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 
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 static float quant_table[96];
44 
45 #define MAX_CHANNELS 2
46 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
47 
48 typedef struct BinkAudioContext {
50  int version_b; ///< Bink version 'b'
51  int first;
52  int channels;
53  int frame_len; ///< transform size (samples)
54  int overlap_len; ///< overlap size (samples)
56  int num_bands;
57  unsigned int *bands;
58  float root;
60  float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
62  union {
65  } trans;
67 
68 
70 {
71  BinkAudioContext *s = avctx->priv_data;
72  int sample_rate = avctx->sample_rate;
73  int sample_rate_half;
74  int i;
75  int frame_len_bits;
76 
77  /* determine frame length */
78  if (avctx->sample_rate < 22050) {
79  frame_len_bits = 9;
80  } else if (avctx->sample_rate < 44100) {
81  frame_len_bits = 10;
82  } else {
83  frame_len_bits = 11;
84  }
85 
86  if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
87  av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
88  return AVERROR_INVALIDDATA;
89  }
90  avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
92 
93  s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
94 
95  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
96  // audio is already interleaved for the RDFT format variant
98  if (sample_rate > INT_MAX / avctx->channels)
99  return AVERROR_INVALIDDATA;
100  sample_rate *= avctx->channels;
101  s->channels = 1;
102  if (!s->version_b)
103  frame_len_bits += av_log2(avctx->channels);
104  } else {
105  s->channels = avctx->channels;
107  }
108 
109  s->frame_len = 1 << frame_len_bits;
110  s->overlap_len = s->frame_len / 16;
111  s->block_size = (s->frame_len - s->overlap_len) * s->channels;
112  sample_rate_half = (sample_rate + 1) / 2;
113  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
114  s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
115  else
116  s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
117  for (i = 0; i < 96; i++) {
118  /* constant is result of 0.066399999/log10(M_E) */
119  quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
120  }
121 
122  /* calculate number of bands */
123  for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
124  if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
125  break;
126 
127  s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
128  if (!s->bands)
129  return AVERROR(ENOMEM);
130 
131  /* populate bands data */
132  s->bands[0] = 2;
133  for (i = 1; i < s->num_bands; i++)
134  s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
135  s->bands[s->num_bands] = s->frame_len;
136 
137  s->first = 1;
138 
139  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
140  ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
141  else if (CONFIG_BINKAUDIO_DCT_DECODER)
142  ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
143  else
144  av_assert0(0);
145 
146  s->pkt = av_packet_alloc();
147  if (!s->pkt)
148  return AVERROR(ENOMEM);
149 
150  return 0;
151 }
152 
153 static float get_float(GetBitContext *gb)
154 {
155  int power = get_bits(gb, 5);
156  float f = ldexpf(get_bits_long(gb, 23), power - 23);
157  if (get_bits1(gb))
158  f = -f;
159  return f;
160 }
161 
162 static const uint8_t rle_length_tab[16] = {
163  2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
164 };
165 
166 /**
167  * Decode Bink Audio block
168  * @param[out] out Output buffer (must contain s->block_size elements)
169  * @return 0 on success, negative error code on failure
170  */
171 static int decode_block(BinkAudioContext *s, float **out, int use_dct)
172 {
173  int ch, i, j, k;
174  float q, quant[25];
175  int width, coeff;
176  GetBitContext *gb = &s->gb;
177 
178  if (use_dct)
179  skip_bits(gb, 2);
180 
181  for (ch = 0; ch < s->channels; ch++) {
182  FFTSample *coeffs = out[ch];
183 
184  if (s->version_b) {
185  if (get_bits_left(gb) < 64)
186  return AVERROR_INVALIDDATA;
187  coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
188  coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
189  } else {
190  if (get_bits_left(gb) < 58)
191  return AVERROR_INVALIDDATA;
192  coeffs[0] = get_float(gb) * s->root;
193  coeffs[1] = get_float(gb) * s->root;
194  }
195 
196  if (get_bits_left(gb) < s->num_bands * 8)
197  return AVERROR_INVALIDDATA;
198  for (i = 0; i < s->num_bands; i++) {
199  int value = get_bits(gb, 8);
200  quant[i] = quant_table[FFMIN(value, 95)];
201  }
202 
203  k = 0;
204  q = quant[0];
205 
206  // parse coefficients
207  i = 2;
208  while (i < s->frame_len) {
209  if (s->version_b) {
210  j = i + 16;
211  } else {
212  int v = get_bits1(gb);
213  if (v) {
214  v = get_bits(gb, 4);
215  j = i + rle_length_tab[v] * 8;
216  } else {
217  j = i + 8;
218  }
219  }
220 
221  j = FFMIN(j, s->frame_len);
222 
223  width = get_bits(gb, 4);
224  if (width == 0) {
225  memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
226  i = j;
227  while (s->bands[k] < i)
228  q = quant[k++];
229  } else {
230  while (i < j) {
231  if (s->bands[k] == i)
232  q = quant[k++];
233  coeff = get_bits(gb, width);
234  if (coeff) {
235  int v;
236  v = get_bits1(gb);
237  if (v)
238  coeffs[i] = -q * coeff;
239  else
240  coeffs[i] = q * coeff;
241  } else {
242  coeffs[i] = 0.0f;
243  }
244  i++;
245  }
246  }
247  }
248 
249  if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
250  coeffs[0] /= 0.5;
251  s->trans.dct.dct_calc(&s->trans.dct, coeffs);
252  }
253  else if (CONFIG_BINKAUDIO_RDFT_DECODER)
254  s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
255  }
256 
257  for (ch = 0; ch < s->channels; ch++) {
258  int j;
259  int count = s->overlap_len * s->channels;
260  if (!s->first) {
261  j = ch;
262  for (i = 0; i < s->overlap_len; i++, j += s->channels)
263  out[ch][i] = (s->previous[ch][i] * (count - j) +
264  out[ch][i] * j) / count;
265  }
266  memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
267  s->overlap_len * sizeof(*s->previous[ch]));
268  }
269 
270  s->first = 0;
271 
272  return 0;
273 }
274 
276 {
277  BinkAudioContext * s = avctx->priv_data;
278  av_freep(&s->bands);
279  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
280  ff_rdft_end(&s->trans.rdft);
281  else if (CONFIG_BINKAUDIO_DCT_DECODER)
282  ff_dct_end(&s->trans.dct);
283 
284  av_packet_free(&s->pkt);
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  if (!s->pkt->data) {
302  ret = ff_decode_get_packet(avctx, s->pkt);
303  if (ret < 0)
304  return ret;
305 
306  if (s->pkt->size < 4) {
307  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
308  ret = AVERROR_INVALIDDATA;
309  goto fail;
310  }
311 
312  ret = init_get_bits8(gb, s->pkt->data, s->pkt->size);
313  if (ret < 0)
314  goto fail;
315 
316  /* skip reported size */
317  skip_bits_long(gb, 32);
318  }
319 
320  /* get output buffer */
321  frame->nb_samples = s->frame_len;
322  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
323  return ret;
324 
325  if (decode_block(s, (float **)frame->extended_data,
326  avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
327  av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
328  return AVERROR_INVALIDDATA;
329  }
330  get_bits_align32(gb);
331  if (!get_bits_left(gb)) {
332  memset(gb, 0, sizeof(*gb));
333  av_packet_unref(s->pkt);
334  }
335 
336  frame->nb_samples = s->block_size / avctx->channels;
337 
338  return 0;
339 fail:
340  av_packet_unref(s->pkt);
341  return ret;
342 }
343 
345  .name = "binkaudio_rdft",
346  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
347  .type = AVMEDIA_TYPE_AUDIO,
349  .priv_data_size = sizeof(BinkAudioContext),
350  .init = decode_init,
351  .close = decode_end,
353  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
354 };
355 
357  .name = "binkaudio_dct",
358  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
359  .type = AVMEDIA_TYPE_AUDIO,
361  .priv_data_size = sizeof(BinkAudioContext),
362  .init = decode_init,
363  .close = decode_end,
365  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
366 };
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:1577
static float get_float(GetBitContext *gb)
Definition: binkaudio.c:153
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define MAX_CHANNELS
Definition: binkaudio.c:45
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
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:275
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:162
int size
Definition: avcodec.h:1481
int av_log2(unsigned v)
Definition: intmath.c:26
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: avcodec.h:3492
#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:1009
#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:62
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
#define av_cold
Definition: attributes.h:82
#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:328
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:108
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1669
unsigned int * bands
Definition: binkaudio.c:57
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:112
uint8_t * data
Definition: avcodec.h:1480
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:295
bitstream reader API header.
float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE/16]
coeffs from previous audio block
Definition: binkaudio.c:60
#define av_log(a,...)
union BinkAudioContext::@46 trans
#define expf(x)
Definition: libm.h:283
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
enum AVCodecID id
Definition: avcodec.h:3506
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define BINK_BLOCK_MAX_SIZE
Definition: binkaudio.c:46
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
static void get_bits_align32(GetBitContext *s)
Definition: binkaudio.c:289
static int decode_block(BinkAudioContext *s, float **out, int use_dct)
Decode Bink Audio block.
Definition: binkaudio.c:171
const char * name
Name of the codec implementation.
Definition: avcodec.h:3499
GLsizei count
Definition: opengl_enc.c:108
float FFTSample
Definition: avfft.h:35
#define fail()
Definition: checkasm.h:122
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2279
GetBitContext gb
Definition: binkaudio.c:49
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
static float quant_table[96]
Definition: binkaudio.c:43
DCTContext dct
Definition: binkaudio.c:64
Definition: dct.h:32
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
#define s(width, name)
Definition: cbs_vp9.c:257
static av_cold int decode_init(AVCodecContext *avctx)
Definition: binkaudio.c:69
int n
Definition: avisynth_c.h:760
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
Definition: rdft.h:38
AVCodec ff_binkaudio_rdft_decoder
Definition: binkaudio.c:344
int overlap_len
overlap size (samples)
Definition: binkaudio.c:54
sample_rate
Libavcodec external API header.
int sample_rate
samples per second
Definition: avcodec.h:2228
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:356
main external API structure.
Definition: avcodec.h:1568
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:599
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1968
#define ldexpf(x, exp)
Definition: libm.h:389
int extradata_size
Definition: avcodec.h:1670
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
AVPacket * pkt
Definition: binkaudio.c:61
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
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){int planes=out->planar?out->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
const uint8_t * quant
int frame_len
transform size (samples)
Definition: binkaudio.c:53
int version_b
Bink version &#39;b&#39;.
Definition: binkaudio.c:50
common internal api header.
RDFTContext rdft
Definition: binkaudio.c:63
FFTSample coeffs[BINK_BLOCK_MAX_SIZE]
Definition: binkaudio.c:59
void * priv_data
Definition: avcodec.h:1595
int channels
number of audio channels
Definition: avcodec.h:2229
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:51
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)
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
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
#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: avcodec.h:1457
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:984