Libav
af_resample.c
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1 /*
2  *
3  * This file is part of Libav.
4  *
5  * Libav is free software; you can redistribute it and/or
6  * modify it under the terms of the GNU Lesser General Public
7  * License as published by the Free Software Foundation; either
8  * version 2.1 of the License, or (at your option) any later version.
9  *
10  * Libav is distributed in the hope that it will be useful,
11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13  * Lesser General Public License for more details.
14  *
15  * You should have received a copy of the GNU Lesser General Public
16  * License along with Libav; if not, write to the Free Software
17  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
18  */
19 
25 #include "libavutil/avassert.h"
26 #include "libavutil/avstring.h"
27 #include "libavutil/common.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/mathematics.h"
30 #include "libavutil/opt.h"
31 
33 
34 #include "audio.h"
35 #include "avfilter.h"
36 #include "formats.h"
37 #include "internal.h"
38 
39 typedef struct ResampleContext {
40  const AVClass *class;
43 
44  int64_t next_pts;
45  int64_t next_in_pts;
46 
47  /* set by filter_frame() to signal an output frame to request_frame() */
50 
51 static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
52 {
53  ResampleContext *s = ctx->priv;
54  const AVClass *avr_class = avresample_get_class();
56 
57  while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
58  if (av_opt_find(&avr_class, e->key, NULL, 0,
60  av_dict_set(&s->options, e->key, e->value, 0);
61  }
62 
63  e = NULL;
64  while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
65  av_dict_set(opts, e->key, NULL, 0);
66 
67  /* do not allow the user to override basic format options */
68  av_dict_set(&s->options, "in_channel_layout", NULL, 0);
69  av_dict_set(&s->options, "out_channel_layout", NULL, 0);
70  av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
71  av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
72  av_dict_set(&s->options, "in_sample_rate", NULL, 0);
73  av_dict_set(&s->options, "out_sample_rate", NULL, 0);
74 
75  return 0;
76 }
77 
78 static av_cold void uninit(AVFilterContext *ctx)
79 {
80  ResampleContext *s = ctx->priv;
81 
82  if (s->avr) {
84  avresample_free(&s->avr);
85  }
86  av_dict_free(&s->options);
87 }
88 
90 {
91  AVFilterLink *inlink = ctx->inputs[0];
92  AVFilterLink *outlink = ctx->outputs[0];
93 
96  AVFilterFormats *in_samplerates = ff_all_samplerates();
97  AVFilterFormats *out_samplerates = ff_all_samplerates();
100 
101  ff_formats_ref(in_formats, &inlink->out_formats);
102  ff_formats_ref(out_formats, &outlink->in_formats);
103 
104  ff_formats_ref(in_samplerates, &inlink->out_samplerates);
105  ff_formats_ref(out_samplerates, &outlink->in_samplerates);
106 
107  ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
108  ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
109 
110  return 0;
111 }
112 
113 static int config_output(AVFilterLink *outlink)
114 {
115  AVFilterContext *ctx = outlink->src;
116  AVFilterLink *inlink = ctx->inputs[0];
117  ResampleContext *s = ctx->priv;
118  char buf1[64], buf2[64];
119  int ret;
120 
121  if (s->avr) {
122  avresample_close(s->avr);
123  avresample_free(&s->avr);
124  }
125 
126  if (inlink->channel_layout == outlink->channel_layout &&
127  inlink->sample_rate == outlink->sample_rate &&
128  (inlink->format == outlink->format ||
131  av_get_planar_sample_fmt(inlink->format) ==
132  av_get_planar_sample_fmt(outlink->format))))
133  return 0;
134 
135  if (!(s->avr = avresample_alloc_context()))
136  return AVERROR(ENOMEM);
137 
138  if (s->options) {
139  int ret;
140  AVDictionaryEntry *e = NULL;
141  while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
142  av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
143 
144  ret = av_opt_set_dict(s->avr, &s->options);
145  if (ret < 0)
146  return ret;
147  }
148 
149  av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
150  av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
151  av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
152  av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
153  av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
154  av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
155 
156  if ((ret = avresample_open(s->avr)) < 0)
157  return ret;
158 
159  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
162 
163  av_get_channel_layout_string(buf1, sizeof(buf1),
164  -1, inlink ->channel_layout);
165  av_get_channel_layout_string(buf2, sizeof(buf2),
166  -1, outlink->channel_layout);
167  av_log(ctx, AV_LOG_VERBOSE,
168  "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
169  av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
170  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
171 
172  return 0;
173 }
174 
175 static int request_frame(AVFilterLink *outlink)
176 {
177  AVFilterContext *ctx = outlink->src;
178  ResampleContext *s = ctx->priv;
179  int ret = 0;
180 
181  s->got_output = 0;
182  while (ret >= 0 && !s->got_output)
183  ret = ff_request_frame(ctx->inputs[0]);
184 
185  /* flush the lavr delay buffer */
186  if (ret == AVERROR_EOF && s->avr) {
187  AVFrame *frame;
188  int nb_samples = avresample_get_out_samples(s->avr, 0);
189 
190  if (!nb_samples)
191  return ret;
192 
193  frame = ff_get_audio_buffer(outlink, nb_samples);
194  if (!frame)
195  return AVERROR(ENOMEM);
196 
197  ret = avresample_convert(s->avr, frame->extended_data,
198  frame->linesize[0], nb_samples,
199  NULL, 0, 0);
200  if (ret <= 0) {
201  av_frame_free(&frame);
202  return (ret == 0) ? AVERROR_EOF : ret;
203  }
204 
205  frame->pts = s->next_pts;
206  return ff_filter_frame(outlink, frame);
207  }
208  return ret;
209 }
210 
211 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
212 {
213  AVFilterContext *ctx = inlink->dst;
214  ResampleContext *s = ctx->priv;
215  AVFilterLink *outlink = ctx->outputs[0];
216  int ret;
217 
218  if (s->avr) {
219  AVFrame *out;
220  int delay, nb_samples;
221 
222  /* maximum possible samples lavr can output */
223  delay = avresample_get_delay(s->avr);
224  nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
225 
226  out = ff_get_audio_buffer(outlink, nb_samples);
227  if (!out) {
228  ret = AVERROR(ENOMEM);
229  goto fail;
230  }
231 
232  ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
233  nb_samples, in->extended_data, in->linesize[0],
234  in->nb_samples);
235  if (ret <= 0) {
236  av_frame_free(&out);
237  if (ret < 0)
238  goto fail;
239  }
240 
242 
243  if (s->next_pts == AV_NOPTS_VALUE) {
244  if (in->pts == AV_NOPTS_VALUE) {
245  av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
246  "assuming 0.\n");
247  s->next_pts = 0;
248  } else
249  s->next_pts = av_rescale_q(in->pts, inlink->time_base,
250  outlink->time_base);
251  }
252 
253  if (ret > 0) {
254  out->nb_samples = ret;
255 
256  ret = av_frame_copy_props(out, in);
257  if (ret < 0) {
258  av_frame_free(&out);
259  goto fail;
260  }
261 
262  out->sample_rate = outlink->sample_rate;
263  /* Only convert in->pts if there is a discontinuous jump.
264  This ensures that out->pts tracks the number of samples actually
265  output by the resampler in the absence of such a jump.
266  Otherwise, the rounding in av_rescale_q() and av_rescale()
267  causes off-by-1 errors. */
268  if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
269  out->pts = av_rescale_q(in->pts, inlink->time_base,
270  outlink->time_base) -
271  av_rescale(delay, outlink->sample_rate,
272  inlink->sample_rate);
273  } else
274  out->pts = s->next_pts;
275 
276  s->next_pts = out->pts + out->nb_samples;
277  s->next_in_pts = in->pts + in->nb_samples;
278 
279  ret = ff_filter_frame(outlink, out);
280  s->got_output = 1;
281  }
282 
283 fail:
284  av_frame_free(&in);
285  } else {
286  in->format = outlink->format;
287  ret = ff_filter_frame(outlink, in);
288  s->got_output = 1;
289  }
290 
291  return ret;
292 }
293 
294 static const AVClass *resample_child_class_next(const AVClass *prev)
295 {
296  return prev ? NULL : avresample_get_class();
297 }
298 
299 static void *resample_child_next(void *obj, void *prev)
300 {
301  ResampleContext *s = obj;
302  return prev ? NULL : s->avr;
303 }
304 
305 static const AVClass resample_class = {
306  .class_name = "resample",
307  .item_name = av_default_item_name,
308  .version = LIBAVUTIL_VERSION_INT,
309  .child_class_next = resample_child_class_next,
311 };
312 
314  {
315  .name = "default",
316  .type = AVMEDIA_TYPE_AUDIO,
317  .filter_frame = filter_frame,
318  },
319  { NULL }
320 };
321 
323  {
324  .name = "default",
325  .type = AVMEDIA_TYPE_AUDIO,
326  .config_props = config_output,
327  .request_frame = request_frame
328  },
329  { NULL }
330 };
331 
333  .name = "resample",
334  .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
335  .priv_size = sizeof(ResampleContext),
336  .priv_class = &resample_class,
337 
338  .init_dict = init,
339  .uninit = uninit,
341 
344 };
AVAudioResampleContext * avr
Definition: af_resample.c:41
static const AVFilterPad avfilter_af_resample_outputs[]
Definition: af_resample.c:322
This structure describes decoded (raw) audio or video data.
Definition: frame.h:135
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:129
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:232
Main libavfilter public API header.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
void avresample_free(AVAudioResampleContext **avr)
Free AVAudioResampleContext and associated AVOption values.
Definition: utils.c:278
const AVClass * avresample_get_class(void)
Get the AVClass for AVAudioResampleContext.
Definition: options.c:110
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:38
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:571
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Public dictionary API.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:733
#define av_cold
Definition: attributes.h:66
AVOptions.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:211
int64_t next_pts
Definition: af_resample.c:44
void ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:261
AVDictionaryEntry * av_dict_get(const AVDictionary *m, const char *key, const AVDictionaryEntry *prev, int flags)
Get a dictionary entry with matching key.
Definition: dict.c:38
#define AVERROR_EOF
End of file.
Definition: error.h:51
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:139
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition: samplefmt.c:73
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_resample.c:78
void avresample_close(AVAudioResampleContext *avr)
Close AVAudioResampleContext.
Definition: utils.c:262
AVFilterFormats * ff_all_formats(enum AVMediaType type)
Return a list of all formats supported by Libav for the given media type.
Definition: formats.c:209
static int config_output(AVFilterLink *outlink)
Definition: af_resample.c:113
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:129
void *(* child_next)(void *obj, void *prev)
Return next AVOptions-enabled child or NULL.
Definition: log.h:79
static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
Definition: af_resample.c:51
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:57
#define AVERROR(e)
Definition: error.h:43
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:69
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:145
int64_t next_in_pts
Definition: af_resample.c:45
void * priv
private data for use by the filter
Definition: avfilter.h:584
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:170
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:268
simple assert() macros that are a bit more flexible than ISO C assert().
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:169
AVDictionary * options
Definition: af_resample.c:42
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:47
AVFilter ff_af_resample
Definition: af_resample.c:332
const AVOption * av_opt_find(void *obj, const char *name, const char *unit, int opt_flags, int search_flags)
Look for an option in an object.
Definition: opt.c:700
static const AVClass resample_class
Definition: af_resample.c:305
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:116
external API header
static void * resample_child_next(void *obj, void *prev)
Definition: af_resample.c:299
#define AV_OPT_SEARCH_CHILDREN
Search in possible children of the given object first.
Definition: opt.h:383
LIBAVUTIL_VERSION_INT
Definition: eval.c:55
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout/s...
Definition: formats.c:248
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:186
int avresample_get_delay(AVAudioResampleContext *avr)
Return the number of samples currently in the resampling delay buffer.
Definition: resample.c:495
NULL
Definition: eval.c:55
int avresample_available(AVAudioResampleContext *avr)
Return the number of available samples in the output FIFO.
Definition: utils.c:744
void ff_formats_ref(AVFilterFormats *f, AVFilterFormats **ref)
Add *ref as a new reference to formats.
Definition: formats.c:266
int av_opt_set_dict(void *obj, AVDictionary **options)
Definition: opt.c:679
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:153
av_default_item_name
Definition: dnxhdenc.c:52
int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
Provide the upper bound on the number of samples the configured conversion would output.
Definition: utils.c:749
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:68
Describe the class of an AVClass context structure.
Definition: log.h:33
int sample_rate
Sample rate of the audio data.
Definition: frame.h:376
Filter definition.
Definition: avfilter.h:421
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:221
rational number numerator/denominator
Definition: rational.h:43
int avresample_convert(AVAudioResampleContext *avr, uint8_t **output, int out_plane_size, int out_samples, uint8_t **input, int in_plane_size, int in_samples)
Convert input samples and write them to the output FIFO.
Definition: utils.c:330
const char * name
Filter name.
Definition: avfilter.h:425
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_resample.c:211
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:578
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:242
static int request_frame(AVFilterLink *outlink)
Definition: af_resample.c:175
AVAudioResampleContext * avresample_alloc_context(void)
Allocate AVAudioResampleContext and set options.
Definition: options.c:96
static int query_formats(AVFilterContext *ctx)
Definition: af_resample.c:89
common internal and external API header
#define AV_OPT_SEARCH_FAKE_OBJ
The obj passed to av_opt_find() is fake – only a double pointer to AVClass instead of a required poi...
Definition: opt.h:392
char * key
Definition: dict.h:75
struct AVFilterPad AVFilterPad
Definition: avfilter.h:67
char * value
Definition: dict.h:76
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:563
#define AV_DICT_IGNORE_SUFFIX
Definition: dict.h:62
static const AVClass * resample_child_class_next(const AVClass *prev)
Definition: af_resample.c:294
static const AVFilterPad avfilter_af_resample_inputs[]
Definition: af_resample.c:313
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:249
internal API functions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:169
int avresample_open(AVAudioResampleContext *avr)
Initialize AVAudioResampleContext.
Definition: utils.c:36
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:179
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:367
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:228
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.