avresample: Document avresample_open() a little better
[libav.git] / libavresample / avresample.h
1 /*
2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3 *
4 * This file is part of Libav.
5 *
6 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #ifndef AVRESAMPLE_AVRESAMPLE_H
22 #define AVRESAMPLE_AVRESAMPLE_H
23
24 /**
25 * @file
26 * @ingroup lavr
27 * external API header
28 */
29
30 /**
31 * @defgroup lavr Libavresample
32 * @{
33 *
34 * Libavresample (lavr) is a library that handles audio resampling, sample
35 * format conversion and mixing.
36 *
37 * Interaction with lavr is done through AVAudioResampleContext, which is
38 * allocated with avresample_alloc_context(). It is opaque, so all parameters
39 * must be set with the @ref avoptions API.
40 *
41 * For example the following code will setup conversion from planar float sample
42 * format to interleaved signed 16-bit integer, downsampling from 48kHz to
43 * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
44 * matrix):
45 * @code
46 * AVAudioResampleContext *avr = avresample_alloc_context();
47 * av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
48 * av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
49 * av_opt_set_int(avr, "in_sample_rate", 48000, 0);
50 * av_opt_set_int(avr, "out_sample_rate", 44100, 0);
51 * av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
52 * av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
53 * @endcode
54 *
55 * Once the context is initialized, it must be opened with avresample_open(). If
56 * you need to change the conversion parameters, you must close the context with
57 * avresample_close(), change the parameters as described above, then reopen it
58 * again.
59 *
60 * The conversion itself is done by repeatedly calling avresample_convert().
61 * Note that the samples may get buffered in two places in lavr. The first one
62 * is the output FIFO, where the samples end up if the output buffer is not
63 * large enough. The data stored in there may be retrieved at any time with
64 * avresample_read(). The second place is the resampling delay buffer,
65 * applicable only when resampling is done. The samples in it require more input
66 * before they can be processed. Their current amount is returned by
67 * avresample_get_delay(). At the end of conversion the resampling buffer can be
68 * flushed by calling avresample_convert() with NULL input.
69 *
70 * The following code demonstrates the conversion loop assuming the parameters
71 * from above and caller-defined functions get_input() and handle_output():
72 * @code
73 * uint8_t **input;
74 * int in_linesize, in_samples;
75 *
76 * while (get_input(&input, &in_linesize, &in_samples)) {
77 * uint8_t *output
78 * int out_linesize;
79 * int out_samples = avresample_get_out_samples(avr, in_samples);
80 *
81 * av_samples_alloc(&output, &out_linesize, 2, out_samples,
82 * AV_SAMPLE_FMT_S16, 0);
83 * out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
84 * input, in_linesize, in_samples);
85 * handle_output(output, out_linesize, out_samples);
86 * av_freep(&output);
87 * }
88 * @endcode
89 *
90 * When the conversion is finished and the FIFOs are flushed if required, the
91 * conversion context and everything associated with it must be freed with
92 * avresample_free().
93 */
94
95 #include "libavutil/avutil.h"
96 #include "libavutil/channel_layout.h"
97 #include "libavutil/dict.h"
98 #include "libavutil/frame.h"
99 #include "libavutil/log.h"
100 #include "libavutil/mathematics.h"
101
102 #include "libavresample/version.h"
103
104 #define AVRESAMPLE_MAX_CHANNELS 32
105
106 typedef struct AVAudioResampleContext AVAudioResampleContext;
107
108 /** Mixing Coefficient Types */
109 enum AVMixCoeffType {
110 AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
111 AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
112 AV_MIX_COEFF_TYPE_FLT, /** floating-point */
113 AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
114 };
115
116 /** Resampling Filter Types */
117 enum AVResampleFilterType {
118 AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
119 AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
120 AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
121 };
122
123 enum AVResampleDitherMethod {
124 AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
125 AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
126 AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
127 AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
128 AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
129 AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
130 };
131
132 /**
133 * Return the LIBAVRESAMPLE_VERSION_INT constant.
134 */
135 unsigned avresample_version(void);
136
137 /**
138 * Return the libavresample build-time configuration.
139 * @return configure string
140 */
141 const char *avresample_configuration(void);
142
143 /**
144 * Return the libavresample license.
145 */
146 const char *avresample_license(void);
147
148 /**
149 * Get the AVClass for AVAudioResampleContext.
150 *
151 * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
152 * without allocating a context.
153 *
154 * @see av_opt_find().
155 *
156 * @return AVClass for AVAudioResampleContext
157 */
158 const AVClass *avresample_get_class(void);
159
160 /**
161 * Allocate AVAudioResampleContext and set options.
162 *
163 * @return allocated audio resample context, or NULL on failure
164 */
165 AVAudioResampleContext *avresample_alloc_context(void);
166
167 /**
168 * Initialize AVAudioResampleContext.
169 * @note The context must be configured using the AVOption API.
170 * @note The fields "in_channel_layout", "out_channel_layout",
171 * "in_sample_rate", "out_sample_rate", "in_sample_fmt",
172 * "out_sample_fmt" must be set.
173 *
174 * @see av_opt_set_int()
175 * @see av_opt_set_dict()
176 * @see av_get_default_channel_layout()
177 *
178 * @param avr audio resample context
179 * @return 0 on success, negative AVERROR code on failure
180 */
181 int avresample_open(AVAudioResampleContext *avr);
182
183 /**
184 * Check whether an AVAudioResampleContext is open or closed.
185 *
186 * @param avr AVAudioResampleContext to check
187 * @return 1 if avr is open, 0 if avr is closed.
188 */
189 int avresample_is_open(AVAudioResampleContext *avr);
190
191 /**
192 * Close AVAudioResampleContext.
193 *
194 * This closes the context, but it does not change the parameters. The context
195 * can be reopened with avresample_open(). It does, however, clear the output
196 * FIFO and any remaining leftover samples in the resampling delay buffer. If
197 * there was a custom matrix being used, that is also cleared.
198 *
199 * @see avresample_convert()
200 * @see avresample_set_matrix()
201 *
202 * @param avr audio resample context
203 */
204 void avresample_close(AVAudioResampleContext *avr);
205
206 /**
207 * Free AVAudioResampleContext and associated AVOption values.
208 *
209 * This also calls avresample_close() before freeing.
210 *
211 * @param avr audio resample context
212 */
213 void avresample_free(AVAudioResampleContext **avr);
214
215 /**
216 * Generate a channel mixing matrix.
217 *
218 * This function is the one used internally by libavresample for building the
219 * default mixing matrix. It is made public just as a utility function for
220 * building custom matrices.
221 *
222 * @param in_layout input channel layout
223 * @param out_layout output channel layout
224 * @param center_mix_level mix level for the center channel
225 * @param surround_mix_level mix level for the surround channel(s)
226 * @param lfe_mix_level mix level for the low-frequency effects channel
227 * @param normalize if 1, coefficients will be normalized to prevent
228 * overflow. if 0, coefficients will not be
229 * normalized.
230 * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
231 * the weight of input channel i in output channel o.
232 * @param stride distance between adjacent input channels in the
233 * matrix array
234 * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
235 * @return 0 on success, negative AVERROR code on failure
236 */
237 int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
238 double center_mix_level, double surround_mix_level,
239 double lfe_mix_level, int normalize, double *matrix,
240 int stride, enum AVMatrixEncoding matrix_encoding);
241
242 /**
243 * Get the current channel mixing matrix.
244 *
245 * If no custom matrix has been previously set or the AVAudioResampleContext is
246 * not open, an error is returned.
247 *
248 * @param avr audio resample context
249 * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
250 * input channel i in output channel o.
251 * @param stride distance between adjacent input channels in the matrix array
252 * @return 0 on success, negative AVERROR code on failure
253 */
254 int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
255 int stride);
256
257 /**
258 * Set channel mixing matrix.
259 *
260 * Allows for setting a custom mixing matrix, overriding the default matrix
261 * generated internally during avresample_open(). This function can be called
262 * anytime on an allocated context, either before or after calling
263 * avresample_open(), as long as the channel layouts have been set.
264 * avresample_convert() always uses the current matrix.
265 * Calling avresample_close() on the context will clear the current matrix.
266 *
267 * @see avresample_close()
268 *
269 * @param avr audio resample context
270 * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
271 * input channel i in output channel o.
272 * @param stride distance between adjacent input channels in the matrix array
273 * @return 0 on success, negative AVERROR code on failure
274 */
275 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
276 int stride);
277
278 /**
279 * Set a customized input channel mapping.
280 *
281 * This function can only be called when the allocated context is not open.
282 * Also, the input channel layout must have already been set.
283 *
284 * Calling avresample_close() on the context will clear the channel mapping.
285 *
286 * The map for each input channel specifies the channel index in the source to
287 * use for that particular channel, or -1 to mute the channel. Source channels
288 * can be duplicated by using the same index for multiple input channels.
289 *
290 * Examples:
291 *
292 * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to Libav order (L,R,C,LFE,Ls,Rs):
293 * { 1, 2, 0, 5, 3, 4 }
294 *
295 * Muting the 3rd channel in 4-channel input:
296 * { 0, 1, -1, 3 }
297 *
298 * Duplicating the left channel of stereo input:
299 * { 0, 0 }
300 *
301 * @param avr audio resample context
302 * @param channel_map customized input channel mapping
303 * @return 0 on success, negative AVERROR code on failure
304 */
305 int avresample_set_channel_mapping(AVAudioResampleContext *avr,
306 const int *channel_map);
307
308 /**
309 * Set compensation for resampling.
310 *
311 * This can be called anytime after avresample_open(). If resampling is not
312 * automatically enabled because of a sample rate conversion, the
313 * "force_resampling" option must have been set to 1 when opening the context
314 * in order to use resampling compensation.
315 *
316 * @param avr audio resample context
317 * @param sample_delta compensation delta, in samples
318 * @param compensation_distance compensation distance, in samples
319 * @return 0 on success, negative AVERROR code on failure
320 */
321 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
322 int compensation_distance);
323
324 /**
325 * Provide the upper bound on the number of samples the configured
326 * conversion would output.
327 *
328 * @param avr audio resample context
329 * @param in_nb_samples number of input samples
330 *
331 * @return number of samples or AVERROR(EINVAL) if the value
332 * would exceed INT_MAX
333 */
334
335 int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
336
337 /**
338 * Convert input samples and write them to the output FIFO.
339 *
340 * The upper bound on the number of output samples can be obtained through
341 * avresample_get_out_samples().
342 *
343 * The output data can be NULL or have fewer allocated samples than required.
344 * In this case, any remaining samples not written to the output will be added
345 * to an internal FIFO buffer, to be returned at the next call to this function
346 * or to avresample_read().
347 *
348 * If converting sample rate, there may be data remaining in the internal
349 * resampling delay buffer. avresample_get_delay() tells the number of remaining
350 * samples. To get this data as output, call avresample_convert() with NULL
351 * input.
352 *
353 * At the end of the conversion process, there may be data remaining in the
354 * internal FIFO buffer. avresample_available() tells the number of remaining
355 * samples. To get this data as output, either call avresample_convert() with
356 * NULL input or call avresample_read().
357 *
358 * @see avresample_get_out_samples()
359 * @see avresample_read()
360 * @see avresample_get_delay()
361 *
362 * @param avr audio resample context
363 * @param output output data pointers
364 * @param out_plane_size output plane size, in bytes.
365 * This can be 0 if unknown, but that will lead to
366 * optimized functions not being used directly on the
367 * output, which could slow down some conversions.
368 * @param out_samples maximum number of samples that the output buffer can hold
369 * @param input input data pointers
370 * @param in_plane_size input plane size, in bytes
371 * This can be 0 if unknown, but that will lead to
372 * optimized functions not being used directly on the
373 * input, which could slow down some conversions.
374 * @param in_samples number of input samples to convert
375 * @return number of samples written to the output buffer,
376 * not including converted samples added to the internal
377 * output FIFO
378 */
379 int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
380 int out_plane_size, int out_samples, uint8_t **input,
381 int in_plane_size, int in_samples);
382
383 /**
384 * Return the number of samples currently in the resampling delay buffer.
385 *
386 * When resampling, there may be a delay between the input and output. Any
387 * unconverted samples in each call are stored internally in a delay buffer.
388 * This function allows the user to determine the current number of samples in
389 * the delay buffer, which can be useful for synchronization.
390 *
391 * @see avresample_convert()
392 *
393 * @param avr audio resample context
394 * @return number of samples currently in the resampling delay buffer
395 */
396 int avresample_get_delay(AVAudioResampleContext *avr);
397
398 /**
399 * Return the number of available samples in the output FIFO.
400 *
401 * During conversion, if the user does not specify an output buffer or
402 * specifies an output buffer that is smaller than what is needed, remaining
403 * samples that are not written to the output are stored to an internal FIFO
404 * buffer. The samples in the FIFO can be read with avresample_read() or
405 * avresample_convert().
406 *
407 * @see avresample_read()
408 * @see avresample_convert()
409 *
410 * @param avr audio resample context
411 * @return number of samples available for reading
412 */
413 int avresample_available(AVAudioResampleContext *avr);
414
415 /**
416 * Read samples from the output FIFO.
417 *
418 * During conversion, if the user does not specify an output buffer or
419 * specifies an output buffer that is smaller than what is needed, remaining
420 * samples that are not written to the output are stored to an internal FIFO
421 * buffer. This function can be used to read samples from that internal FIFO.
422 *
423 * @see avresample_available()
424 * @see avresample_convert()
425 *
426 * @param avr audio resample context
427 * @param output output data pointers. May be NULL, in which case
428 * nb_samples of data is discarded from output FIFO.
429 * @param nb_samples number of samples to read from the FIFO
430 * @return the number of samples written to output
431 */
432 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
433
434 /**
435 * Convert the samples in the input AVFrame and write them to the output AVFrame.
436 *
437 * Input and output AVFrames must have channel_layout, sample_rate and format set.
438 *
439 * The upper bound on the number of output samples is obtained through
440 * avresample_get_out_samples().
441 *
442 * If the output AVFrame does not have the data pointers allocated the nb_samples
443 * field will be set using avresample_get_out_samples() and av_frame_get_buffer()
444 * is called to allocate the frame.
445 *
446 * The output AVFrame can be NULL or have fewer allocated samples than required.
447 * In this case, any remaining samples not written to the output will be added
448 * to an internal FIFO buffer, to be returned at the next call to this function
449 * or to avresample_convert() or to avresample_read().
450 *
451 * If converting sample rate, there may be data remaining in the internal
452 * resampling delay buffer. avresample_get_delay() tells the number of
453 * remaining samples. To get this data as output, call this function or
454 * avresample_convert() with NULL input.
455 *
456 * At the end of the conversion process, there may be data remaining in the
457 * internal FIFO buffer. avresample_available() tells the number of remaining
458 * samples. To get this data as output, either call this function or
459 * avresample_convert() with NULL input or call avresample_read().
460 *
461 * If the AVAudioResampleContext configuration does not match the output and
462 * input AVFrame settings the conversion does not take place and depending on
463 * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
464 * or AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned.
465 *
466 * @see avresample_get_out_samples()
467 * @see avresample_available()
468 * @see avresample_convert()
469 * @see avresample_read()
470 * @see avresample_get_delay()
471 *
472 * @param avr audio resample context
473 * @param output output AVFrame
474 * @param input input AVFrame
475 * @return 0 on success, AVERROR on failure or nonmatching
476 * configuration.
477 */
478 int avresample_convert_frame(AVAudioResampleContext *avr,
479 AVFrame *output, AVFrame *input);
480
481 /**
482 * Configure or reconfigure the AVAudioResampleContext using the information
483 * provided by the AVFrames.
484 *
485 * The original resampling context is reset even on failure.
486 * The function calls avresample_close() internally if the context is open.
487 *
488 * @see avresample_open();
489 * @see avresample_close();
490 *
491 * @param avr audio resample context
492 * @param output output AVFrame
493 * @param input input AVFrame
494 * @return 0 on success, AVERROR on failure.
495 */
496 int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in);
497
498 /**
499 * @}
500 */
501
502 #endif /* AVRESAMPLE_AVRESAMPLE_H */