1 /*
2 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3 *
4 * This file is part of libswresample
5 *
6 * libswresample 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 * libswresample 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 libswresample; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #ifndef SWRESAMPLE_SWRESAMPLE_H
22 #define SWRESAMPLE_SWRESAMPLE_H
23
24 /**
25 * @file
26 * @ingroup lswr
27 * libswresample public header
28 */
29
30 /**
31 * @defgroup lswr Libswresample
32 * @{
33 *
34 * Libswresample (lswr) is a library that handles audio resampling, sample
35 * format conversion and mixing.
36 *
37 * Interaction with lswr is done through SwrContext, which is
38 * allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
39 * must be set with the @ref avoptions API.
40 *
41 * The first thing you will need to do in order to use lswr is to allocate
42 * SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts(). If you
43 * are using the former, you must set options through the @ref avoptions API.
44 * The latter function provides the same feature, but it allows you to set some
45 * common options in the same statement.
46 *
47 * For example the following code will setup conversion from planar float sample
48 * format to interleaved signed 16-bit integer, downsampling from 48kHz to
49 * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
50 * matrix). This is using the swr_alloc() function.
51 * @code
52 * SwrContext *swr = swr_alloc();
53 * av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
54 * av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
55 * av_opt_set_int(swr, "in_sample_rate", 48000, 0);
56 * av_opt_set_int(swr, "out_sample_rate", 44100, 0);
57 * av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
58 * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
59 * @endcode
60 *
61 * The same job can be done using swr_alloc_set_opts() as well:
62 * @code
63 * SwrContext *swr = swr_alloc_set_opts(NULL, // we're allocating a new context
64 * AV_CH_LAYOUT_STEREO, // out_ch_layout
65 * AV_SAMPLE_FMT_S16, // out_sample_fmt
66 * 44100, // out_sample_rate
67 * AV_CH_LAYOUT_5POINT1, // in_ch_layout
68 * AV_SAMPLE_FMT_FLTP, // in_sample_fmt
69 * 48000, // in_sample_rate
70 * 0, // log_offset
71 * NULL); // log_ctx
72 * @endcode
73 *
74 * Once all values have been set, it must be initialized with swr_init(). If
75 * you need to change the conversion parameters, you can change the parameters
76 * using @ref AVOptions, as described above in the first example; or by using
77 * swr_alloc_set_opts(), but with the first argument the allocated context.
78 * You must then call swr_init() again.
79 *
80 * The conversion itself is done by repeatedly calling swr_convert().
81 * Note that the samples may get buffered in swr if you provide insufficient
82 * output space or if sample rate conversion is done, which requires "future"
83 * samples. Samples that do not require future input can be retrieved at any
84 * time by using swr_convert() (in_count can be set to 0).
85 * At the end of conversion the resampling buffer can be flushed by calling
86 * swr_convert() with NULL in and 0 in_count.
87 *
88 * The samples used in the conversion process can be managed with the libavutil
89 * @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
90 * function used in the following example.
91 *
92 * The delay between input and output, can at any time be found by using
93 * swr_get_delay().
94 *
95 * The following code demonstrates the conversion loop assuming the parameters
96 * from above and caller-defined functions get_input() and handle_output():
97 * @code
98 * uint8_t **input;
99 * int in_samples;
100 *
101 * while (get_input(&input, &in_samples)) {
102 * uint8_t *output;
103 * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
104 * in_samples, 44100, 48000, AV_ROUND_UP);
105 * av_samples_alloc(&output, NULL, 2, out_samples,
106 * AV_SAMPLE_FMT_S16, 0);
107 * out_samples = swr_convert(swr, &output, out_samples,
108 * input, in_samples);
109 * handle_output(output, out_samples);
110 * av_freep(&output);
111 * }
112 * @endcode
113 *
114 * When the conversion is finished, the conversion
115 * context and everything associated with it must be freed with swr_free().
116 * A swr_close() function is also available, but it exists mainly for
117 * compatibility with libavresample, and is not required to be called.
118 *
119 * There will be no memory leak if the data is not completely flushed before
120 * swr_free().
121 */
122
123 #include <stdint.h>
126
128
129 #if LIBSWRESAMPLE_VERSION_MAJOR < 1
130 #define SWR_CH_MAX 32 ///< Maximum number of channels
131 #endif
132
133 /**
134 * @name Option constants
135 * These constants are used for the @ref avoptions interface for lswr.
136 * @{
137 *
138 */
139
140 #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
141 //TODO use int resample ?
142 //long term TODO can we enable this dynamically?
143
144 /** Dithering algorithms */
150
160 };
161
162 /** Resampling Engines */
167 };
168
169 /** Resampling Filter Types */
174 };
175
176 /**
177 * @}
178 */
179
180 /**
181 * The libswresample context. Unlike libavcodec and libavformat, this structure
182 * is opaque. This means that if you would like to set options, you must use
183 * the @ref avoptions API and cannot directly set values to members of the
184 * structure.
185 */
187
188 /**
189 * Get the AVClass for SwrContext. It can be used in combination with
190 * AV_OPT_SEARCH_FAKE_OBJ for examining options.
191 *
192 * @see av_opt_find().
193 * @return the AVClass of SwrContext
194 */
196
197 /**
198 * @name SwrContext constructor functions
199 * @{
200 */
201
202 /**
203 * Allocate SwrContext.
204 *
205 * If you use this function you will need to set the parameters (manually or
206 * with swr_alloc_set_opts()) before calling swr_init().
207 *
208 * @see swr_alloc_set_opts(), swr_init(), swr_free()
209 * @return NULL on error, allocated context otherwise
210 */
212
213 /**
214 * Initialize context after user parameters have been set.
215 * @note The context must be configured using the AVOption API.
216 *
217 * @see av_opt_set_int()
218 * @see av_opt_set_dict()
219 *
220 * @param[in,out] s Swr context to initialize
221 * @return AVERROR error code in case of failure.
222 */
224
225 /**
226 * Check whether an swr context has been initialized or not.
227 *
228 * @param[in] s Swr context to check
229 * @see swr_init()
230 * @return positive if it has been initialized, 0 if not initialized
231 */
233
234 /**
235 * Allocate SwrContext if needed and set/reset common parameters.
236 *
237 * This function does not require s to be allocated with swr_alloc(). On the
238 * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
239 * on the allocated context.
240 *
241 * @param s existing Swr context if available, or NULL if not
242 * @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
243 * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
244 * @param out_sample_rate output sample rate (frequency in Hz)
245 * @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
246 * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
247 * @param in_sample_rate input sample rate (frequency in Hz)
248 * @param log_offset logging level offset
249 * @param log_ctx parent logging context, can be NULL
250 *
251 * @see swr_init(), swr_free()
252 * @return NULL on error, allocated context otherwise
253 */
257 int log_offset,
void *
log_ctx);
258
259 /**
260 * @}
261 *
262 * @name SwrContext destructor functions
263 * @{
264 */
265
266 /**
267 * Free the given SwrContext and set the pointer to NULL.
268 *
269 * @param[in] s a pointer to a pointer to Swr context
270 */
272
273 /**
274 * Closes the context so that swr_is_initialized() returns 0.
275 *
276 * The context can be brought back to life by running swr_init(),
277 * swr_init() can also be used without swr_close().
278 * This function is mainly provided for simplifying the usecase
279 * where one tries to support libavresample and libswresample.
280 *
281 * @param[in,out] s Swr context to be closed
282 */
284
285 /**
286 * @}
287 *
288 * @name Core conversion functions
289 * @{
290 */
291
292 /** Convert audio.
293 *
294 * in and in_count can be set to 0 to flush the last few samples out at the
295 * end.
296 *
297 * If more input is provided than output space, then the input will be buffered.
298 * You can avoid this buffering by using swr_get_out_samples() to retrieve an
299 * upper bound on the required number of output samples for the given number of
300 * input samples. Conversion will run directly without copying whenever possible.
301 *
302 * @param s allocated Swr context, with parameters set
303 * @param out output buffers, only the first one need be set in case of packed audio
304 * @param out_count amount of space available for output in samples per channel
305 * @param in input buffers, only the first one need to be set in case of packed audio
306 * @param in_count number of input samples available in one channel
307 *
308 * @return number of samples output per channel, negative value on error
309 */
312
313 /**
314 * Convert the next timestamp from input to output
315 * timestamps are in 1/(in_sample_rate * out_sample_rate) units.
316 *
317 * @note There are 2 slightly differently behaving modes.
318 * @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
319 * in this case timestamps will be passed through with delays compensated
320 * @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
321 * in this case the output timestamps will match output sample numbers.
322 * See ffmpeg-resampler(1) for the two modes of compensation.
323 *
324 * @param s[in] initialized Swr context
325 * @param pts[in] timestamp for the next input sample, INT64_MIN if unknown
326 * @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
327 * function used internally for timestamp compensation.
328 * @return the output timestamp for the next output sample
329 */
331
332 /**
333 * @}
334 *
335 * @name Low-level option setting functions
336 * These functons provide a means to set low-level options that is not possible
337 * with the AVOption API.
338 * @{
339 */
340
341 /**
342 * Activate resampling compensation ("soft" compensation). This function is
343 * internally called when needed in swr_next_pts().
344 *
345 * @param[in,out] s allocated Swr context. If it is not initialized,
346 * or SWR_FLAG_RESAMPLE is not set, swr_init() is
347 * called with the flag set.
348 * @param[in] sample_delta delta in PTS per sample
349 * @param[in] compensation_distance number of samples to compensate for
350 * @return >= 0 on success, AVERROR error codes if:
351 * @li @c s is NULL,
352 * @li @c compensation_distance is less than 0,
353 * @li @c compensation_distance is 0 but sample_delta is not,
354 * @li compensation unsupported by resampler, or
355 * @li swr_init() fails when called.
356 */
358
359 /**
360 * Set a customized input channel mapping.
361 *
362 * @param[in,out] s allocated Swr context, not yet initialized
363 * @param[in] channel_map customized input channel mapping (array of channel
364 * indexes, -1 for a muted channel)
365 * @return >= 0 on success, or AVERROR error code in case of failure.
366 */
368
369 /**
370 * Set a customized remix matrix.
371 *
372 * @param s allocated Swr context, not yet initialized
373 * @param matrix remix coefficients; matrix[i + stride * o] is
374 * the weight of input channel i in output channel o
375 * @param stride offset between lines of the matrix
376 * @return >= 0 on success, or AVERROR error code in case of failure.
377 */
379
380 /**
381 * @}
382 *
383 * @name Sample handling functions
384 * @{
385 */
386
387 /**
388 * Drops the specified number of output samples.
389 *
390 * This function, along with swr_inject_silence(), is called by swr_next_pts()
391 * if needed for "hard" compensation.
392 *
393 * @param s allocated Swr context
394 * @param count number of samples to be dropped
395 *
396 * @return >= 0 on success, or a negative AVERROR code on failure
397 */
399
400 /**
401 * Injects the specified number of silence samples.
402 *
403 * This function, along with swr_drop_output(), is called by swr_next_pts()
404 * if needed for "hard" compensation.
405 *
406 * @param s allocated Swr context
407 * @param count number of samples to be dropped
408 *
409 * @return >= 0 on success, or a negative AVERROR code on failure
410 */
412
413 /**
414 * Gets the delay the next input sample will experience relative to the next output sample.
415 *
416 * Swresample can buffer data if more input has been provided than available
417 * output space, also converting between sample rates needs a delay.
418 * This function returns the sum of all such delays.
419 * The exact delay is not necessarily an integer value in either input or
420 * output sample rate. Especially when downsampling by a large value, the
421 * output sample rate may be a poor choice to represent the delay, similarly
422 * for upsampling and the input sample rate.
423 *
424 * @param s swr context
425 * @param base timebase in which the returned delay will be:
426 * @li if it's set to 1 the returned delay is in seconds
427 * @li if it's set to 1000 the returned delay is in milliseconds
428 * @li if it's set to the input sample rate then the returned
429 * delay is in input samples
430 * @li if it's set to the output sample rate then the returned
431 * delay is in output samples
432 * @li if it's the least common multiple of in_sample_rate and
433 * out_sample_rate then an exact rounding-free delay will be
434 * returned
435 * @returns the delay in 1 / @c base units.
436 */
438
439 /**
440 * Find an upper bound on the number of samples that the next swr_convert
441 * call will output, if called with in_samples of input samples. This
442 * depends on the internal state, and anything changing the internal state
443 * (like further swr_convert() calls) will may change the number of samples
444 * swr_get_out_samples() returns for the same number of input samples.
445 *
446 * @param in_samples number of input samples.
447 * @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
448 * or swr_set_compensation() invalidates this limit
449 * @note it is recommended to pass the correct available buffer size
450 * to all functions like swr_convert() even if swr_get_out_samples()
451 * indicates that less would be used.
452 * @returns an upper bound on the number of samples that the next swr_convert
453 * will output or a negative value to indicate an error
454 */
456
457 /**
458 * @}
459 *
460 * @name Configuration accessors
461 * @{
462 */
463
464 /**
465 * Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
466 *
467 * This is useful to check if the build-time libswresample has the same version
468 * as the run-time one.
469 *
470 * @returns the unsigned int-typed version
471 */
473
474 /**
475 * Return the swr build-time configuration.
476 *
477 * @returns the build-time @c ./configure flags
478 */
480
481 /**
482 * Return the swr license.
483 *
484 * @returns the license of libswresample, determined at build-time
485 */
487
488 /**
489 * @}
490 *
491 * @name AVFrame based API
492 * @{
493 */
494
495 /**
496 * Convert the samples in the input AVFrame and write them to the output AVFrame.
497 *
498 * Input and output AVFrames must have channel_layout, sample_rate and format set.
499 *
500 * If the output AVFrame does not have the data pointers allocated the nb_samples
501 * field will be set using av_frame_get_buffer()
502 * is called to allocate the frame.
503 *
504 * The output AVFrame can be NULL or have fewer allocated samples than required.
505 * In this case, any remaining samples not written to the output will be added
506 * to an internal FIFO buffer, to be returned at the next call to this function
507 * or to swr_convert().
508 *
509 * If converting sample rate, there may be data remaining in the internal
510 * resampling delay buffer. swr_get_delay() tells the number of
511 * remaining samples. To get this data as output, call this function or
512 * swr_convert() with NULL input.
513 *
514 * If the SwrContext configuration does not match the output and
515 * input AVFrame settings the conversion does not take place and depending on
516 * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
517 * or the result of a bitwise-OR of them is returned.
518 *
519 * @see swr_delay()
520 * @see swr_convert()
521 * @see swr_get_delay()
522 *
523 * @param swr audio resample context
524 * @param output output AVFrame
525 * @param input input AVFrame
526 * @return 0 on success, AVERROR on failure or nonmatching
527 * configuration.
528 */
531
532 /**
533 * Configure or reconfigure the SwrContext using the information
534 * provided by the AVFrames.
535 *
536 * The original resampling context is reset even on failure.
537 * The function calls swr_close() internally if the context is open.
538 *
539 * @see swr_close();
540 *
541 * @param swr audio resample context
542 * @param output output AVFrame
543 * @param input input AVFrame
544 * @return 0 on success, AVERROR on failure.
545 */
547
548 /**
549 * @}
550 * @}
551 */
552
553 #endif /* SWRESAMPLE_SWRESAMPLE_H */
void swr_close(struct SwrContext *s)
Closes the context so that swr_is_initialized() returns 0.
int out_sample_rate
output sample rate
This structure describes decoded (raw) audio or video data.
SwrFilterType
Resampling Filter Types.
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...
const int * channel_map
channel index (or -1 if muted channel) map
int swr_get_out_samples(struct SwrContext *s, int in_samples)
Find an upper bound on the number of samples that the next swr_convert call will output, if called with in_samples of input samples.
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance)
Activate resampling compensation ("soft" compensation).
struct SwrContext * swr_alloc(void)
Allocate SwrContext.
SwrDitherType
Dithering algorithms.
void * log_ctx
parent logging context
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count, const uint8_t **in, int in_count)
Convert audio.
enum AVSampleFormat out_sample_fmt
output sample format
SwrEngine
Resampling Engines.
Blackman Nuttall windowed sinc.
The libswresample context.
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.
reference-counted frame API
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride)
Set a customized remix matrix.
int64_t out_ch_layout
output channel layout
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.
int in_sample_rate
input sample rate
const AVClass * swr_get_class(void)
Get the AVClass for SwrContext.
AVSampleFormat
Audio sample formats.
void swr_free(struct SwrContext **s)
Free the given SwrContext and set the pointer to NULL.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map)
Set a customized input channel mapping.
Describe the class of an AVClass context structure.
const char * swresample_license(void)
Return the swr license.
enum AVSampleFormat in_sample_fmt
input sample format
Libswresample version macros.
static int64_t pts
Global timestamp for the audio frames.
int64_t in_ch_layout
input channel layout
int swr_convert_frame(SwrContext *swr, AVFrame *output, const AVFrame *input)
Convert the samples in the input AVFrame and write them to the output AVFrame.
GLint GLenum GLboolean GLsizei stride
int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in)
Configure or reconfigure the SwrContext using the information provided by the AVFrames.
unsigned swresample_version(void)
Return the LIBSWRESAMPLE_VERSION_INT constant.
float matrix[SWR_CH_MAX][SWR_CH_MAX]
floating point rematrixing coefficients
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
const char * swresample_configuration(void)
Return the swr build-time configuration.
int swr_inject_silence(struct SwrContext *s, int count)
Injects the specified number of silence samples.
int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.