FFmpeg  4.3.9
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 "libavutil/intfloat.h"
33 
34 #define BITSTREAM_READER_LE
35 #include "avcodec.h"
36 #include "dct.h"
37 #include "decode.h"
38 #include "get_bits.h"
39 #include "internal.h"
40 #include "rdft.h"
41 #include "wma_freqs.h"
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
59  float quant_table[96];
61  union {
64  } trans;
66 
67 
69 {
70  BinkAudioContext *s = avctx->priv_data;
71  int sample_rate = avctx->sample_rate;
72  int sample_rate_half;
73  int i;
74  int frame_len_bits;
75 
76  /* determine frame length */
77  if (avctx->sample_rate < 22050) {
78  frame_len_bits = 9;
79  } else if (avctx->sample_rate < 44100) {
80  frame_len_bits = 10;
81  } else {
82  frame_len_bits = 11;
83  }
84 
85  if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
86  av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
87  return AVERROR_INVALIDDATA;
88  }
89  avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
91 
92  s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
93 
94  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
95  // audio is already interleaved for the RDFT format variant
97  if (sample_rate > INT_MAX / avctx->channels)
98  return AVERROR_INVALIDDATA;
99  sample_rate *= avctx->channels;
100  s->channels = 1;
101  if (!s->version_b)
102  frame_len_bits += av_log2(avctx->channels);
103  } else {
104  s->channels = avctx->channels;
106  }
107 
108  s->frame_len = 1 << frame_len_bits;
109  s->overlap_len = s->frame_len / 16;
110  s->block_size = (s->frame_len - s->overlap_len) * s->channels;
111  sample_rate_half = (sample_rate + 1LL) / 2;
112  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
113  s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
114  else
115  s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
116  for (i = 0; i < 96; i++) {
117  /* constant is result of 0.066399999/log10(M_E) */
118  s->quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
119  }
120 
121  /* calculate number of bands */
122  for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
123  if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
124  break;
125 
126  s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
127  if (!s->bands)
128  return AVERROR(ENOMEM);
129 
130  /* populate bands data */
131  s->bands[0] = 2;
132  for (i = 1; i < s->num_bands; i++)
133  s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
134  s->bands[s->num_bands] = s->frame_len;
135 
136  s->first = 1;
137 
139  ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
141  ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
142  else
143  av_assert0(0);
144 
145  s->pkt = av_packet_alloc();
146  if (!s->pkt)
147  return AVERROR(ENOMEM);
148 
149  return 0;
150 }
151 
152 static float get_float(GetBitContext *gb)
153 {
154  int power = get_bits(gb, 5);
155  float f = ldexpf(get_bits(gb, 23), power - 23);
156  if (get_bits1(gb))
157  f = -f;
158  return f;
159 }
160 
161 static const uint8_t rle_length_tab[16] = {
162  2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
163 };
164 
165 /**
166  * Decode Bink Audio block
167  * @param[out] out Output buffer (must contain s->block_size elements)
168  * @return 0 on success, negative error code on failure
169  */
170 static int decode_block(BinkAudioContext *s, float **out, int use_dct)
171 {
172  int ch, i, j, k;
173  float q, quant[25];
174  int width, coeff;
175  GetBitContext *gb = &s->gb;
176 
177  if (use_dct)
178  skip_bits(gb, 2);
179 
180  for (ch = 0; ch < s->channels; ch++) {
181  FFTSample *coeffs = out[ch];
182 
183  if (s->version_b) {
184  if (get_bits_left(gb) < 64)
185  return AVERROR_INVALIDDATA;
186  coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
187  coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
188  } else {
189  if (get_bits_left(gb) < 58)
190  return AVERROR_INVALIDDATA;
191  coeffs[0] = get_float(gb) * s->root;
192  coeffs[1] = get_float(gb) * s->root;
193  }
194 
195  if (get_bits_left(gb) < s->num_bands * 8)
196  return AVERROR_INVALIDDATA;
197  for (i = 0; i < s->num_bands; i++) {
198  int value = get_bits(gb, 8);
199  quant[i] = s->quant_table[FFMIN(value, 95)];
200  }
201 
202  k = 0;
203  q = quant[0];
204 
205  // parse coefficients
206  i = 2;
207  while (i < s->frame_len) {
208  if (s->version_b) {
209  j = i + 16;
210  } else {
211  int v = get_bits1(gb);
212  if (v) {
213  v = get_bits(gb, 4);
214  j = i + rle_length_tab[v] * 8;
215  } else {
216  j = i + 8;
217  }
218  }
219 
220  j = FFMIN(j, s->frame_len);
221 
222  width = get_bits(gb, 4);
223  if (width == 0) {
224  memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
225  i = j;
226  while (s->bands[k] < i)
227  q = quant[k++];
228  } else {
229  while (i < j) {
230  if (s->bands[k] == i)
231  q = quant[k++];
232  coeff = get_bits(gb, width);
233  if (coeff) {
234  int v;
235  v = get_bits1(gb);
236  if (v)
237  coeffs[i] = -q * coeff;
238  else
239  coeffs[i] = q * coeff;
240  } else {
241  coeffs[i] = 0.0f;
242  }
243  i++;
244  }
245  }
246  }
247 
248  if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
249  coeffs[0] /= 0.5;
250  s->trans.dct.dct_calc(&s->trans.dct, coeffs);
251  }
253  s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
254  }
255 
256  for (ch = 0; ch < s->channels; ch++) {
257  int j;
258  int count = s->overlap_len * s->channels;
259  if (!s->first) {
260  j = ch;
261  for (i = 0; i < s->overlap_len; i++, j += s->channels)
262  out[ch][i] = (s->previous[ch][i] * (count - j) +
263  out[ch][i] * j) / count;
264  }
265  memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
266  s->overlap_len * sizeof(*s->previous[ch]));
267  }
268 
269  s->first = 0;
270 
271  return 0;
272 }
273 
275 {
276  BinkAudioContext * s = avctx->priv_data;
277  av_freep(&s->bands);
279  ff_rdft_end(&s->trans.rdft);
281  ff_dct_end(&s->trans.dct);
282 
283  av_packet_free(&s->pkt);
284 
285  return 0;
286 }
287 
289 {
290  int n = (-get_bits_count(s)) & 31;
291  if (n) skip_bits(s, n);
292 }
293 
295 {
296  BinkAudioContext *s = avctx->priv_data;
297  GetBitContext *gb = &s->gb;
298  int ret;
299 
300  if (!s->pkt->data) {
301  ret = ff_decode_get_packet(avctx, s->pkt);
302  if (ret < 0)
303  return ret;
304 
305  if (s->pkt->size < 4) {
306  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
307  ret = AVERROR_INVALIDDATA;
308  goto fail;
309  }
310 
311  ret = init_get_bits8(gb, s->pkt->data, s->pkt->size);
312  if (ret < 0)
313  goto fail;
314 
315  /* skip reported size */
316  skip_bits_long(gb, 32);
317  }
318 
319  /* get output buffer */
320  frame->nb_samples = s->frame_len;
321  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
322  return ret;
323 
324  if (decode_block(s, (float **)frame->extended_data,
325  avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
326  av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
327  return AVERROR_INVALIDDATA;
328  }
329  get_bits_align32(gb);
330  if (!get_bits_left(gb)) {
331  memset(gb, 0, sizeof(*gb));
332  av_packet_unref(s->pkt);
333  }
334 
335  frame->nb_samples = s->block_size / avctx->channels;
336 
337  return 0;
338 fail:
339  av_packet_unref(s->pkt);
340  return ret;
341 }
342 
344  .name = "binkaudio_rdft",
345  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
346  .type = AVMEDIA_TYPE_AUDIO,
348  .priv_data_size = sizeof(BinkAudioContext),
349  .init = decode_init,
350  .close = decode_end,
352  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
353 };
354 
356  .name = "binkaudio_dct",
357  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
358  .type = AVMEDIA_TYPE_AUDIO,
360  .priv_data_size = sizeof(BinkAudioContext),
361  .init = decode_init,
362  .close = decode_end,
364  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
365 };
av_cold void ff_rdft_end(RDFTContext *s)
Definition: rdft.c:114
float, planar
Definition: samplefmt.h:69
const struct AVCodec * codec
Definition: avcodec.h:535
static float get_float(GetBitContext *gb)
Definition: binkaudio.c:152
#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:300
Definition: avfft.h:75
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static av_cold int decode_end(AVCodecContext *avctx)
Definition: binkaudio.c:274
Definition: avfft.h:95
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
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:161
int size
Definition: packet.h:356
const uint16_t ff_wma_critical_freqs[25]
Definition: wma_freqs.c:23
static CopyRet receive_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame)
Definition: crystalhd.c:560
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: codec.h:190
#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: codec.h:75
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition: avpacket.c:64
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
uint8_t
#define av_cold
Definition: attributes.h:88
#define av_malloc(s)
#define f(width, name)
Definition: cbs_vp9.c:255
int ff_decode_get_packet(AVCodecContext *avctx, AVPacket *pkt)
Called by decoders to get the next packet for decoding.
Definition: decode.c:238
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:627
unsigned int * bands
Definition: binkaudio.c:55
static AVFrame * frame
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:112
uint8_t * data
Definition: packet.h:355
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Definition: binkaudio.c:294
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:283
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
enum AVCodecID id
Definition: codec.h:204
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
#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:188
static void get_bits_align32(GetBitContext *s)
Definition: binkaudio.c:288
static int decode_block(BinkAudioContext *s, float **out, int use_dct)
Decode Bink Audio block.
Definition: binkaudio.c:170
const char * name
Name of the codec implementation.
Definition: codec.h:197
float FFTSample
Definition: avfft.h:35
#define fail()
Definition: checkasm.h:123
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
GetBitContext gb
Definition: binkaudio.c:47
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
#define width
void(* dct_calc)(struct DCTContext *s, FFTSample *data)
Definition: dct.h:38
DCTContext dct
Definition: binkaudio.c:63
Definition: dct.h:32
#define s(width, name)
Definition: cbs_vp9.c:257
static av_cold int decode_init(AVCodecContext *avctx)
Definition: binkaudio.c:68
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
Definition: rdft.h:38
AVCodec ff_binkaudio_rdft_decoder
Definition: binkaudio.c:343
float quant_table[96]
Definition: binkaudio.c:59
#define av_log2
Definition: intmath.h:83
int overlap_len
overlap size (samples)
Definition: binkaudio.c:52
sample_rate
#define CONFIG_BINKAUDIO_DCT_DECODER
Definition: config.h:1027
Libavcodec external API header.
int sample_rate
samples per second
Definition: avcodec.h:1186
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
AVCodec ff_binkaudio_dct_decoder
Definition: binkaudio.c:355
main external API structure.
Definition: avcodec.h:526
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:605
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1854
#define ldexpf(x, exp)
Definition: libm.h:389
int extradata_size
Definition: avcodec.h:628
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
#define CONFIG_BINKAUDIO_RDFT_DECODER
Definition: config.h:1028
AVPacket * pkt
Definition: binkaudio.c:60
double value
Definition: eval.c:98
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
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:546
const uint8_t * quant
int frame_len
transform size (samples)
Definition: binkaudio.c:51
int version_b
Bink version &#39;b&#39;.
Definition: binkaudio.c:48
common internal api header.
RDFTContext rdft
Definition: binkaudio.c:62
FFTSample coeffs[BINK_BLOCK_MAX_SIZE]
Definition: binkaudio.c:57
void * priv_data
Definition: avcodec.h:553
int channels
number of audio channels
Definition: avcodec.h:1187
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:53
static const double coeff[2][5]
Definition: vf_owdenoise.c:72
av_cold void ff_dct_end(DCTContext *s)
Definition: dct.c:221
FILE * out
Definition: movenc.c:54
#define av_freep(p)
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:347
#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:88
This structure stores compressed data.
Definition: packet.h:332
union BinkAudioContext::@23 trans
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50