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af_aresample.c
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1 /*
2  * Copyright (c) 2011 Stefano Sabatini
3  * Copyright (c) 2011 Mina Nagy Zaki
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  * resampling audio filter
25  */
26 
27 #include "libavutil/avstring.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/samplefmt.h"
31 #include "libavutil/avassert.h"
33 #include "avfilter.h"
34 #include "audio.h"
35 #include "internal.h"
36 
37 typedef struct AResampleContext {
38  const AVClass *class;
40  double ratio;
41  struct SwrContext *swr;
42  int64_t next_pts;
43  int more_data;
45 
47 {
48  AResampleContext *aresample = ctx->priv;
49  int ret = 0;
50 
51  aresample->next_pts = AV_NOPTS_VALUE;
52  aresample->swr = swr_alloc();
53  if (!aresample->swr) {
54  ret = AVERROR(ENOMEM);
55  goto end;
56  }
57 
58  if (opts) {
60 
61  while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
62  if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
63  goto end;
64  }
65  av_dict_free(opts);
66  }
67  if (aresample->sample_rate_arg > 0)
68  av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
69 end:
70  return ret;
71 }
72 
74 {
75  AResampleContext *aresample = ctx->priv;
76  swr_free(&aresample->swr);
77 }
78 
80 {
81  AResampleContext *aresample = ctx->priv;
82  enum AVSampleFormat out_format;
83  int64_t out_rate, out_layout;
84 
85  AVFilterLink *inlink = ctx->inputs[0];
86  AVFilterLink *outlink = ctx->outputs[0];
87 
88  AVFilterFormats *in_formats, *out_formats;
89  AVFilterFormats *in_samplerates, *out_samplerates;
90  AVFilterChannelLayouts *in_layouts, *out_layouts;
91  int ret;
92 
93  av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
94  av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
95  av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
96 
97  in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
98  if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0)
99  return ret;
100 
101  in_samplerates = ff_all_samplerates();
102  if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0)
103  return ret;
104 
105  in_layouts = ff_all_channel_counts();
106  if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0)
107  return ret;
108 
109  if(out_rate > 0) {
110  int ratelist[] = { out_rate, -1 };
111  out_samplerates = ff_make_format_list(ratelist);
112  } else {
113  out_samplerates = ff_all_samplerates();
114  }
115 
116  if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0)
117  return ret;
118 
119  if(out_format != AV_SAMPLE_FMT_NONE) {
120  int formatlist[] = { out_format, -1 };
121  out_formats = ff_make_format_list(formatlist);
122  } else
123  out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
124  if ((ret = ff_formats_ref(out_formats, &outlink->in_formats)) < 0)
125  return ret;
126 
127  if(out_layout) {
128  int64_t layout_list[] = { out_layout, -1 };
129  out_layouts = avfilter_make_format64_list(layout_list);
130  } else
131  out_layouts = ff_all_channel_counts();
132 
133  return ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
134 }
135 
136 
137 static int config_output(AVFilterLink *outlink)
138 {
139  int ret;
140  AVFilterContext *ctx = outlink->src;
141  AVFilterLink *inlink = ctx->inputs[0];
142  AResampleContext *aresample = ctx->priv;
143  int64_t out_rate, out_layout;
144  enum AVSampleFormat out_format;
145  char inchl_buf[128], outchl_buf[128];
146 
147  aresample->swr = swr_alloc_set_opts(aresample->swr,
148  outlink->channel_layout, outlink->format, outlink->sample_rate,
149  inlink->channel_layout, inlink->format, inlink->sample_rate,
150  0, ctx);
151  if (!aresample->swr)
152  return AVERROR(ENOMEM);
153  if (!inlink->channel_layout)
154  av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
155  if (!outlink->channel_layout)
156  av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
157 
158  ret = swr_init(aresample->swr);
159  if (ret < 0)
160  return ret;
161 
162  av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
163  av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
164  av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
165  outlink->time_base = (AVRational) {1, out_rate};
166 
167  av_assert0(outlink->sample_rate == out_rate);
168  av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
169  av_assert0(outlink->format == out_format);
170 
171  aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
172 
173  av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
174  av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
175 
176  av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
177  inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
178  outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
179  return 0;
180 }
181 
182 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
183 {
184  AResampleContext *aresample = inlink->dst->priv;
185  const int n_in = insamplesref->nb_samples;
186  int64_t delay;
187  int n_out = n_in * aresample->ratio + 32;
188  AVFilterLink *const outlink = inlink->dst->outputs[0];
189  AVFrame *outsamplesref;
190  int ret;
191 
192  delay = swr_get_delay(aresample->swr, outlink->sample_rate);
193  if (delay > 0)
194  n_out += FFMIN(delay, FFMAX(4096, n_out));
195 
196  outsamplesref = ff_get_audio_buffer(outlink, n_out);
197 
198  if(!outsamplesref) {
199  av_frame_free(&insamplesref);
200  return AVERROR(ENOMEM);
201  }
202 
203  av_frame_copy_props(outsamplesref, insamplesref);
204  outsamplesref->format = outlink->format;
205  outsamplesref->channels = outlink->channels;
206  outsamplesref->channel_layout = outlink->channel_layout;
207  outsamplesref->sample_rate = outlink->sample_rate;
208 
209  if(insamplesref->pts != AV_NOPTS_VALUE) {
210  int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
211  int64_t outpts= swr_next_pts(aresample->swr, inpts);
212  aresample->next_pts =
213  outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
214  } else {
215  outsamplesref->pts = AV_NOPTS_VALUE;
216  }
217  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
218  (void *)insamplesref->extended_data, n_in);
219  if (n_out <= 0) {
220  av_frame_free(&outsamplesref);
221  av_frame_free(&insamplesref);
222  return 0;
223  }
224 
225  aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
226 
227  outsamplesref->nb_samples = n_out;
228 
229  ret = ff_filter_frame(outlink, outsamplesref);
230  av_frame_free(&insamplesref);
231  return ret;
232 }
233 
234 static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
235 {
236  AVFilterContext *ctx = outlink->src;
237  AResampleContext *aresample = ctx->priv;
238  AVFilterLink *const inlink = outlink->src->inputs[0];
239  AVFrame *outsamplesref;
240  int n_out = 4096;
241  int64_t pts;
242 
243  outsamplesref = ff_get_audio_buffer(outlink, n_out);
244  *outsamplesref_ret = outsamplesref;
245  if (!outsamplesref)
246  return AVERROR(ENOMEM);
247 
248  pts = swr_next_pts(aresample->swr, INT64_MIN);
249  pts = ROUNDED_DIV(pts, inlink->sample_rate);
250 
251  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
252  if (n_out <= 0) {
253  av_frame_free(&outsamplesref);
254  return (n_out == 0) ? AVERROR_EOF : n_out;
255  }
256 
257  outsamplesref->sample_rate = outlink->sample_rate;
258  outsamplesref->nb_samples = n_out;
259 
260  outsamplesref->pts = pts;
261 
262  return 0;
263 }
264 
265 static int request_frame(AVFilterLink *outlink)
266 {
267  AVFilterContext *ctx = outlink->src;
268  AResampleContext *aresample = ctx->priv;
269  int ret;
270 
271  // First try to get data from the internal buffers
272  if (aresample->more_data) {
273  AVFrame *outsamplesref;
274 
275  if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
276  return ff_filter_frame(outlink, outsamplesref);
277  }
278  }
279  aresample->more_data = 0;
280 
281  // Second request more data from the input
282  ret = ff_request_frame(ctx->inputs[0]);
283 
284  // Third if we hit the end flush
285  if (ret == AVERROR_EOF) {
286  AVFrame *outsamplesref;
287 
288  if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
289  return ret;
290 
291  return ff_filter_frame(outlink, outsamplesref);
292  }
293  return ret;
294 }
295 
296 static const AVClass *resample_child_class_next(const AVClass *prev)
297 {
298  return prev ? NULL : swr_get_class();
299 }
300 
301 static void *resample_child_next(void *obj, void *prev)
302 {
303  AResampleContext *s = obj;
304  return prev ? NULL : s->swr;
305 }
306 
307 #define OFFSET(x) offsetof(AResampleContext, x)
308 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
309 
310 static const AVOption options[] = {
311  {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
312  {NULL}
313 };
314 
315 static const AVClass aresample_class = {
316  .class_name = "aresample",
317  .item_name = av_default_item_name,
318  .option = options,
319  .version = LIBAVUTIL_VERSION_INT,
320  .child_class_next = resample_child_class_next,
322 };
323 
324 static const AVFilterPad aresample_inputs[] = {
325  {
326  .name = "default",
327  .type = AVMEDIA_TYPE_AUDIO,
328  .filter_frame = filter_frame,
329  },
330  { NULL }
331 };
332 
333 static const AVFilterPad aresample_outputs[] = {
334  {
335  .name = "default",
336  .config_props = config_output,
337  .request_frame = request_frame,
338  .type = AVMEDIA_TYPE_AUDIO,
339  },
340  { NULL }
341 };
342 
344  .name = "aresample",
345  .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
346  .init_dict = init_dict,
347  .uninit = uninit,
348  .query_formats = query_formats,
349  .priv_size = sizeof(AResampleContext),
350  .priv_class = &aresample_class,
351  .inputs = aresample_inputs,
352  .outputs = aresample_outputs,
353 };
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:768
This structure describes decoded (raw) audio or video data.
Definition: frame.h:201
AVOption.
Definition: opt.h:246
static const AVFilterPad aresample_inputs[]
Definition: af_aresample.c:324
#define LIBAVUTIL_VERSION_INT
Definition: version.h:86
Main libavfilter public API header.
int num
Numerator.
Definition: rational.h:59
int64_t swr_next_pts(struct SwrContext *s, int64_t pts)
Convert the next timestamp from input to output timestamps are in 1/(in_sample_rate * out_sample_rate...
Definition: swresample.c:909
static const AVOption options[]
Definition: af_aresample.c:310
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
const char * name
Pad name.
Definition: internal.h:60
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
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
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:435
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1151
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:149
#define av_cold
Definition: attributes.h:82
AVOptions.
#define FLAGS
Definition: af_aresample.c:308
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:294
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:40
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
#define av_log(a,...)
AVFilterFormats * ff_all_formats(enum AVMediaType type)
Return a list of all formats supported by FFmpeg for the given media type.
Definition: formats.c:350
#define ROUNDED_DIV(a, b)
Definition: common.h:56
A filter pad used for either input or output.
Definition: internal.h:54
libswresample public header
static void * resample_child_next(void *obj, void *prev)
Definition: af_aresample.c:301
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
av_default_item_name
#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:163
The libswresample context.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:179
void * priv
private data for use by the filter
Definition: avfilter.h:353
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:203
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:558
simple assert() macros that are a bit more flexible than ISO C assert().
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:49
int64_t swr_get_delay(struct SwrContext *s, int64_t base)
Gets the delay the next input sample will experience relative to the next output sample.
Definition: swresample.c:859
#define FFMAX(a, b)
Definition: common.h:94
uint64_t channel_layout
Channel layout of the audio data.
Definition: frame.h:379
AVDictionary * opts
Definition: movenc.c:50
int av_opt_get_sample_fmt(void *obj, const char *name, int search_flags, enum AVSampleFormat *out_fmt)
Definition: opt.c:972
int channels
number of audio channels, only used for audio.
Definition: frame.h:506
audio channel layout utility functions
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:129
#define FFMIN(a, b)
Definition: common.h:96
int ff_formats_ref(AVFilterFormats *f, AVFilterFormats **ref)
Add *ref as a new reference to formats.
Definition: formats.c:440
static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
Definition: af_aresample.c:234
static const AVClass aresample_class
Definition: af_aresample.c:315
static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
Definition: af_aresample.c:46
AVFormatContext * ctx
Definition: movenc.c:48
struct SwrContext * swr_alloc_set_opts(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx)
Allocate SwrContext if needed and set/reset common parameters.
Definition: swresample.c:59
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:389
const AVClass * swr_get_class(void)
Get the AVClass for SwrContext.
Definition: options.c:144
A list of supported channel layouts.
Definition: formats.h:85
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.
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_aresample.c:73
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:274
int av_opt_get_int(void *obj, const char *name, int search_flags, int64_t *out_val)
Definition: opt.c:875
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:379
int64_t outpts
output PTS
AVFilterChannelLayouts * avfilter_make_format64_list(const int64_t *fmts)
Definition: formats.c:303
AVFilter ff_af_aresample
Definition: af_aresample.c:343
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
static const AVFilterPad aresample_outputs[]
Definition: af_aresample.c:333
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:374
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
struct SwrContext * swr
Definition: af_aresample.c:41
const char * name
Filter name.
Definition: avfilter.h:148
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)
Definition: swresample.c:706
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int64_t pts
Global timestamp for the audio frames.
static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
Definition: af_aresample.c:182
static const AVClass * resample_child_class_next(const AVClass *prev)
Definition: af_aresample.c:296
void *(* child_next)(void *obj, void *prev)
Return next AVOptions-enabled child or NULL.
Definition: log.h:113
char * key
Definition: dict.h:86
int den
Denominator.
Definition: rational.h:60
#define OFFSET(x)
Definition: af_aresample.c:307
char * value
Definition: dict.h:87
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
#define AV_DICT_IGNORE_SUFFIX
Return first entry in a dictionary whose first part corresponds to the search key, ignoring the suffix of the found key string.
Definition: dict.h:70
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:405
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
static int query_formats(AVFilterContext *ctx)
Definition: af_aresample.c:79
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:449
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:600
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
static int request_frame(AVFilterLink *outlink)
Definition: af_aresample.c:265
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152
static int config_output(AVFilterLink *outlink)
Definition: af_aresample.c:137