FFmpeg  4.4.6
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 #include "libavutil/mem_internal.h"
34 
35 #define BITSTREAM_READER_LE
36 #include "avcodec.h"
37 #include "dct.h"
38 #include "decode.h"
39 #include "get_bits.h"
40 #include "internal.h"
41 #include "rdft.h"
42 #include "wma_freqs.h"
43 
44 #define MAX_CHANNELS 2
45 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
46 
47 typedef struct BinkAudioContext {
49  int version_b; ///< Bink version 'b'
50  int first;
51  int channels;
52  int frame_len; ///< transform size (samples)
53  int overlap_len; ///< overlap size (samples)
55  int num_bands;
56  float root;
57  unsigned int bands[26];
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, ret;
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  /* populate bands data */
127  s->bands[0] = 2;
128  for (i = 1; i < s->num_bands; i++)
129  s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
130  s->bands[s->num_bands] = s->frame_len;
131 
132  s->first = 1;
133 
135  ret = ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
137  ret = ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
138  else
139  av_assert0(0);
140  if (ret < 0)
141  return ret;
142 
143  s->pkt = av_packet_alloc();
144  if (!s->pkt)
145  return AVERROR(ENOMEM);
146 
147  return 0;
148 }
149 
150 static float get_float(GetBitContext *gb)
151 {
152  int power = get_bits(gb, 5);
153  float f = ldexpf(get_bits(gb, 23), power - 23);
154  if (get_bits1(gb))
155  f = -f;
156  return f;
157 }
158 
159 static const uint8_t rle_length_tab[16] = {
160  2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
161 };
162 
163 /**
164  * Decode Bink Audio block
165  * @param[out] out Output buffer (must contain s->block_size elements)
166  * @return 0 on success, negative error code on failure
167  */
168 static int decode_block(BinkAudioContext *s, float **out, int use_dct)
169 {
170  int ch, i, j, k;
171  float q, quant[25];
172  int width, coeff;
173  GetBitContext *gb = &s->gb;
174 
175  if (use_dct)
176  skip_bits(gb, 2);
177 
178  for (ch = 0; ch < s->channels; ch++) {
179  FFTSample *coeffs = out[ch];
180 
181  if (s->version_b) {
182  if (get_bits_left(gb) < 64)
183  return AVERROR_INVALIDDATA;
184  coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
185  coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
186  } else {
187  if (get_bits_left(gb) < 58)
188  return AVERROR_INVALIDDATA;
189  coeffs[0] = get_float(gb) * s->root;
190  coeffs[1] = get_float(gb) * s->root;
191  }
192 
193  if (get_bits_left(gb) < s->num_bands * 8)
194  return AVERROR_INVALIDDATA;
195  for (i = 0; i < s->num_bands; i++) {
196  int value = get_bits(gb, 8);
197  quant[i] = s->quant_table[FFMIN(value, 95)];
198  }
199 
200  k = 0;
201  q = quant[0];
202 
203  // parse coefficients
204  i = 2;
205  while (i < s->frame_len) {
206  if (s->version_b) {
207  j = i + 16;
208  } else {
209  int v = get_bits1(gb);
210  if (v) {
211  v = get_bits(gb, 4);
212  j = i + rle_length_tab[v] * 8;
213  } else {
214  j = i + 8;
215  }
216  }
217 
218  j = FFMIN(j, s->frame_len);
219 
220  width = get_bits(gb, 4);
221  if (width == 0) {
222  memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
223  i = j;
224  while (s->bands[k] < i)
225  q = quant[k++];
226  } else {
227  while (i < j) {
228  if (s->bands[k] == i)
229  q = quant[k++];
230  coeff = get_bits(gb, width);
231  if (coeff) {
232  int v;
233  v = get_bits1(gb);
234  if (v)
235  coeffs[i] = -q * coeff;
236  else
237  coeffs[i] = q * coeff;
238  } else {
239  coeffs[i] = 0.0f;
240  }
241  i++;
242  }
243  }
244  }
245 
246  if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
247  coeffs[0] /= 0.5;
248  s->trans.dct.dct_calc(&s->trans.dct, coeffs);
249  }
251  s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
252  }
253 
254  for (ch = 0; ch < s->channels; ch++) {
255  int j;
256  int count = s->overlap_len * s->channels;
257  if (!s->first) {
258  j = ch;
259  for (i = 0; i < s->overlap_len; i++, j += s->channels)
260  out[ch][i] = (s->previous[ch][i] * (count - j) +
261  out[ch][i] * j) / count;
262  }
263  memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
264  s->overlap_len * sizeof(*s->previous[ch]));
265  }
266 
267  s->first = 0;
268 
269  return 0;
270 }
271 
273 {
274  BinkAudioContext * s = avctx->priv_data;
276  ff_rdft_end(&s->trans.rdft);
278  ff_dct_end(&s->trans.dct);
279 
280  av_packet_free(&s->pkt);
281 
282  return 0;
283 }
284 
286 {
287  int n = (-get_bits_count(s)) & 31;
288  if (n) skip_bits(s, n);
289 }
290 
292 {
293  BinkAudioContext *s = avctx->priv_data;
294  GetBitContext *gb = &s->gb;
295  int ret;
296 
297  if (!s->pkt->data) {
298  ret = ff_decode_get_packet(avctx, s->pkt);
299  if (ret < 0)
300  return ret;
301 
302  if (s->pkt->size < 4) {
303  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
304  ret = AVERROR_INVALIDDATA;
305  goto fail;
306  }
307 
308  ret = init_get_bits8(gb, s->pkt->data, s->pkt->size);
309  if (ret < 0)
310  goto fail;
311 
312  /* skip reported size */
313  skip_bits_long(gb, 32);
314  }
315 
316  /* get output buffer */
317  frame->nb_samples = s->frame_len;
318  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
319  return ret;
320 
321  if (decode_block(s, (float **)frame->extended_data,
322  avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
323  av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
324  return AVERROR_INVALIDDATA;
325  }
326  get_bits_align32(gb);
327  if (!get_bits_left(gb)) {
328  memset(gb, 0, sizeof(*gb));
329  av_packet_unref(s->pkt);
330  }
331 
332  frame->nb_samples = s->block_size / avctx->channels;
333 
334  return 0;
335 fail:
336  av_packet_unref(s->pkt);
337  return ret;
338 }
339 
341  .name = "binkaudio_rdft",
342  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
343  .type = AVMEDIA_TYPE_AUDIO,
345  .priv_data_size = sizeof(BinkAudioContext),
346  .init = decode_init,
347  .close = decode_end,
349  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
350  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
351 };
352 
354  .name = "binkaudio_dct",
355  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
356  .type = AVMEDIA_TYPE_AUDIO,
358  .priv_data_size = sizeof(BinkAudioContext),
359  .init = decode_init,
360  .close = decode_end,
362  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
363  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
364 };
#define av_cold
Definition: attributes.h:88
uint8_t
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
static const uint8_t rle_length_tab[16]
Definition: binkaudio.c:159
static int decode_block(BinkAudioContext *s, float **out, int use_dct)
Decode Bink Audio block.
Definition: binkaudio.c:168
#define BINK_BLOCK_MAX_SIZE
Definition: binkaudio.c:45
AVCodec ff_binkaudio_rdft_decoder
Definition: binkaudio.c:340
AVCodec ff_binkaudio_dct_decoder
Definition: binkaudio.c:353
static av_cold int decode_init(AVCodecContext *avctx)
Definition: binkaudio.c:68
static av_cold int decode_end(AVCodecContext *avctx)
Definition: binkaudio.c:272
static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Definition: binkaudio.c:291
#define MAX_CHANNELS
Definition: binkaudio.c:44
static void get_bits_align32(GetBitContext *s)
Definition: binkaudio.c:285
static float get_float(GetBitContext *gb)
Definition: binkaudio.c:150
#define s(width, name)
Definition: cbs_vp9.c:257
#define f(width, name)
Definition: cbs_vp9.c:255
audio channel layout utility functions
#define fail()
Definition: checkasm.h:133
#define FFMIN(a, b)
Definition: common.h:105
#define CONFIG_BINKAUDIO_RDFT_DECODER
Definition: config.h:1049
#define CONFIG_BINKAUDIO_DCT_DECODER
Definition: config.h:1048
static CopyRet receive_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame)
Definition: crystalhd.c:560
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
Set up DCT.
Definition: dct.c:177
av_cold void ff_dct_end(DCTContext *s)
Definition: dct.c:221
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
int ff_decode_get_packet(AVCodecContext *avctx, AVPacket *pkt)
Called by decoders to get the next packet for decoding.
Definition: decode.c:222
static AVFrame * frame
double value
Definition: eval.c:98
sample_rate
bitstream reader API header.
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:546
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
#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:77
#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_BINKAUDIO_DCT
Definition: codec_id.h:472
@ AV_CODEC_ID_BINKAUDIO_RDFT
Definition: codec_id.h:471
float FFTSample
Definition: avfft.h:35
@ DFT_C2R
Definition: avfft.h:75
@ DCT_III
Definition: avfft.h:95
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition: avpacket.c:75
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:634
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:64
#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
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
int i
Definition: input.c:407
static av_always_inline float av_int2float(uint32_t i)
Reinterpret a 32-bit integer as a float.
Definition: intfloat.h:40
#define av_log2
Definition: intmath.h:83
#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
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
#define expf(x)
Definition: libm.h:283
#define ldexpf(x, exp)
Definition: libm.h:389
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
Definition: rdft.c:88
av_cold void ff_rdft_end(RDFTContext *s)
Definition: rdft.c:114
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
const struct AVCodec * codec
Definition: avcodec.h:545
int sample_rate
samples per second
Definition: avcodec.h:1196
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int channels
number of audio channels
Definition: avcodec.h:1197
int extradata_size
Definition: avcodec.h:638
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
enum AVCodecID id
Definition: codec.h:211
const char * name
Name of the codec implementation.
Definition: codec.h:204
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
float quant_table[96]
Definition: binkaudio.c:59
AVPacket * pkt
Definition: binkaudio.c:60
union BinkAudioContext::@23 trans
int overlap_len
overlap size (samples)
Definition: binkaudio.c:53
RDFTContext rdft
Definition: binkaudio.c:62
int frame_len
transform size (samples)
Definition: binkaudio.c:52
DCTContext dct
Definition: binkaudio.c:63
GetBitContext gb
Definition: binkaudio.c:48
int version_b
Bink version 'b'.
Definition: binkaudio.c:49
unsigned int bands[26]
Definition: binkaudio.c:57
float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE/16]
coeffs from previous audio block
Definition: binkaudio.c:58
Definition: dct.h:32
#define av_log(a,...)
FILE * out
Definition: movenc.c:54
#define width
static const double coeff[2][5]
Definition: vf_owdenoise.c:73
const uint8_t * quant
const uint16_t ff_wma_critical_freqs[25]
Definition: wma_freqs.c:23