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 
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 
134  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
135  ret = ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
136  else if (CONFIG_BINKAUDIO_DCT_DECODER)
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  }
250  else if (CONFIG_BINKAUDIO_RDFT_DECODER)
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;
275  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
276  ff_rdft_end(&s->trans.rdft);
277  else if (CONFIG_BINKAUDIO_DCT_DECODER)
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");
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,
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,
364 };
av_packet_unref
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:403
AVCodec
AVCodec.
Definition: codec.h:197
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
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:222
FF_CODEC_CAP_INIT_THREADSAFE
#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
BinkAudioContext::first
int first
Definition: binkaudio.c:50
BinkAudioContext::version_b
int version_b
Bink version 'b'.
Definition: binkaudio.c:49
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
AVERROR
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 all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1149
mem_internal.h
out
FILE * out
Definition: movenc.c:54
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1098
wma_freqs.h
BinkAudioContext::channels
int channels
Definition: binkaudio.c:51
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:90
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:150
BINK_BLOCK_MAX_SIZE
#define BINK_BLOCK_MAX_SIZE
Definition: binkaudio.c:45
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:303
internal.h
BinkAudioContext::trans
union BinkAudioContext::@23 trans
expf
#define expf(x)
Definition: libm.h:283
intfloat.h
sample_rate
sample_rate
Definition: ffmpeg_filter.c:156
av_packet_free
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition: avpacket.c:70
BinkAudioContext::gb
GetBitContext gb
Definition: binkaudio.c:48
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
DCT_III
@ DCT_III
Definition: avfft.h:95
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:510
fail
#define fail()
Definition: checkasm.h:134
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:114
ff_binkaudio_rdft_decoder
const AVCodec ff_binkaudio_rdft_decoder
Definition: binkaudio.c:340
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:91
BinkAudioContext
Definition: binkaudio.c:47
BinkAudioContext::quant_table
float quant_table[96]
Definition: binkaudio.c:59
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:181
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:677
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:603
width
#define width
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
decode_end
static av_cold int decode_end(AVCodecContext *avctx)
Definition: binkaudio.c:272
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:291
AV_CODEC_ID_BINKAUDIO_DCT
@ AV_CODEC_ID_BINKAUDIO_DCT
Definition: codec_id.h:469
decode.h
get_bits.h
ff_wma_critical_freqs
const uint16_t ff_wma_critical_freqs[25]
Definition: wma_freqs.c:23
f
#define f(width, name)
Definition: cbs_vp9.c:255
ldexpf
#define ldexpf(x, exp)
Definition: libm.h:389
BinkAudioContext::overlap_len
int overlap_len
overlap size (samples)
Definition: binkaudio.c:53
BinkAudioContext::previous
float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE/16]
coeffs from previous audio block
Definition: binkaudio.c:58
receive_frame
static CopyRet receive_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame)
Definition: crystalhd.c:555
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
FFTSample
float FFTSample
Definition: avfft.h:35
BinkAudioContext::pkt
AVPacket * pkt
Definition: binkaudio.c:60
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1638
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
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
BinkAudioContext::root
float root
Definition: binkaudio.c:56
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1106
FFMIN
#define FFMIN(a, b)
Definition: common.h:105
rle_length_tab
static const uint8_t rle_length_tab[16]
Definition: binkaudio.c:159
BinkAudioContext::frame_len
int frame_len
transform size (samples)
Definition: binkaudio.c:52
av_packet_alloc
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:59
ff_dct_end
av_cold void ff_dct_end(DCTContext *s)
Definition: dct.c:221
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1099
decode_init
static av_cold int decode_init(AVCodecContext *avctx)
Definition: binkaudio.c:68
AVCodec::id
enum AVCodecID id
Definition: codec.h:211
ff_rdft_init
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
Definition: rdft.c:88
BinkAudioContext::rdft
RDFTContext rdft
Definition: binkaudio.c:62
ff_dct_init
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
Set up DCT.
Definition: dct.c:177
i
int i
Definition: input.c:407
decode_block
static int decode_block(BinkAudioContext *s, float **out, int use_dct)
Decode Bink Audio block.
Definition: binkaudio.c:168
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:602
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: internal.h:49
get_bits_align32
static void get_bits_align32(GetBitContext *s)
Definition: binkaudio.c:285
RDFTContext
Definition: rdft.h:28
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
DCTContext
Definition: dct.h:32
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:204
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:63
AVCodecContext
main external API structure.
Definition: avcodec.h:501
channel_layout.h
ff_binkaudio_dct_decoder
const AVCodec ff_binkaudio_dct_decoder
Definition: binkaudio.c:353
AV_CODEC_CAP_DELAY
#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
quant
const uint8_t * quant
Definition: vorbis_enc_data.h:458
BinkAudioContext::num_bands
int num_bands
Definition: binkaudio.c:55
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:528
AVPacket
This structure stores compressed data.
Definition: packet.h:342
BinkAudioContext::block_size
int block_size
Definition: binkaudio.c:54
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:73
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
AV_CODEC_ID_BINKAUDIO_RDFT
@ AV_CODEC_ID_BINKAUDIO_RDFT
Definition: codec_id.h:468
MAX_CHANNELS
#define MAX_CHANNELS
Definition: binkaudio.c:44
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
BinkAudioContext::bands
unsigned int bands[26]
Definition: binkaudio.c:57