1 /*
2 * ALAC (Apple Lossless Audio Codec) decoder
3 * Copyright (c) 2005 David Hammerton
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * ALAC (Apple Lossless Audio Codec) decoder
25 * @author 2005 David Hammerton
26 * @see http://crazney.net/programs/itunes/alac.html
27 *
28 * Note: This decoder expects a 36-byte QuickTime atom to be
29 * passed through the extradata[_size] fields. This atom is tacked onto
30 * the end of an 'alac' stsd atom and has the following format:
31 *
32 * 32 bits atom size
33 * 32 bits tag ("alac")
34 * 32 bits tag version (0)
35 * 32 bits samples per frame (used when not set explicitly in the frames)
36 * 8 bits compatible version (0)
37 * 8 bits sample size
38 * 8 bits history mult (40)
39 * 8 bits initial history (10)
40 * 8 bits rice param limit (14)
41 * 8 bits channels
42 * 16 bits maxRun (255)
43 * 32 bits max coded frame size (0 means unknown)
44 * 32 bits average bitrate (0 means unknown)
45 * 32 bits samplerate
46 */
47
48 #include <inttypes.h>
49
61
62 #define ALAC_EXTRADATA_SIZE 36
63
69
73
80
81 int extra_bits;
/**< number of extra bits beyond 16-bit */
82 int nb_samples;
/**< number of samples in the current frame */
83
86
89
91 {
93
94 if (x > 8) { /* RICE THRESHOLD */
95 /* use alternative encoding */
97 } else if (k != 1) {
99
100 /* multiply x by 2^k - 1, as part of their strange algorithm */
101 x = (x << k) - x;
102
103 if (extrabits > 1) {
104 x += extrabits - 1;
106 } else
108 }
109 return x;
110 }
111
113 int nb_samples,
int bps,
int rice_history_mult)
114 {
115 int i;
117 int sign_modifier = 0;
118
119 for (i = 0; i < nb_samples; i++) {
120 int k;
121 unsigned int x;
122
124 return -1;
125
126 /* calculate rice param and decode next value */
127 k =
av_log2((history >> 9) + 3);
130 x += sign_modifier;
131 sign_modifier = 0;
132 output_buffer[i] = (x >> 1) ^ -(x & 1);
133
134 /* update the history */
135 if (x > 0xffff)
136 history = 0xffff;
137 else
138 history += x * rice_history_mult -
139 ((history * rice_history_mult) >> 9);
140
141 /* special case: there may be compressed blocks of 0 */
142 if ((history < 128) && (i + 1 < nb_samples)) {
143 int block_size;
144
145 /* calculate rice param and decode block size */
146 k = 7 -
av_log2(history) + ((history + 16) >> 6);
149
150 if (block_size > 0) {
151 if (block_size >= nb_samples - i) {
153 "invalid zero block size of %d %d %d\n", block_size,
154 nb_samples, i);
155 block_size = nb_samples - i - 1;
156 }
157 memset(&output_buffer[i + 1], 0,
158 block_size * sizeof(*output_buffer));
159 i += block_size;
160 }
161 if (block_size <= 0xffff)
162 sign_modifier = 1;
163 history = 0;
164 }
165 }
166 return 0;
167 }
168
170 {
172 }
173
175 int nb_samples,
int bps, int16_t *lpc_coefs,
176 int lpc_order, int lpc_quant)
177 {
178 int i;
180
181 /* first sample always copies */
182 *buffer_out = *error_buffer;
183
184 if (nb_samples <= 1)
185 return;
186
187 if (!lpc_order) {
188 memcpy(&buffer_out[1], &error_buffer[1],
189 (nb_samples - 1) * sizeof(*buffer_out));
190 return;
191 }
192
193 if (lpc_order == 31) {
194 /* simple 1st-order prediction */
195 for (i = 1; i < nb_samples; i++) {
196 buffer_out[i] =
sign_extend(buffer_out[i - 1] + error_buffer[i],
197 bps);
198 }
199 return;
200 }
201
202 /* read warm-up samples */
203 for (i = 1; i <= lpc_order && i < nb_samples; i++)
204 buffer_out[i] =
sign_extend(buffer_out[i - 1] + error_buffer[i], bps);
205
206 /* NOTE: 4 and 8 are very common cases that could be optimized. */
207
208 for (; i < nb_samples; i++) {
209 int j;
211 int error_val = error_buffer[i];
212 int error_sign;
213 int d = *pred++;
214
215 /* LPC prediction */
216 for (j = 0; j < lpc_order; j++)
217 val += (pred[j] - d) * lpc_coefs[j];
218 val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
219 val += d + error_val;
221
222 /* adapt LPC coefficients */
224 if (error_sign) {
225 for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
226 int sign;
227 val = d - pred[j];
229 lpc_coefs[j] -= sign;
230 val *= sign;
231 error_val -= (val >> lpc_quant) * (j + 1);
232 }
233 }
234 }
235 }
236
239 {
241 int has_size,
bps, is_compressed, decorr_shift, decorr_left_weight, ret;
242 uint32_t output_samples;
244
245 skip_bits(&alac->
gb, 4);
/* element instance tag */
247
248 /* the number of output samples is stored in the frame */
250
256 }
257
258 /* whether the frame is compressed */
260
261 if (has_size)
263 else
267 output_samples);
269 }
272 /* get output buffer */
275 return ret;
276 }
else if (output_samples != alac->
nb_samples) {
280 }
285 }
286
287 if (is_compressed) {
288 int16_t lpc_coefs[2][32];
289 int lpc_order[2];
290 int prediction_type[2];
291 int lpc_quant[2];
292 int rice_history_mult[2];
293
296 "Compression with rice limit 0");
298 }
299
302
308
311
312 /* read the predictor table */
313 for (i = lpc_order[ch] - 1; i >= 0; i--)
315 }
316
320 return -1;
323 }
324 }
329 if(ret<0)
330 return ret;
331
332 /* adaptive FIR filter */
333 if (prediction_type[ch] == 15) {
334 /* Prediction type 15 runs the adaptive FIR twice.
335 * The first pass uses the special-case coef_num = 31, while
336 * the second pass uses the coefs from the bitstream.
337 *
338 * However, this prediction type is not currently used by the
339 * reference encoder.
340 */
344 } else if (prediction_type[ch] > 0) {
346 prediction_type[ch]);
347 }
350 bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
351 }
352 } else {
353 /* not compressed, easy case */
356 return -1;
360 }
361 }
363 decorr_shift = 0;
364 decorr_left_weight = 0;
365 }
366
367 if (channels == 2) {
371 }
372
373 if (decorr_left_weight) {
375 decorr_shift, decorr_left_weight);
376 }
377
381 }
385 }
386
388 case 16: {
390 int16_t *outbuffer = (int16_t *)frame->
extended_data[ch_index + ch];
393 }}
394 break;
395 case 20: {
399 }}
400 break;
401 case 24: {
405 }}
406 break;
407 }
408
409 return 0;
410 }
411
413 int *got_frame_ptr,
AVPacket *avpkt)
414 {
419 int ch, ret, got_end;
420
422 return ret;
423
424 got_end = 0;
426 ch = 0;
430 got_end = 1;
431 break;
432 }
436 }
437
438 channels = (element ==
TYPE_CPE) ? 2 : 1;
439 if (ch + channels > alac->
channels ||
443 }
444
447 channels);
449 return ret;
450
452 }
453 if (!got_end) {
456 }
457
461 }
462
464 *got_frame_ptr = 1;
465 else
467
469 }
470
472 {
474
481 }
482
483 return 0;
484 }
485
487 {
490
491 for (ch = 0; ch < 2; ch++) {
495 }
496
499 buf_size, buf_alloc_fail);
500
505 }
506
509 }
510 return 0;
511 buf_alloc_fail:
514 }
515
517 {
519
522
524
529 "max samples per frame invalid: %"PRIu32"\n",
532 }
537 alac->
rice_limit = bytestream2_get_byteu(&gb);
538 alac->
channels = bytestream2_get_byteu(&gb);
539 bytestream2_get_be16u(&gb); // maxRun
540 bytestream2_get_be32u(&gb); // max coded frame size
541 bytestream2_get_be32u(&gb); // average bitrate
543
544 return 0;
545 }
546
548 {
549 int ret;
552
553 /* initialize from the extradata */
557 }
560 return -1;
561 }
562
565 break;
566 case 20:
567 case 24:
569 break;
572 }
575
579 } else {
582 else
584 }
589 }
591
594 return ret;
595 }
596
598
599 return 0;
600 }
601
602 #if HAVE_THREADS
604 {
608 }
609 #endif
610
612 { "extra_bits_bug", "Force non-standard decoding process",
616 };
617
623 };
624
636 .priv_class = &alac_class
637 };
const char const char void * val
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
int nb_samples
number of samples in the current frame
This structure describes decoded (raw) audio or video data.
#define ALAC_EXTRADATA_SIZE
ptrdiff_t const GLvoid * data
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static int init_thread_copy(AVCodecContext *avctx)
static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, int channels)
#define LIBAVUTIL_VERSION_INT
static av_cold int init(AVCodecContext *avctx)
#define AV_OPT_FLAG_AUDIO_PARAM
const char * av_default_item_name(void *ptr)
Return the context name.
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
static const AVOption options[]
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
int32_t * extra_bits_buffer[2]
static int get_sbits(GetBitContext *s, int n)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static int get_sbits_long(GetBitContext *s, int n)
Read 0-32 bits as a signed integer.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
int32_t * predict_error_buffer[2]
static int get_unary_0_9(GetBitContext *gb)
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
uint8_t rice_initial_history
static av_cold int alac_decode_close(AVCodecContext *avctx)
Multithreading support functions.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static int get_bits_count(const GetBitContext *s)
static av_always_inline void bytestream2_skipu(GetByteContext *g, unsigned int size)
static const AVClass alac_class
bitstream reader API header.
int32_t * output_samples_buffer[2]
static int alac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
int extra_bits
number of extra bits beyond 16-bit
static int get_bits_left(GetBitContext *gb)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out, int nb_samples, int bps, int16_t *lpc_coefs, int lpc_order, int lpc_quant)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static int sign_only(int v)
const char * name
Name of the codec implementation.
#define AV_CODEC_CAP_FRAME_THREADS
Codec supports frame-level multithreading.
uint64_t channel_layout
Audio channel layout.
#define ALAC_MAX_CHANNELS
uint32_t max_samples_per_frame
#define ONLY_IF_THREADS_ENABLED(x)
Define a function with only the non-default version specified.
audio channel layout utility functions
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
static int alac_set_info(ALACContext *alac)
uint8_t rice_history_mult
const uint64_t ff_alac_channel_layouts[ALAC_MAX_CHANNELS+1]
static const float pred[4]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Libavcodec external API header.
int sample_rate
samples per second
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
int ff_thread_get_buffer(AVCodecContext *avctx, ThreadFrame *f, int flags)
Wrapper around get_buffer() for frame-multithreaded codecs.
main external API structure.
static unsigned int get_bits1(GetBitContext *s)
Describe the class of an AVClass context structure.
static void skip_bits(GetBitContext *s, int n)
static unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
void(* append_extra_bits[2])(int32_t *buffer[2], int32_t *extra_bits_buffer[2], int extra_bits, int channels, int nb_samples)
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
const uint8_t ff_alac_channel_layout_offsets[ALAC_MAX_CHANNELS][ALAC_MAX_CHANNELS]
static av_const int sign_extend(int val, unsigned bits)
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
common internal api header.
static av_cold int alac_decode_init(AVCodecContext *avctx)
#define FF_ALLOC_OR_GOTO(ctx, p, size, label)
void(* decorrelate_stereo)(int32_t *buffer[2], int nb_samples, int decorr_shift, int decorr_left_weight)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
static int allocate_buffers(ALACContext *alac)
int channels
number of audio channels
static int rice_decompress(ALACContext *alac, int32_t *output_buffer, int nb_samples, int bps, int rice_history_mult)
uint8_t ** extended_data
pointers to the data planes/channels.
av_cold void ff_alacdsp_init(ALACDSPContext *c)
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
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