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fifo.c
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1 /*
2  * Copyright (c) 2007 Bobby Bingham
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 /**
22  * @file
23  * FIFO buffering filter
24  */
25 
26 #include "libavutil/avassert.h"
28 #include "libavutil/common.h"
29 #include "libavutil/mathematics.h"
30 #include "libavutil/samplefmt.h"
31 
32 #include "audio.h"
33 #include "avfilter.h"
34 #include "internal.h"
35 #include "video.h"
36 
37 typedef struct Buf {
39  struct Buf *next;
40 } Buf;
41 
42 typedef struct FifoContext {
44  Buf *last; ///< last buffered frame
45 
46  /**
47  * When a specific number of output samples is requested, the partial
48  * buffer is stored here
49  */
51  int allocated_samples; ///< number of samples out was allocated for
52 } FifoContext;
53 
55 {
56  FifoContext *fifo = ctx->priv;
57  fifo->last = &fifo->root;
58 
59  return 0;
60 }
61 
63 {
64  FifoContext *fifo = ctx->priv;
65  Buf *buf, *tmp;
66 
67  for (buf = fifo->root.next; buf; buf = tmp) {
68  tmp = buf->next;
69  av_frame_free(&buf->frame);
70  av_free(buf);
71  }
72 
73  av_frame_free(&fifo->out);
74 }
75 
76 static int add_to_queue(AVFilterLink *inlink, AVFrame *frame)
77 {
78  FifoContext *fifo = inlink->dst->priv;
79 
80  fifo->last->next = av_mallocz(sizeof(Buf));
81  if (!fifo->last->next) {
82  av_frame_free(&frame);
83  return AVERROR(ENOMEM);
84  }
85 
86  fifo->last = fifo->last->next;
87  fifo->last->frame = frame;
88 
89  return 0;
90 }
91 
92 static void queue_pop(FifoContext *s)
93 {
94  Buf *tmp = s->root.next->next;
95  if (s->last == s->root.next)
96  s->last = &s->root;
97  av_freep(&s->root.next);
98  s->root.next = tmp;
99 }
100 
101 /**
102  * Move data pointers and pts offset samples forward.
103  */
105  int offset)
106 {
107  int nb_channels = link->channels;
109  int planes = planar ? nb_channels : 1;
110  int block_align = av_get_bytes_per_sample(link->format) * (planar ? 1 : nb_channels);
111  int i;
112 
113  av_assert0(frame->nb_samples > offset);
114 
115  for (i = 0; i < planes; i++)
116  frame->extended_data[i] += block_align * offset;
117  if (frame->data != frame->extended_data)
118  memcpy(frame->data, frame->extended_data,
119  FFMIN(planes, FF_ARRAY_ELEMS(frame->data)) * sizeof(*frame->data));
120  frame->linesize[0] -= block_align*offset;
121  frame->nb_samples -= offset;
122 
123  if (frame->pts != AV_NOPTS_VALUE) {
124  frame->pts += av_rescale_q(offset, (AVRational){1, link->sample_rate},
125  link->time_base);
126  }
127 }
128 
130 {
131  int planes = av_sample_fmt_is_planar(frame->format) ?
132  frame->channels : 1;
133  int min_align = 128;
134  int p;
135 
136  for (p = 0; p < planes; p++) {
137  int cur_align = 128;
138  while ((intptr_t)frame->extended_data[p] % cur_align)
139  cur_align >>= 1;
140  if (cur_align < min_align)
141  min_align = cur_align;
142  }
143  return min_align;
144 }
145 
147 {
148  AVFilterLink *link = ctx->outputs[0];
149  FifoContext *s = ctx->priv;
150  AVFrame *head = s->root.next ? s->root.next->frame : NULL;
151  AVFrame *out;
152  int ret;
153 
154  /* if head is NULL then we're flushing the remaining samples in out */
155  if (!head && !s->out)
156  return AVERROR_EOF;
157 
158  if (!s->out &&
159  head->nb_samples >= link->request_samples &&
160  calc_ptr_alignment(head) >= 32) {
161  if (head->nb_samples == link->request_samples) {
162  out = head;
163  queue_pop(s);
164  } else {
165  out = av_frame_clone(head);
166  if (!out)
167  return AVERROR(ENOMEM);
168 
169  out->nb_samples = link->request_samples;
170  buffer_offset(link, head, link->request_samples);
171  }
172  } else {
173  int nb_channels = link->channels;
174 
175  if (!s->out) {
176  s->out = ff_get_audio_buffer(link, link->request_samples);
177  if (!s->out)
178  return AVERROR(ENOMEM);
179 
180  s->out->nb_samples = 0;
181  s->out->pts = head->pts;
183  } else if (link->request_samples != s->allocated_samples) {
184  av_log(ctx, AV_LOG_ERROR, "request_samples changed before the "
185  "buffer was returned.\n");
186  return AVERROR(EINVAL);
187  }
188 
189  while (s->out->nb_samples < s->allocated_samples) {
190  int len;
191 
192  if (!s->root.next) {
193  ret = ff_request_frame(ctx->inputs[0]);
194  if (ret == AVERROR_EOF) {
196  s->out->nb_samples,
197  s->allocated_samples -
198  s->out->nb_samples,
199  nb_channels, link->format);
201  break;
202  } else if (ret < 0)
203  return ret;
204  if (!s->root.next)
205  return 0;
206  }
207  head = s->root.next->frame;
208 
209  len = FFMIN(s->allocated_samples - s->out->nb_samples,
210  head->nb_samples);
211 
213  s->out->nb_samples, 0, len, nb_channels,
214  link->format);
215  s->out->nb_samples += len;
216 
217  if (len == head->nb_samples) {
218  av_frame_free(&head);
219  queue_pop(s);
220  } else {
221  buffer_offset(link, head, len);
222  }
223  }
224  out = s->out;
225  s->out = NULL;
226  }
227  return ff_filter_frame(link, out);
228 }
229 
230 static int request_frame(AVFilterLink *outlink)
231 {
232  FifoContext *fifo = outlink->src->priv;
233  int ret = 0;
234 
235  if (!fifo->root.next) {
236  if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0) {
237  if (ret == AVERROR_EOF && outlink->request_samples)
238  return return_audio_frame(outlink->src);
239  return ret;
240  }
241  if (!fifo->root.next)
242  return 0;
243  }
244 
245  if (outlink->request_samples) {
246  return return_audio_frame(outlink->src);
247  } else {
248  ret = ff_filter_frame(outlink, fifo->root.next->frame);
249  queue_pop(fifo);
250  }
251 
252  return ret;
253 }
254 
256  {
257  .name = "default",
258  .type = AVMEDIA_TYPE_VIDEO,
259  .filter_frame = add_to_queue,
260  },
261  { NULL }
262 };
263 
265  {
266  .name = "default",
267  .type = AVMEDIA_TYPE_VIDEO,
268  .request_frame = request_frame,
269  },
270  { NULL }
271 };
272 
274  .name = "fifo",
275  .description = NULL_IF_CONFIG_SMALL("Buffer input images and send them when they are requested."),
276 
277  .init = init,
278  .uninit = uninit,
279 
280  .priv_size = sizeof(FifoContext),
281 
282  .inputs = avfilter_vf_fifo_inputs,
283  .outputs = avfilter_vf_fifo_outputs,
284 };
285 
287  {
288  .name = "default",
289  .type = AVMEDIA_TYPE_AUDIO,
290  .filter_frame = add_to_queue,
291  },
292  { NULL }
293 };
294 
296  {
297  .name = "default",
298  .type = AVMEDIA_TYPE_AUDIO,
299  .request_frame = request_frame,
300  },
301  { NULL }
302 };
303 
305  .name = "afifo",
306  .description = NULL_IF_CONFIG_SMALL("Buffer input frames and send them when they are requested."),
307 
308  .init = init,
309  .uninit = uninit,
310 
311  .priv_size = sizeof(FifoContext),
312 
313  .inputs = avfilter_af_afifo_inputs,
314  .outputs = avfilter_af_afifo_outputs,
315 };
static int calc_ptr_alignment(AVFrame *frame)
Definition: fifo.c:129
AVFrame * out
When a specific number of output samples is requested, the partial buffer is stored here...
Definition: fifo.c:50
#define NULL
Definition: coverity.c:32
AVFilter ff_vf_fifo
Definition: fifo.c:273
This structure describes decoded (raw) audio or video data.
Definition: frame.h:226
Buf * last
last buffered frame
Definition: fifo.c:44
static av_cold int init(AVFilterContext *ctx)
Definition: fifo.c:54
static const AVFilterPad avfilter_vf_fifo_inputs[]
Definition: fifo.c:255
Main libavfilter public API header.
static void queue_pop(FifoContext *s)
Definition: fifo.c:92
static av_cold void uninit(AVFilterContext *ctx)
Definition: fifo.c:62
static int add_to_queue(AVFilterLink *inlink, AVFrame *frame)
Definition: fifo.c:76
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AVFrame * frame
Definition: fifo.c:38
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define av_cold
Definition: attributes.h:82
Definition: fifo.c:37
AVFilter ff_af_afifo
Definition: fifo.c:304
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:319
static AVFrame * frame
#define AVERROR_EOF
End of file.
Definition: error.h:55
static const AVFilterPad avfilter_vf_fifo_outputs[]
Definition: fifo.c:264
Buf root
Definition: fifo.c:43
#define av_log(a,...)
static const AVFilterPad avfilter_af_afifo_inputs[]
Definition: fifo.c:286
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
A filter pad used for either input or output.
Definition: internal.h:54
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
struct Buf * next
Definition: fifo.c:39
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:237
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
simple assert() macros that are a bit more flexible than ISO C assert().
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
int allocated_samples
number of samples out was allocated for
Definition: fifo.c:51
static const struct @304 planes[]
int channels
number of audio channels, only used for audio.
Definition: frame.h:531
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
AVFormatContext * ctx
Definition: movenc.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
static void buffer_offset(AVFilterLink *link, AVFrame *frame, int offset)
Move data pointers and pts offset samples forward.
Definition: fifo.c:104
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
Definition: frame.c:540
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
#define FF_ARRAY_ELEMS(a)
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:299
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:257
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 const uint8_t **in ch off *out planar
Definition: audioconvert.c:56
int av_samples_copy(uint8_t **dst, uint8_t *const *src, int dst_offset, int src_offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Copy samples from src to dst.
Definition: samplefmt.c:213
void * buf
Definition: avisynth_c.h:690
static int request_frame(AVFilterLink *outlink)
Definition: fifo.c:230
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:240
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
common internal and external API header
if(ret< 0)
Definition: vf_mcdeint.c:279
static int return_audio_frame(AVFilterContext *ctx)
Definition: fifo.c:146
#define av_free(p)
int len
static const AVFilterPad avfilter_af_afifo_outputs[]
Definition: fifo.c:295
An instance of a filter.
Definition: avfilter.h:338
FILE * out
Definition: movenc.c:54
#define av_freep(p)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:407
int nb_channels
internal API functions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:273
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
static uint8_t tmp[11]
Definition: aes_ctr.c:26