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libopusdec.c
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1 /*
2  * Opus decoder using libopus
3  * Copyright (c) 2012 Nicolas George
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 #include <opus.h>
23 #include <opus_multistream.h>
24 
25 #include "libavutil/internal.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/ffmath.h"
28 #include "libavutil/opt.h"
29 
30 #include "avcodec.h"
31 #include "internal.h"
32 #include "vorbis.h"
33 #include "mathops.h"
34 #include "libopus.h"
35 
37  AVClass *class;
38  OpusMSDecoder *dec;
39  int pre_skip;
40 #ifndef OPUS_SET_GAIN
41  union { int i; double d; } gain;
42 #endif
43 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
44  int apply_phase_inv;
45 #endif
46 };
47 
48 #define OPUS_HEAD_SIZE 19
49 
51 {
52  struct libopus_context *opus = avc->priv_data;
53  int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled;
54  uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
55 
56  avc->channels = avc->extradata_size >= 10 ? avc->extradata[9] : (avc->channels == 1) ? 1 : 2;
57  if (avc->channels <= 0) {
59  "Invalid number of channels %d, defaulting to stereo\n", avc->channels);
60  avc->channels = 2;
61  }
62 
63  avc->sample_rate = 48000;
66 
67  if (avc->extradata_size >= OPUS_HEAD_SIZE) {
68  opus->pre_skip = AV_RL16(avc->extradata + 10);
69  gain_db = sign_extend(AV_RL16(avc->extradata + 16), 16);
70  channel_map = AV_RL8 (avc->extradata + 18);
71  }
72  if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) {
74  nb_coupled = avc->extradata[OPUS_HEAD_SIZE + 1];
75  if (nb_streams + nb_coupled != avc->channels)
76  av_log(avc, AV_LOG_WARNING, "Inconsistent channel mapping.\n");
77  mapping = avc->extradata + OPUS_HEAD_SIZE + 2;
78  } else {
79  if (avc->channels > 2 || channel_map) {
80  av_log(avc, AV_LOG_ERROR,
81  "No channel mapping for %d channels.\n", avc->channels);
82  return AVERROR(EINVAL);
83  }
84  nb_streams = 1;
85  nb_coupled = avc->channels > 1;
86  mapping = mapping_arr;
87  }
88 
89  if (channel_map == 1) {
90  avc->channel_layout = avc->channels > 8 ? 0 :
92  if (avc->channels > 2 && avc->channels <= 8) {
93  const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
94  int ch;
95 
96  /* Remap channels from Vorbis order to ffmpeg order */
97  for (ch = 0; ch < avc->channels; ch++)
98  mapping_arr[ch] = mapping[vorbis_offset[ch]];
99  mapping = mapping_arr;
100  }
101  } else if (channel_map == 2) {
102  int ambisonic_order = ff_sqrt(avc->channels) - 1;
103  if (avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) &&
104  avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) + 2) {
105  av_log(avc, AV_LOG_ERROR,
106  "Channel mapping 2 is only specified for channel counts"
107  " which can be written as (n + 1)^2 or (n + 2)^2 + 2"
108  " for nonnegative integer n\n");
109  return AVERROR_INVALIDDATA;
110  }
111  if (avc->channels > 227) {
112  av_log(avc, AV_LOG_ERROR, "Too many channels\n");
113  return AVERROR_INVALIDDATA;
114  }
115  avc->channel_layout = 0;
116  } else {
117  avc->channel_layout = 0;
118  }
119 
120  opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels,
121  nb_streams, nb_coupled,
122  mapping, &ret);
123  if (!opus->dec) {
124  av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",
125  opus_strerror(ret));
126  return ff_opus_error_to_averror(ret);
127  }
128 
129 #ifdef OPUS_SET_GAIN
130  ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db));
131  if (ret != OPUS_OK)
132  av_log(avc, AV_LOG_WARNING, "Failed to set gain: %s\n",
133  opus_strerror(ret));
134 #else
135  {
136  double gain_lin = ff_exp10(gain_db / (20.0 * 256));
137  if (avc->sample_fmt == AV_SAMPLE_FMT_FLT)
138  opus->gain.d = gain_lin;
139  else
140  opus->gain.i = FFMIN(gain_lin * 65536, INT_MAX);
141  }
142 #endif
143 
144 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
145  ret = opus_multistream_decoder_ctl(opus->dec,
146  OPUS_SET_PHASE_INVERSION_DISABLED(!opus->apply_phase_inv));
147  if (ret != OPUS_OK)
148  av_log(avc, AV_LOG_WARNING,
149  "Unable to set phase inversion: %s\n",
150  opus_strerror(ret));
151 #endif
152 
153  /* Decoder delay (in samples) at 48kHz */
154  avc->delay = avc->internal->skip_samples = opus->pre_skip;
155 
156  return 0;
157 }
158 
160 {
161  struct libopus_context *opus = avc->priv_data;
162 
163  if (opus->dec) {
164  opus_multistream_decoder_destroy(opus->dec);
165  opus->dec = NULL;
166  }
167  return 0;
168 }
169 
170 #define MAX_FRAME_SIZE (960 * 6)
171 
172 static int libopus_decode(AVCodecContext *avc, void *data,
173  int *got_frame_ptr, AVPacket *pkt)
174 {
175  struct libopus_context *opus = avc->priv_data;
176  AVFrame *frame = data;
177  int ret, nb_samples;
178 
179  frame->nb_samples = MAX_FRAME_SIZE;
180  if ((ret = ff_get_buffer(avc, frame, 0)) < 0)
181  return ret;
182 
183  if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
184  nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
185  (opus_int16 *)frame->data[0],
186  frame->nb_samples, 0);
187  else
188  nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
189  (float *)frame->data[0],
190  frame->nb_samples, 0);
191 
192  if (nb_samples < 0) {
193  av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
194  opus_strerror(nb_samples));
195  return ff_opus_error_to_averror(nb_samples);
196  }
197 
198 #ifndef OPUS_SET_GAIN
199  {
200  int i = avc->channels * nb_samples;
201  if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
202  float *pcm = (float *)frame->data[0];
203  for (; i > 0; i--, pcm++)
204  *pcm = av_clipf(*pcm * opus->gain.d, -1, 1);
205  } else {
206  int16_t *pcm = (int16_t *)frame->data[0];
207  for (; i > 0; i--, pcm++)
208  *pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16);
209  }
210  }
211 #endif
212 
213  frame->nb_samples = nb_samples;
214  *got_frame_ptr = 1;
215 
216  return pkt->size;
217 }
218 
219 static void libopus_flush(AVCodecContext *avc)
220 {
221  struct libopus_context *opus = avc->priv_data;
222 
223  opus_multistream_decoder_ctl(opus->dec, OPUS_RESET_STATE);
224  /* The stream can have been extracted by a tool that is not Opus-aware.
225  Therefore, any packet can become the first of the stream. */
226  avc->internal->skip_samples = opus->pre_skip;
227 }
228 
229 
230 #define OFFSET(x) offsetof(struct libopus_context, x)
231 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
232 static const AVOption libopusdec_options[] = {
233 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
234  { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS },
235 #endif
236  { NULL },
237 };
238 
239 static const AVClass libopusdec_class = {
240  .class_name = "libopusdec",
241  .item_name = av_default_item_name,
242  .option = libopusdec_options,
243  .version = LIBAVUTIL_VERSION_INT,
244 };
245 
246 
248  .name = "libopus",
249  .long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
250  .type = AVMEDIA_TYPE_AUDIO,
251  .id = AV_CODEC_ID_OPUS,
252  .priv_data_size = sizeof(struct libopus_context),
253  .init = libopus_decode_init,
254  .close = libopus_decode_close,
255  .decode = libopus_decode,
256  .flush = libopus_flush,
257  .capabilities = AV_CODEC_CAP_DR1,
258  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
259  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
262  .priv_class = &libopusdec_class,
263  .wrapper_name = "libopus",
264 };
#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:48
static int libopus_decode(AVCodecContext *avc, void *data, int *got_frame_ptr, AVPacket *pkt)
Definition: libopusdec.c:172
#define NULL
Definition: coverity.c:32
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:226
AVOption.
Definition: opt.h:246
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
#define OFFSET(x)
Definition: libopusdec.c:230
int size
Definition: avcodec.h:1453
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
static AVPacket pkt
AVCodec.
Definition: avcodec.h:3441
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
Definition: bytestream.h:87
AVCodec ff_libopus_decoder
Definition: libopusdec.c:247
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
int ff_opus_error_to_averror(int err)
Definition: libopus.c:28
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2204
uint8_t
static int nb_streams
Definition: ffprobe.c:276
#define av_cold
Definition: attributes.h:82
AVOptions.
#define MAX_FRAME_SIZE
Definition: libopusdec.c:170
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1641
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1452
#define av_log(a,...)
#define ff_sqrt
Definition: mathops.h:206
#define AV_RL8(x)
Definition: intreadwrite.h:398
OpusMSDecoder * dec
Definition: libopusdec.c:38
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define AVERROR(e)
Definition: error.h:43
enum AVSampleFormat request_sample_fmt
desired sample format
Definition: avcodec.h:2269
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * name
Name of the codec implementation.
Definition: avcodec.h:3448
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2247
common internal API header
#define FFMIN(a, b)
Definition: common.h:96
static av_cold int libopus_decode_init(AVCodecContext *avc)
Definition: libopusdec.c:50
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int sample_rate
samples per second
Definition: avcodec.h:2196
main external API structure.
Definition: avcodec.h:1540
const uint64_t ff_vorbis_channel_layouts[9]
Definition: vorbis_data.c:47
static void libopus_flush(AVCodecContext *avc)
Definition: libopusdec.c:219
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1918
int extradata_size
Definition: avcodec.h:1642
union libopus_context::@107 gain
Describe the class of an AVClass context structure.
Definition: log.h:67
static const AVClass libopusdec_class
Definition: libopusdec.c:239
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
Definition: internal.h:185
#define OPUS_HEAD_SIZE
Definition: libopusdec.c:48
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:240
internal math functions header
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
static const AVOption libopusdec_options[]
Definition: libopusdec.c:232
void * priv_data
Definition: avcodec.h:1567
int channels
number of audio channels
Definition: avcodec.h:2197
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:1575
static av_cold int libopus_decode_close(AVCodecContext *avc)
Definition: libopusdec.c:159
const uint8_t ff_vorbis_channel_layout_offsets[8][8]
Definition: vorbis_data.c:25
#define FLAGS
Definition: libopusdec.c:231
This structure stores compressed data.
Definition: avcodec.h:1429
int delay
Codec delay.
Definition: avcodec.h:1696
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:968
for(j=16;j >0;--j)
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