bitstream: Avoid undefined behavior in bitstream_skip()
[libav.git] / libavresample / avresample.h
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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
01b76019 26 * @ingroup lavr
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27 * external API header
28 */
29
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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);
be6cde3c 52 * av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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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;
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79 * int out_samples = avresample_get_out_samples(avr, in_samples);
80 *
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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
c8af852b 95#include "libavutil/avutil.h"
a903f8f0 96#include "libavutil/channel_layout.h"
c8af852b 97#include "libavutil/dict.h"
fb1ddcdc 98#include "libavutil/frame.h"
c8af852b 99#include "libavutil/log.h"
b2d45654 100#include "libavutil/mathematics.h"
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101
102#include "libavresample/version.h"
103
104#define AVRESAMPLE_MAX_CHANNELS 32
105
106typedef struct AVAudioResampleContext AVAudioResampleContext;
107
108/** Mixing Coefficient Types */
109enum AVMixCoeffType {
4010d724 110 AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
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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
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116/** Resampling Filter Types */
117enum 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
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123enum 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
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132/**
133 * Return the LIBAVRESAMPLE_VERSION_INT constant.
134 */
135unsigned avresample_version(void);
136
137/**
138 * Return the libavresample build-time configuration.
139 * @return configure string
140 */
141const char *avresample_configuration(void);
142
143/**
144 * Return the libavresample license.
145 */
146const 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 */
158const 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 */
165AVAudioResampleContext *avresample_alloc_context(void);
166
167/**
168 * Initialize AVAudioResampleContext.
fb1ddcdc 169 * @note The context must be configured using the AVOption API.
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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.
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173 *
174 * @see av_opt_set_int()
175 * @see av_opt_set_dict()
e2854e73 176 * @see av_get_default_channel_layout()
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177 *
178 * @param avr audio resample context
179 * @return 0 on success, negative AVERROR code on failure
180 */
181int avresample_open(AVAudioResampleContext *avr);
182
183/**
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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 */
189int avresample_is_open(AVAudioResampleContext *avr);
190
191/**
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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 */
204void 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 */
213void 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
3971be0e 234 * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
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235 * @return 0 on success, negative AVERROR code on failure
236 */
237int 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,
3971be0e 240 int stride, enum AVMatrixEncoding matrix_encoding);
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241
242/**
243 * Get the current channel mixing matrix.
244 *
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245 * If no custom matrix has been previously set or the AVAudioResampleContext is
246 * not open, an error is returned.
247 *
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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 */
254int 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
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263 * avresample_open(), as long as the channel layouts have been set.
264 * avresample_convert() always uses the current matrix.
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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 */
275int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
276 int stride);
277
278/**
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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 */
305int avresample_set_channel_mapping(AVAudioResampleContext *avr,
306 const int *channel_map);
307
308/**
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309 * Set compensation for resampling.
310 *
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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.
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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 */
321int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
322 int compensation_distance);
323
324/**
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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
335int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
336
337/**
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338 * Convert input samples and write them to the output FIFO.
339 *
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340 * The upper bound on the number of output samples can be obtained through
341 * avresample_get_out_samples().
dc579306 342 *
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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 *
b2d45654 358 * @see avresample_get_out_samples()
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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 */
e7ba5b1d 379int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
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380 int out_plane_size, int out_samples,
381 uint8_t * const *input, int in_plane_size,
382 int in_samples);
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383
384/**
385 * Return the number of samples currently in the resampling delay buffer.
386 *
387 * When resampling, there may be a delay between the input and output. Any
388 * unconverted samples in each call are stored internally in a delay buffer.
389 * This function allows the user to determine the current number of samples in
390 * the delay buffer, which can be useful for synchronization.
391 *
392 * @see avresample_convert()
393 *
394 * @param avr audio resample context
395 * @return number of samples currently in the resampling delay buffer
396 */
397int avresample_get_delay(AVAudioResampleContext *avr);
398
399/**
400 * Return the number of available samples in the output FIFO.
401 *
402 * During conversion, if the user does not specify an output buffer or
403 * specifies an output buffer that is smaller than what is needed, remaining
404 * samples that are not written to the output are stored to an internal FIFO
405 * buffer. The samples in the FIFO can be read with avresample_read() or
406 * avresample_convert().
407 *
408 * @see avresample_read()
409 * @see avresample_convert()
410 *
411 * @param avr audio resample context
412 * @return number of samples available for reading
413 */
414int avresample_available(AVAudioResampleContext *avr);
415
416/**
417 * Read samples from the output FIFO.
418 *
419 * During conversion, if the user does not specify an output buffer or
420 * specifies an output buffer that is smaller than what is needed, remaining
421 * samples that are not written to the output are stored to an internal FIFO
422 * buffer. This function can be used to read samples from that internal FIFO.
423 *
424 * @see avresample_available()
425 * @see avresample_convert()
426 *
427 * @param avr audio resample context
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428 * @param output output data pointers. May be NULL, in which case
429 * nb_samples of data is discarded from output FIFO.
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430 * @param nb_samples number of samples to read from the FIFO
431 * @return the number of samples written to output
432 */
e7ba5b1d 433int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
c8af852b 434
01b76019 435/**
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436 * Convert the samples in the input AVFrame and write them to the output AVFrame.
437 *
438 * Input and output AVFrames must have channel_layout, sample_rate and format set.
439 *
440 * The upper bound on the number of output samples is obtained through
441 * avresample_get_out_samples().
442 *
443 * If the output AVFrame does not have the data pointers allocated the nb_samples
444 * field will be set using avresample_get_out_samples() and av_frame_get_buffer()
445 * is called to allocate the frame.
446 *
447 * The output AVFrame can be NULL or have fewer allocated samples than required.
448 * In this case, any remaining samples not written to the output will be added
449 * to an internal FIFO buffer, to be returned at the next call to this function
450 * or to avresample_convert() or to avresample_read().
451 *
452 * If converting sample rate, there may be data remaining in the internal
453 * resampling delay buffer. avresample_get_delay() tells the number of
454 * remaining samples. To get this data as output, call this function or
455 * avresample_convert() with NULL input.
456 *
457 * At the end of the conversion process, there may be data remaining in the
458 * internal FIFO buffer. avresample_available() tells the number of remaining
459 * samples. To get this data as output, either call this function or
460 * avresample_convert() with NULL input or call avresample_read().
461 *
462 * If the AVAudioResampleContext configuration does not match the output and
463 * input AVFrame settings the conversion does not take place and depending on
464 * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
465 * or AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned.
466 *
467 * @see avresample_get_out_samples()
468 * @see avresample_available()
469 * @see avresample_convert()
470 * @see avresample_read()
471 * @see avresample_get_delay()
472 *
473 * @param avr audio resample context
474 * @param output output AVFrame
475 * @param input input AVFrame
476 * @return 0 on success, AVERROR on failure or nonmatching
477 * configuration.
478 */
479int avresample_convert_frame(AVAudioResampleContext *avr,
480 AVFrame *output, AVFrame *input);
481
482/**
483 * Configure or reconfigure the AVAudioResampleContext using the information
484 * provided by the AVFrames.
485 *
486 * The original resampling context is reset even on failure.
487 * The function calls avresample_close() internally if the context is open.
488 *
489 * @see avresample_open();
490 * @see avresample_close();
491 *
492 * @param avr audio resample context
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493 * @param out output AVFrame
494 * @param in input AVFrame
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495 * @return 0 on success, AVERROR on failure.
496 */
497int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in);
498
499/**
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500 * @}
501 */
502
c8af852b 503#endif /* AVRESAMPLE_AVRESAMPLE_H */