f
Definition: af_crystalizer.c:122
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1450
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
BinkAudioContext::previous
float previous[MAX_DCT_CHANNELS][BINK_BLOCK_MAX_SIZE/16]
coeffs from previous audio block
Definition: binkaudio.c:63
codec_internal.h
BinkAudioContext::root
float root
Definition: binkaudio.c:61
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1023
ff_binkaudio_rdft_decoder
const FFCodec ff_binkaudio_rdft_decoder
Definition: binkaudio.c:368
rle_length_tab
static const uint8_t rle_length_tab[16]
Definition: binkaudio.c:164
BinkAudioContext::frame_len
int frame_len
transform size (samples)
Definition: binkaudio.c:57
ff_dct_end
av_cold void ff_dct_end(DCTContext *s)
Definition: dct.c:224
decode_init
static av_cold int decode_init(AVCodecContext *avctx)
Definition: binkaudio.c:73
AVCodec::id
enum AVCodecID id
Definition: codec.h:218
ff_rdft_init
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
Definition: rdft.c:89
BinkAudioContext::rdft
RDFTContext rdft
Definition: binkaudio.c:67
ff_dct_init
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
Set up DCT.
Definition: dct.c:179
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:499
get_bits_align32
static void get_bits_align32(GetBitContext *s)
Definition: binkaudio.c:289
BinkAudioContext::trans
union BinkAudioContext::@25 trans
RDFTContext
Definition: rdft.h:28
AVCodecInternal::in_pkt
AVPacket * in_pkt
This packet is used to hold the packet given to decoders implementing the .decode API; it is unused b...
Definition: internal.h:83
value
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default value
Definition: writing_filters.txt:86
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
DCTContext
Definition: dct.h:32
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:211
avcodec.h
ret
ret
Definition: filter_design.txt:187
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
BinkAudioContext::dct
DCTContext dct
Definition: binkaudio.c:68
AVCodecContext
main external API structure.
Definition: avcodec.h:398
power
static float power(float r, float g, float b, float max)
Definition: preserve_color.h:45
channel_layout.h
FF_CODEC_RECEIVE_FRAME_CB
#define FF_CODEC_RECEIVE_FRAME_CB(func)
Definition: codec_internal.h:285
again
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 again
Definition: filter_design.txt:25
BinkAudioContext::num_bands
int num_bands
Definition: binkaudio.c:60
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:425
AVPacket
This structure stores compressed data.
Definition: packet.h:351
BinkAudioContext::block_size
int block_size
Definition: binkaudio.c:59
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:78
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
decode_flush
static void decode_flush(AVCodecContext *avctx)
Definition: binkaudio.c:358
AV_CODEC_ID_BINKAUDIO_RDFT
@ AV_CODEC_ID_BINKAUDIO_RDFT
Definition: codec_id.h:480
MAX_CHANNELS
#define MAX_CHANNELS
Definition: binkaudio.c:48
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
BinkAudioContext::bands
unsigned int bands[26]
Definition: binkaudio.c:62