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swresample.h
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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>
124 #include "libavutil/frame.h"
125 #include "libavutil/samplefmt.h"
126 
127 #include "libswresample/version.h"
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 
151  SWR_DITHER_NS = 64, ///< not part of API/ABI
159  SWR_DITHER_NB, ///< not part of API/ABI
160 };
161 
162 /** Resampling Engines */
163 enum SwrEngine {
164  SWR_ENGINE_SWR, /**< SW Resampler */
165  SWR_ENGINE_SOXR, /**< SoX Resampler */
166  SWR_ENGINE_NB, ///< not part of API/ABI
167 };
168 
169 /** Resampling Filter Types */
171  SWR_FILTER_TYPE_CUBIC, /**< Cubic */
172  SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
173  SWR_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
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  */
186 typedef struct SwrContext SwrContext;
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  */
195 const AVClass *swr_get_class(void);
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  */
211 struct SwrContext *swr_alloc(void);
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  */
223 int swr_init(struct SwrContext *s);
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  */
232 int swr_is_initialized(struct SwrContext *s);
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  */
254 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
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  */
271 void swr_free(struct SwrContext **s);
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  */
283 void swr_close(struct SwrContext *s);
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  */
310 int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
311  const uint8_t **in , int in_count);
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  */
330 int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
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  */
357 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
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  */
367 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
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  */
378 int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
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  */
398 int swr_drop_output(struct SwrContext *s, int count);
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  */
411 int swr_inject_silence(struct SwrContext *s, int count);
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  */
437 int64_t swr_get_delay(struct SwrContext *s, int64_t base);
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  */
455 int swr_get_out_samples(struct SwrContext *s, int in_samples);
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  */
472 unsigned swresample_version(void);
473 
474 /**
475  * Return the swr build-time configuration.
476  *
477  * @returns the build-time @c ./configure flags
478  */
479 const char *swresample_configuration(void);
480 
481 /**
482  * Return the swr license.
483  *
484  * @returns the license of libswresample, determined at build-time
485  */
486 const char *swresample_license(void);
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  */
530  AVFrame *output, const AVFrame *input);
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  */
546 int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in);
547 
548 /**
549  * @}
550  * @}
551  */
552 
553 #endif /* SWRESAMPLE_SWRESAMPLE_H */
const char * s
Definition: avisynth_c.h:631
void swr_close(struct SwrContext *s)
Closes the context so that swr_is_initialized() returns 0.
Definition: swresample.c:151
int out_sample_rate
output sample rate
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
SoX Resampler.
Definition: swresample.h:165
SwrFilterType
Resampling Filter Types.
Definition: swresample.h:170
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:898
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.
Definition: swresample.c:856
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance)
Activate resampling compensation ("soft" compensation).
Definition: swresample.c:878
uint8_t
struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:148
SwrDitherType
Dithering algorithms.
Definition: swresample.h:145
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.
Kaiser Windowed Sinc.
Definition: swresample.h:173
enum AVSampleFormat out_sample_fmt
output sample format
SwrEngine
Resampling Engines.
Definition: swresample.h:163
Blackman Nuttall Windowed Sinc.
Definition: swresample.h:172
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.
Definition: swresample.c:848
GLsizei count
Definition: opengl_enc.c:109
not part of API/ABI
Definition: swresample.h:166
reference-counted frame API
not part of API/ABI
Definition: swresample.h:159
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
Definition: swresample.c:809
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride)
Set a customized remix matrix.
Definition: rematrix.c:61
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.
Definition: swresample.c:59
not part of API/ABI
Definition: swresample.h:151
int in_sample_rate
input sample rate
const AVClass * swr_get_class(void)
Get the AVClass for SwrContext.
Definition: options.c:143
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
void swr_free(struct SwrContext **s)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:140
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.
Definition: swresample.c:52
Describe the class of an AVClass context structure.
Definition: log.h:67
const char * swresample_license(void)
Return the swr license.
Definition: swresample.c:46
enum AVSampleFormat in_sample_fmt
input sample format
Libswresample version macros.
static int64_t pts
Global timestamp for the audio frames.
SW Resampler.
Definition: swresample.h:164
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
Definition: opengl_enc.c:105
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.
Definition: swresample.c:35
float matrix[SWR_CH_MAX][SWR_CH_MAX]
floating point rematrixing coefficients
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-> out
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
Definition: swresample.c:691
const char * swresample_configuration(void)
Return the swr build-time configuration.
Definition: swresample.c:41
int swr_inject_silence(struct SwrContext *s, int count)
Injects the specified number of silence samples.
Definition: swresample.c:820
int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:155