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avresample.h
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1 /*
2  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #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"
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 
107 
108 /** Mixing Coefficient Types */
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 */
119  AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
120  AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
121 };
122 
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  */
166 
167 /**
168  * Initialize AVAudioResampleContext.
169  * @note The context must be configured using the AVOption API.
170  *
171  * @see av_opt_set_int()
172  * @see av_opt_set_dict()
173  *
174  * @param avr audio resample context
175  * @return 0 on success, negative AVERROR code on failure
176  */
178 
179 /**
180  * Check whether an AVAudioResampleContext is open or closed.
181  *
182  * @param avr AVAudioResampleContext to check
183  * @return 1 if avr is open, 0 if avr is closed.
184  */
186 
187 /**
188  * Close AVAudioResampleContext.
189  *
190  * This closes the context, but it does not change the parameters. The context
191  * can be reopened with avresample_open(). It does, however, clear the output
192  * FIFO and any remaining leftover samples in the resampling delay buffer. If
193  * there was a custom matrix being used, that is also cleared.
194  *
195  * @see avresample_convert()
196  * @see avresample_set_matrix()
197  *
198  * @param avr audio resample context
199  */
201 
202 /**
203  * Free AVAudioResampleContext and associated AVOption values.
204  *
205  * This also calls avresample_close() before freeing.
206  *
207  * @param avr audio resample context
208  */
210 
211 /**
212  * Generate a channel mixing matrix.
213  *
214  * This function is the one used internally by libavresample for building the
215  * default mixing matrix. It is made public just as a utility function for
216  * building custom matrices.
217  *
218  * @param in_layout input channel layout
219  * @param out_layout output channel layout
220  * @param center_mix_level mix level for the center channel
221  * @param surround_mix_level mix level for the surround channel(s)
222  * @param lfe_mix_level mix level for the low-frequency effects channel
223  * @param normalize if 1, coefficients will be normalized to prevent
224  * overflow. if 0, coefficients will not be
225  * normalized.
226  * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
227  * the weight of input channel i in output channel o.
228  * @param stride distance between adjacent input channels in the
229  * matrix array
230  * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
231  * @return 0 on success, negative AVERROR code on failure
232  */
233 int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
234  double center_mix_level, double surround_mix_level,
235  double lfe_mix_level, int normalize, double *matrix,
237 
238 /**
239  * Get the current channel mixing matrix.
240  *
241  * If no custom matrix has been previously set or the AVAudioResampleContext is
242  * not open, an error is returned.
243  *
244  * @param avr audio resample context
245  * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
246  * input channel i in output channel o.
247  * @param stride distance between adjacent input channels in the matrix array
248  * @return 0 on success, negative AVERROR code on failure
249  */
250 int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
251  int stride);
252 
253 /**
254  * Set channel mixing matrix.
255  *
256  * Allows for setting a custom mixing matrix, overriding the default matrix
257  * generated internally during avresample_open(). This function can be called
258  * anytime on an allocated context, either before or after calling
259  * avresample_open(), as long as the channel layouts have been set.
260  * avresample_convert() always uses the current matrix.
261  * Calling avresample_close() on the context will clear the current matrix.
262  *
263  * @see avresample_close()
264  *
265  * @param avr audio resample context
266  * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
267  * input channel i in output channel o.
268  * @param stride distance between adjacent input channels in the matrix array
269  * @return 0 on success, negative AVERROR code on failure
270  */
271 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
272  int stride);
273 
274 /**
275  * Set a customized input channel mapping.
276  *
277  * This function can only be called when the allocated context is not open.
278  * Also, the input channel layout must have already been set.
279  *
280  * Calling avresample_close() on the context will clear the channel mapping.
281  *
282  * The map for each input channel specifies the channel index in the source to
283  * use for that particular channel, or -1 to mute the channel. Source channels
284  * can be duplicated by using the same index for multiple input channels.
285  *
286  * Examples:
287  *
288  * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
289  * { 1, 2, 0, 5, 3, 4 }
290  *
291  * Muting the 3rd channel in 4-channel input:
292  * { 0, 1, -1, 3 }
293  *
294  * Duplicating the left channel of stereo input:
295  * { 0, 0 }
296  *
297  * @param avr audio resample context
298  * @param channel_map customized input channel mapping
299  * @return 0 on success, negative AVERROR code on failure
300  */
302  const int *channel_map);
303 
304 /**
305  * Set compensation for resampling.
306  *
307  * This can be called anytime after avresample_open(). If resampling is not
308  * automatically enabled because of a sample rate conversion, the
309  * "force_resampling" option must have been set to 1 when opening the context
310  * in order to use resampling compensation.
311  *
312  * @param avr audio resample context
313  * @param sample_delta compensation delta, in samples
314  * @param compensation_distance compensation distance, in samples
315  * @return 0 on success, negative AVERROR code on failure
316  */
317 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
318  int compensation_distance);
319 
320 /**
321  * Provide the upper bound on the number of samples the configured
322  * conversion would output.
323  *
324  * @param avr audio resample context
325  * @param in_nb_samples number of input samples
326  *
327  * @return number of samples or AVERROR(EINVAL) if the value
328  * would exceed INT_MAX
329  */
330 
331 int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
332 
333 /**
334  * Convert input samples and write them to the output FIFO.
335  *
336  * The upper bound on the number of output samples can be obtained through
337  * avresample_get_out_samples().
338  *
339  * The output data can be NULL or have fewer allocated samples than required.
340  * In this case, any remaining samples not written to the output will be added
341  * to an internal FIFO buffer, to be returned at the next call to this function
342  * or to avresample_read().
343  *
344  * If converting sample rate, there may be data remaining in the internal
345  * resampling delay buffer. avresample_get_delay() tells the number of remaining
346  * samples. To get this data as output, call avresample_convert() with NULL
347  * input.
348  *
349  * At the end of the conversion process, there may be data remaining in the
350  * internal FIFO buffer. avresample_available() tells the number of remaining
351  * samples. To get this data as output, either call avresample_convert() with
352  * NULL input or call avresample_read().
353  *
354  * @see avresample_get_out_samples()
355  * @see avresample_read()
356  * @see avresample_get_delay()
357  *
358  * @param avr audio resample context
359  * @param output output data pointers
360  * @param out_plane_size output plane size, in bytes.
361  * This can be 0 if unknown, but that will lead to
362  * optimized functions not being used directly on the
363  * output, which could slow down some conversions.
364  * @param out_samples maximum number of samples that the output buffer can hold
365  * @param input input data pointers
366  * @param in_plane_size input plane size, in bytes
367  * This can be 0 if unknown, but that will lead to
368  * optimized functions not being used directly on the
369  * input, which could slow down some conversions.
370  * @param in_samples number of input samples to convert
371  * @return number of samples written to the output buffer,
372  * not including converted samples added to the internal
373  * output FIFO
374  */
376  int out_plane_size, int out_samples, uint8_t **input,
377  int in_plane_size, int in_samples);
378 
379 /**
380  * Return the number of samples currently in the resampling delay buffer.
381  *
382  * When resampling, there may be a delay between the input and output. Any
383  * unconverted samples in each call are stored internally in a delay buffer.
384  * This function allows the user to determine the current number of samples in
385  * the delay buffer, which can be useful for synchronization.
386  *
387  * @see avresample_convert()
388  *
389  * @param avr audio resample context
390  * @return number of samples currently in the resampling delay buffer
391  */
393 
394 /**
395  * Return the number of available samples in the output FIFO.
396  *
397  * During conversion, if the user does not specify an output buffer or
398  * specifies an output buffer that is smaller than what is needed, remaining
399  * samples that are not written to the output are stored to an internal FIFO
400  * buffer. The samples in the FIFO can be read with avresample_read() or
401  * avresample_convert().
402  *
403  * @see avresample_read()
404  * @see avresample_convert()
405  *
406  * @param avr audio resample context
407  * @return number of samples available for reading
408  */
410 
411 /**
412  * Read samples from the output FIFO.
413  *
414  * During conversion, if the user does not specify an output buffer or
415  * specifies an output buffer that is smaller than what is needed, remaining
416  * samples that are not written to the output are stored to an internal FIFO
417  * buffer. This function can be used to read samples from that internal FIFO.
418  *
419  * @see avresample_available()
420  * @see avresample_convert()
421  *
422  * @param avr audio resample context
423  * @param output output data pointers. May be NULL, in which case
424  * nb_samples of data is discarded from output FIFO.
425  * @param nb_samples number of samples to read from the FIFO
426  * @return the number of samples written to output
427  */
428 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
429 
430 /**
431  * Convert the samples in the input AVFrame and write them to the output AVFrame.
432  *
433  * Input and output AVFrames must have channel_layout, sample_rate and format set.
434  *
435  * The upper bound on the number of output samples is obtained through
436  * avresample_get_out_samples().
437  *
438  * If the output AVFrame does not have the data pointers allocated the nb_samples
439  * field will be set using avresample_get_out_samples() and av_frame_get_buffer()
440  * is called to allocate the frame.
441  *
442  * The output AVFrame can be NULL or have fewer allocated samples than required.
443  * In this case, any remaining samples not written to the output will be added
444  * to an internal FIFO buffer, to be returned at the next call to this function
445  * or to avresample_convert() or to avresample_read().
446  *
447  * If converting sample rate, there may be data remaining in the internal
448  * resampling delay buffer. avresample_get_delay() tells the number of
449  * remaining samples. To get this data as output, call this function or
450  * avresample_convert() with NULL input.
451  *
452  * At the end of the conversion process, there may be data remaining in the
453  * internal FIFO buffer. avresample_available() tells the number of remaining
454  * samples. To get this data as output, either call this function or
455  * avresample_convert() with NULL input or call avresample_read().
456  *
457  * If the AVAudioResampleContext configuration does not match the output and
458  * input AVFrame settings the conversion does not take place and depending on
459  * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
460  * or AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned.
461  *
462  * @see avresample_get_out_samples()
463  * @see avresample_available()
464  * @see avresample_convert()
465  * @see avresample_read()
466  * @see avresample_get_delay()
467  *
468  * @param avr audio resample context
469  * @param output output AVFrame
470  * @param input input AVFrame
471  * @return 0 on success, AVERROR on failure or nonmatching
472  * configuration.
473  */
475  AVFrame *output, AVFrame *input);
476 
477 /**
478  * Configure or reconfigure the AVAudioResampleContext using the information
479  * provided by the AVFrames.
480  *
481  * The original resampling context is reset even on failure.
482  * The function calls avresample_close() internally if the context is open.
483  *
484  * @see avresample_open();
485  * @see avresample_close();
486  *
487  * @param avr audio resample context
488  * @param output output AVFrame
489  * @param input input AVFrame
490  * @return 0 on success, AVERROR on failure.
491  */
493 
494 /**
495  * @}
496  */
497 
498 #endif /* AVRESAMPLE_AVRESAMPLE_H */