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filter_audio.c
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1 /*
2  * copyright (c) 2013 Andrew Kelley
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  * libavfilter API usage example.
24  *
25  * @example filter_audio.c
26  * This example will generate a sine wave audio,
27  * pass it through a simple filter chain, and then compute the MD5 checksum of
28  * the output data.
29  *
30  * The filter chain it uses is:
31  * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
32  *
33  * abuffer: This provides the endpoint where you can feed the decoded samples.
34  * volume: In this example we hardcode it to 0.90.
35  * aformat: This converts the samples to the samplefreq, channel layout,
36  * and sample format required by the audio device.
37  * abuffersink: This provides the endpoint where you can read the samples after
38  * they have passed through the filter chain.
39  */
40 
41 #include <inttypes.h>
42 #include <math.h>
43 #include <stdio.h>
44 #include <stdlib.h>
45 
47 #include "libavutil/md5.h"
48 #include "libavutil/mem.h"
49 #include "libavutil/opt.h"
50 #include "libavutil/samplefmt.h"
51 
52 #include "libavfilter/avfilter.h"
53 #include "libavfilter/buffersink.h"
54 #include "libavfilter/buffersrc.h"
55 
56 #define INPUT_SAMPLERATE 48000
57 #define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
58 #define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
59 
60 #define VOLUME_VAL 0.90
61 
63  AVFilterContext **sink)
64 {
66  AVFilterContext *abuffer_ctx;
67  AVFilter *abuffer;
68  AVFilterContext *volume_ctx;
69  AVFilter *volume;
70  AVFilterContext *aformat_ctx;
71  AVFilter *aformat;
72  AVFilterContext *abuffersink_ctx;
73  AVFilter *abuffersink;
74 
75  AVDictionary *options_dict = NULL;
76  uint8_t options_str[1024];
77  uint8_t ch_layout[64];
78 
79  int err;
80 
81  /* Create a new filtergraph, which will contain all the filters. */
82  filter_graph = avfilter_graph_alloc();
83  if (!filter_graph) {
84  fprintf(stderr, "Unable to create filter graph.\n");
85  return AVERROR(ENOMEM);
86  }
87 
88  /* Create the abuffer filter;
89  * it will be used for feeding the data into the graph. */
90  abuffer = avfilter_get_by_name("abuffer");
91  if (!abuffer) {
92  fprintf(stderr, "Could not find the abuffer filter.\n");
94  }
95 
96  abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
97  if (!abuffer_ctx) {
98  fprintf(stderr, "Could not allocate the abuffer instance.\n");
99  return AVERROR(ENOMEM);
100  }
101 
102  /* Set the filter options through the AVOptions API. */
103  av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
104  av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
106  av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
107  av_opt_set_int(abuffer_ctx, "sample_rate", INPUT_SAMPLERATE, AV_OPT_SEARCH_CHILDREN);
108 
109  /* Now initialize the filter; we pass NULL options, since we have already
110  * set all the options above. */
111  err = avfilter_init_str(abuffer_ctx, NULL);
112  if (err < 0) {
113  fprintf(stderr, "Could not initialize the abuffer filter.\n");
114  return err;
115  }
116 
117  /* Create volume filter. */
118  volume = avfilter_get_by_name("volume");
119  if (!volume) {
120  fprintf(stderr, "Could not find the volume filter.\n");
122  }
123 
124  volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
125  if (!volume_ctx) {
126  fprintf(stderr, "Could not allocate the volume instance.\n");
127  return AVERROR(ENOMEM);
128  }
129 
130  /* A different way of passing the options is as key/value pairs in a
131  * dictionary. */
132  av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
133  err = avfilter_init_dict(volume_ctx, &options_dict);
134  av_dict_free(&options_dict);
135  if (err < 0) {
136  fprintf(stderr, "Could not initialize the volume filter.\n");
137  return err;
138  }
139 
140  /* Create the aformat filter;
141  * it ensures that the output is of the format we want. */
142  aformat = avfilter_get_by_name("aformat");
143  if (!aformat) {
144  fprintf(stderr, "Could not find the aformat filter.\n");
146  }
147 
148  aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
149  if (!aformat_ctx) {
150  fprintf(stderr, "Could not allocate the aformat instance.\n");
151  return AVERROR(ENOMEM);
152  }
153 
154  /* A third way of passing the options is in a string of the form
155  * key1=value1:key2=value2.... */
156  snprintf(options_str, sizeof(options_str),
157  "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
159  (uint64_t)AV_CH_LAYOUT_STEREO);
160  err = avfilter_init_str(aformat_ctx, options_str);
161  if (err < 0) {
162  av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
163  return err;
164  }
165 
166  /* Finally create the abuffersink filter;
167  * it will be used to get the filtered data out of the graph. */
168  abuffersink = avfilter_get_by_name("abuffersink");
169  if (!abuffersink) {
170  fprintf(stderr, "Could not find the abuffersink filter.\n");
172  }
173 
174  abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
175  if (!abuffersink_ctx) {
176  fprintf(stderr, "Could not allocate the abuffersink instance.\n");
177  return AVERROR(ENOMEM);
178  }
179 
180  /* This filter takes no options. */
181  err = avfilter_init_str(abuffersink_ctx, NULL);
182  if (err < 0) {
183  fprintf(stderr, "Could not initialize the abuffersink instance.\n");
184  return err;
185  }
186 
187  /* Connect the filters;
188  * in this simple case the filters just form a linear chain. */
189  err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
190  if (err >= 0)
191  err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
192  if (err >= 0)
193  err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
194  if (err < 0) {
195  fprintf(stderr, "Error connecting filters\n");
196  return err;
197  }
198 
199  /* Configure the graph. */
200  err = avfilter_graph_config(filter_graph, NULL);
201  if (err < 0) {
202  av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
203  return err;
204  }
205 
206  *graph = filter_graph;
207  *src = abuffer_ctx;
208  *sink = abuffersink_ctx;
209 
210  return 0;
211 }
212 
213 /* Do something useful with the filtered data: this simple
214  * example just prints the MD5 checksum of each plane to stdout. */
215 static int process_output(struct AVMD5 *md5, AVFrame *frame)
216 {
217  int planar = av_sample_fmt_is_planar(frame->format);
218  int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
219  int planes = planar ? channels : 1;
220  int bps = av_get_bytes_per_sample(frame->format);
221  int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
222  int i, j;
223 
224  for (i = 0; i < planes; i++) {
225  uint8_t checksum[16];
226 
227  av_md5_init(md5);
228  av_md5_sum(checksum, frame->extended_data[i], plane_size);
229 
230  fprintf(stdout, "plane %d: 0x", i);
231  for (j = 0; j < sizeof(checksum); j++)
232  fprintf(stdout, "%02X", checksum[j]);
233  fprintf(stdout, "\n");
234  }
235  fprintf(stdout, "\n");
236 
237  return 0;
238 }
239 
240 /* Construct a frame of audio data to be filtered;
241  * this simple example just synthesizes a sine wave. */
242 static int get_input(AVFrame *frame, int frame_num)
243 {
244  int err, i, j;
245 
246 #define FRAME_SIZE 1024
247 
248  /* Set up the frame properties and allocate the buffer for the data. */
249  frame->sample_rate = INPUT_SAMPLERATE;
250  frame->format = INPUT_FORMAT;
252  frame->nb_samples = FRAME_SIZE;
253  frame->pts = frame_num * FRAME_SIZE;
254 
255  err = av_frame_get_buffer(frame, 0);
256  if (err < 0)
257  return err;
258 
259  /* Fill the data for each channel. */
260  for (i = 0; i < 5; i++) {
261  float *data = (float*)frame->extended_data[i];
262 
263  for (j = 0; j < frame->nb_samples; j++)
264  data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
265  }
266 
267  return 0;
268 }
269 
270 int main(int argc, char *argv[])
271 {
272  struct AVMD5 *md5;
273  AVFilterGraph *graph;
274  AVFilterContext *src, *sink;
275  AVFrame *frame;
276  uint8_t errstr[1024];
277  float duration;
278  int err, nb_frames, i;
279 
280  if (argc < 2) {
281  fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
282  return 1;
283  }
284 
285  duration = atof(argv[1]);
286  nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
287  if (nb_frames <= 0) {
288  fprintf(stderr, "Invalid duration: %s\n", argv[1]);
289  return 1;
290  }
291 
293 
294  /* Allocate the frame we will be using to store the data. */
295  frame = av_frame_alloc();
296  if (!frame) {
297  fprintf(stderr, "Error allocating the frame\n");
298  return 1;
299  }
300 
301  md5 = av_md5_alloc();
302  if (!md5) {
303  fprintf(stderr, "Error allocating the MD5 context\n");
304  return 1;
305  }
306 
307  /* Set up the filtergraph. */
308  err = init_filter_graph(&graph, &src, &sink);
309  if (err < 0) {
310  fprintf(stderr, "Unable to init filter graph:");
311  goto fail;
312  }
313 
314  /* the main filtering loop */
315  for (i = 0; i < nb_frames; i++) {
316  /* get an input frame to be filtered */
317  err = get_input(frame, i);
318  if (err < 0) {
319  fprintf(stderr, "Error generating input frame:");
320  goto fail;
321  }
322 
323  /* Send the frame to the input of the filtergraph. */
324  err = av_buffersrc_add_frame(src, frame);
325  if (err < 0) {
326  av_frame_unref(frame);
327  fprintf(stderr, "Error submitting the frame to the filtergraph:");
328  goto fail;
329  }
330 
331  /* Get all the filtered output that is available. */
332  while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
333  /* now do something with our filtered frame */
334  err = process_output(md5, frame);
335  if (err < 0) {
336  fprintf(stderr, "Error processing the filtered frame:");
337  goto fail;
338  }
339  av_frame_unref(frame);
340  }
341 
342  if (err == AVERROR(EAGAIN)) {
343  /* Need to feed more frames in. */
344  continue;
345  } else if (err == AVERROR_EOF) {
346  /* Nothing more to do, finish. */
347  break;
348  } else if (err < 0) {
349  /* An error occurred. */
350  fprintf(stderr, "Error filtering the data:");
351  goto fail;
352  }
353  }
354 
355  avfilter_graph_free(&graph);
356  av_frame_free(&frame);
357  av_freep(&md5);
358 
359  return 0;
360 
361 fail:
362  av_strerror(err, errstr, sizeof(errstr));
363  fprintf(stderr, "%s\n", errstr);
364  return 1;
365 }
#define NULL
Definition: coverity.c:32
This structure describes decoded (raw) audio or video data.
Definition: frame.h:187
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
int av_opt_set_q(void *obj, const char *name, AVRational val, int search_flags)
Definition: opt.c:568
AVFilterGraph * avfilter_graph_alloc(void)
Allocate a filter graph.
Definition: avfiltergraph.c:80
Main libavfilter public API header.
Memory handling functions.
Memory buffer source API.
AVFilterGraph * filter_graph
int avfilter_graph_config(AVFilterGraph *graphctx, void *log_ctx)
Check validity and configure all the links and formats in the graph.
void avfilter_graph_free(AVFilterGraph **graph)
Free a graph, destroy its links, and set *graph to NULL.
#define AV_CH_LAYOUT_STEREO
static int get_input(AVFrame *frame, int frame_num)
Definition: filter_audio.c:242
#define src
Definition: vp8dsp.c:254
#define INPUT_SAMPLERATE
Definition: filter_audio.c:56
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
memory buffer sink API for audio and video
int avfilter_link(AVFilterContext *src, unsigned srcpad, AVFilterContext *dst, unsigned dstpad)
Link two filters together.
Definition: avfilter.c:135
struct AVMD5 * av_md5_alloc(void)
Allocate an AVMD5 context.
Definition: md5.c:48
uint8_t
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:150
#define INPUT_FORMAT
Definition: filter_audio.c:57
AVOptions.
#define FRAME_SIZE
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:271
int64_t duration
Definition: movenc.c:63
void avfilter_register_all(void)
Initialize the filter system.
Definition: allfilters.c:382
static AVFrame * frame
struct AVMD5 * md5
Definition: movenc.c:56
#define AVERROR_EOF
End of file.
Definition: error.h:55
void av_md5_sum(uint8_t *dst, const uint8_t *src, const int len)
Hash an array of data.
Definition: md5.c:203
Definition: md5.c:40
#define av_log(a,...)
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
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_YASM &&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
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#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
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
static int process_output(struct AVMD5 *md5, AVFrame *frame)
Definition: filter_audio.c:215
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
#define fail()
Definition: checkasm.h:89
int main(int argc, char *argv[])
Definition: filter_audio.c:270
uint64_t channel_layout
Channel layout of the audio data.
Definition: frame.h:356
const AVFilter * avfilter_get_by_name(const char *name)
Get a filter definition matching the given name.
Definition: avfilter.c:564
audio channel layout utility functions
#define AV_OPT_SEARCH_CHILDREN
Search in possible children of the given object first.
Definition: opt.h:557
static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src, AVFilterContext **sink)
Definition: filter_audio.c:62
static volatile int checksum
Definition: adler32.c:28
int avfilter_init_str(AVFilterContext *filter, const char *args)
Initialize a filter with the supplied parameters.
Definition: avfilter.c:960
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.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:251
void av_md5_init(AVMD5 *ctx)
Initialize MD5 hashing.
Definition: md5.c:143
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:70
#define AV_STRINGIFY(s)
Definition: macros.h:36
#define INPUT_CHANNEL_LAYOUT
Definition: filter_audio.c:58
int sample_rate
Sample rate of the audio data.
Definition: frame.h:351
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
#define snprintf
Definition: snprintf.h:34
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:280
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
Definition: frame.c:498
#define AVERROR_FILTER_NOT_FOUND
Filter not found.
Definition: error.h:58
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
int avfilter_init_dict(AVFilterContext *ctx, AVDictionary **options)
Initialize a filter with the supplied dictionary of options.
Definition: avfilter.c:923
int av_strerror(int errnum, char *errbuf, size_t errbuf_size)
Put a description of the AVERROR code errnum in errbuf.
Definition: error.c:105
signed 16 bits
Definition: samplefmt.h:61
unsigned bps
Definition: movenc.c:1414
#define VOLUME_VAL
Definition: filter_audio.c:60
AVFilterContext * avfilter_graph_alloc_filter(AVFilterGraph *graph, const AVFilter *filter, const char *name)
Create a new filter instance in a filter graph.
An instance of a filter.
Definition: avfilter.h:323
Public header for MD5 hash function implementation.
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:234
int attribute_align_arg av_buffersink_get_frame(AVFilterContext *ctx, AVFrame *frame)
Get a frame with filtered data from sink and put it in frame.
Definition: buffersink.c:67
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:244
for(j=16;j >0;--j)
int attribute_align_arg av_buffersrc_add_frame(AVFilterContext *ctx, AVFrame *frame)
Add a frame to the buffer source.
Definition: buffersrc.c:144