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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  sample_rate *= avctx->channels;
99  s->channels = 1;
100  if (!s->version_b)
101  frame_len_bits += av_log2(avctx->channels);
102  } else {
103  s->channels = avctx->channels;
105  }
106 
107  s->frame_len = 1 << frame_len_bits;
108  s->overlap_len = s->frame_len / 16;
109  s->block_size = (s->frame_len - s->overlap_len) * s->channels;
110  sample_rate_half = (sample_rate + 1) / 2;
111  if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
112  s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
113  else
114  s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
115  for (i = 0; i < 96; i++) {
116  /* constant is result of 0.066399999/log10(M_E) */
117  quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
118  }
119 
120  /* calculate number of bands */
121  for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
122  if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
123  break;
124 
125  s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
126  if (!s->bands)
127  return AVERROR(ENOMEM);
128 
129  /* populate bands data */
130  s->bands[0] = 2;
131  for (i = 1; i < s->num_bands; i++)
132  s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
133  s->bands[s->num_bands] = s->frame_len;
134 
135  s->first = 1;
136 
137  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
138  ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
139  else if (CONFIG_BINKAUDIO_DCT_DECODER)
140  ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
141  else
142  return -1;
143 
144  s->pkt = av_packet_alloc();
145  if (!s->pkt)
146  return AVERROR(ENOMEM);
147 
148  return 0;
149 }
150 
151 static float get_float(GetBitContext *gb)
152 {
153  int power = get_bits(gb, 5);
154  float f = ldexpf(get_bits_long(gb, 23), power - 23);
155  if (get_bits1(gb))
156  f = -f;
157  return f;
158 }
159 
160 static const uint8_t rle_length_tab[16] = {
161  2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
162 };
163 
164 /**
165  * Decode Bink Audio block
166  * @param[out] out Output buffer (must contain s->block_size elements)
167  * @return 0 on success, negative error code on failure
168  */
169 static int decode_block(BinkAudioContext *s, float **out, int use_dct)
170 {
171  int ch, i, j, k;
172  float q, quant[25];
173  int width, coeff;
174  GetBitContext *gb = &s->gb;
175 
176  if (use_dct)
177  skip_bits(gb, 2);
178 
179  for (ch = 0; ch < s->channels; ch++) {
180  FFTSample *coeffs = out[ch];
181 
182  if (s->version_b) {
183  if (get_bits_left(gb) < 64)
184  return AVERROR_INVALIDDATA;
185  coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
186  coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
187  } else {
188  if (get_bits_left(gb) < 58)
189  return AVERROR_INVALIDDATA;
190  coeffs[0] = get_float(gb) * s->root;
191  coeffs[1] = get_float(gb) * s->root;
192  }
193 
194  if (get_bits_left(gb) < s->num_bands * 8)
195  return AVERROR_INVALIDDATA;
196  for (i = 0; i < s->num_bands; i++) {
197  int value = get_bits(gb, 8);
198  quant[i] = quant_table[FFMIN(value, 95)];
199  }
200 
201  k = 0;
202  q = quant[0];
203 
204  // parse coefficients
205  i = 2;
206  while (i < s->frame_len) {
207  if (s->version_b) {
208  j = i + 16;
209  } else {
210  int v = get_bits1(gb);
211  if (v) {
212  v = get_bits(gb, 4);
213  j = i + rle_length_tab[v] * 8;
214  } else {
215  j = i + 8;
216  }
217  }
218 
219  j = FFMIN(j, s->frame_len);
220 
221  width = get_bits(gb, 4);
222  if (width == 0) {
223  memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
224  i = j;
225  while (s->bands[k] < i)
226  q = quant[k++];
227  } else {
228  while (i < j) {
229  if (s->bands[k] == i)
230  q = quant[k++];
231  coeff = get_bits(gb, width);
232  if (coeff) {
233  int v;
234  v = get_bits1(gb);
235  if (v)
236  coeffs[i] = -q * coeff;
237  else
238  coeffs[i] = q * coeff;
239  } else {
240  coeffs[i] = 0.0f;
241  }
242  i++;
243  }
244  }
245  }
246 
247  if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
248  coeffs[0] /= 0.5;
249  s->trans.dct.dct_calc(&s->trans.dct, coeffs);
250  }
251  else if (CONFIG_BINKAUDIO_RDFT_DECODER)
252  s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
253  }
254 
255  for (ch = 0; ch < s->channels; ch++) {
256  int j;
257  int count = s->overlap_len * s->channels;
258  if (!s->first) {
259  j = ch;
260  for (i = 0; i < s->overlap_len; i++, j += s->channels)
261  out[ch][i] = (s->previous[ch][i] * (count - j) +
262  out[ch][i] * j) / count;
263  }
264  memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
265  s->overlap_len * sizeof(*s->previous[ch]));
266  }
267 
268  s->first = 0;
269 
270  return 0;
271 }
272 
274 {
275  BinkAudioContext * s = avctx->priv_data;
276  av_freep(&s->bands);
277  if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
278  ff_rdft_end(&s->trans.rdft);
279  else if (CONFIG_BINKAUDIO_DCT_DECODER)
280  ff_dct_end(&s->trans.dct);
281 
282  av_packet_free(&s->pkt);
283 
284  return 0;
285 }
286 
288 {
289  int n = (-get_bits_count(s)) & 31;
290  if (n) skip_bits(s, n);
291 }
292 
294 {
295  BinkAudioContext *s = avctx->priv_data;
296  GetBitContext *gb = &s->gb;
297  int ret;
298 
299  if (!s->pkt->data) {
300  ret = ff_decode_get_packet(avctx, s->pkt);
301  if (ret < 0)
302  return ret;
303 
304  if (s->pkt->size < 4) {
305  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
306  ret = AVERROR_INVALIDDATA;
307  goto fail;
308  }
309 
310  ret = init_get_bits8(gb, s->pkt->data, s->pkt->size);
311  if (ret < 0)
312  goto fail;
313 
314  /* skip reported size */
315  skip_bits_long(gb, 32);
316  }
317 
318  /* get output buffer */
319  frame->nb_samples = s->frame_len;
320  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
321  return ret;
322 
323  if (decode_block(s, (float **)frame->extended_data,
324  avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
325  av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
326  return AVERROR_INVALIDDATA;
327  }
328  get_bits_align32(gb);
329  if (!get_bits_left(gb)) {
330  memset(gb, 0, sizeof(*gb));
331  av_packet_unref(s->pkt);
332  }
333 
334  frame->nb_samples = s->block_size / avctx->channels;
335 
336  return 0;
337 fail:
338  av_packet_unref(s->pkt);
339  return ret;
340 }
341 
343  .name = "binkaudio_rdft",
344  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
345  .type = AVMEDIA_TYPE_AUDIO,
347  .priv_data_size = sizeof(BinkAudioContext),
348  .init = decode_init,
349  .close = decode_end,
351  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
352 };
353 
355  .name = "binkaudio_dct",
356  .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
357  .type = AVMEDIA_TYPE_AUDIO,
359  .priv_data_size = sizeof(BinkAudioContext),
360  .init = decode_init,
361  .close = decode_end,
363  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
364 };
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:1497
static float get_float(GetBitContext *gb)
Definition: binkaudio.c:151
const char * s
Definition: avisynth_c.h:768
#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:201
Definition: avfft.h:75
void(* dct_calc)(struct DCTContext *s, FFTSample *data)
Definition: dct.h:38
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
static av_cold int decode_end(AVCodecContext *avctx)
Definition: binkaudio.c:273
Definition: avfft.h:95
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:204
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:160
int size
Definition: avcodec.h:1401
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:3351
#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:981
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:2151
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
int ff_decode_get_packet(AVCodecContext *avctx, AVPacket *pkt)
Called by decoders to get the next packet for decoding.
Definition: decode.c:294
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1588
unsigned int * bands
Definition: binkaudio.c:57
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: avcodec.h:1400
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:199
static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Definition: binkaudio.c:293
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,...)
#define expf(x)
Definition: libm.h:283
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:587
enum AVCodecID id
Definition: avcodec.h:3365
#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 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:186
static void get_bits_align32(GetBitContext *s)
Definition: binkaudio.c:287
static int decode_block(BinkAudioContext *s, float **out, int use_dct)
Decode Bink Audio block.
Definition: binkaudio.c:169
uint16_t width
Definition: gdv.c:47
const char * name
Name of the codec implementation.
Definition: avcodec.h:3358
GLsizei count
Definition: opengl_enc.c:109
float FFTSample
Definition: avfft.h:35
#define fail()
Definition: checkasm.h:112
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2194
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
Definition: rdft.h:38
GetBitContext gb
Definition: binkaudio.c:49
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
static float quant_table[96]
Definition: binkaudio.c:43
DCTContext dct
Definition: binkaudio.c:64
Definition: dct.h:32
static av_cold int decode_init(AVCodecContext *avctx)
Definition: binkaudio.c:69
int n
Definition: avisynth_c.h:684
AVCodec ff_binkaudio_rdft_decoder
Definition: binkaudio.c:342
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:2143
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:456
AVCodec ff_binkaudio_dct_decoder
Definition: binkaudio.c:354
main external API structure.
Definition: avcodec.h:1488
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:590
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1886
#define ldexpf(x, exp)
Definition: libm.h:389
int extradata_size
Definition: avcodec.h:1589
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:313
AVPacket * pkt
Definition: binkaudio.c:61
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:306
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:346
const uint8_t * quant
int frame_len
transform size (samples)
Definition: binkaudio.c:53
int version_b
Bink version 'b'.
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:1515
static const int16_t coeffs[]
int channels
number of audio channels
Definition: avcodec.h:2144
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:220
FILE * out
Definition: movenc.c:54
#define av_freep(p)
union BinkAudioContext::@53 trans
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
#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:1377
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:956
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_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),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){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) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;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)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8: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);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=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){intplanes=out->planar?out->ch_count:1;unsignedm=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){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56