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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 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_available(avr) +
80  * av_rescale_rnd(avresample_get_delay(avr) +
81  * in_samples, 44100, 48000, AV_ROUND_UP);
82  * av_samples_alloc(&output, &out_linesize, 2, out_samples,
83  * AV_SAMPLE_FMT_S16, 0);
84  * out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
85  * input, in_linesize, in_samples);
86  * handle_output(output, out_linesize, out_samples);
87  * av_freep(&output);
88  * }
89  * @endcode
90  *
91  * When the conversion is finished and the FIFOs are flushed if required, the
92  * conversion context and everything associated with it must be freed with
93  * avresample_free().
94  */
95 
96 #include "libavutil/avutil.h"
98 #include "libavutil/dict.h"
99 #include "libavutil/log.h"
100 
101 #include "libavresample/version.h"
102 
103 #define AVRESAMPLE_MAX_CHANNELS 32
104 
106 
107 /** Mixing Coefficient Types */
109  AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
110  AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
111  AV_MIX_COEFF_TYPE_FLT, /** floating-point */
112  AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
113 };
114 
115 /** Resampling Filter Types */
118  AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
119  AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
120 };
121 
123  AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
124  AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
125  AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
126  AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
127  AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
128  AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
129 };
130 
131 /**
132  * Return the LIBAVRESAMPLE_VERSION_INT constant.
133  */
134 unsigned avresample_version(void);
135 
136 /**
137  * Return the libavresample build-time configuration.
138  * @return configure string
139  */
140 const char *avresample_configuration(void);
141 
142 /**
143  * Return the libavresample license.
144  */
145 const char *avresample_license(void);
146 
147 /**
148  * Get the AVClass for AVAudioResampleContext.
149  *
150  * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
151  * without allocating a context.
152  *
153  * @see av_opt_find().
154  *
155  * @return AVClass for AVAudioResampleContext
156  */
157 const AVClass *avresample_get_class(void);
158 
159 /**
160  * Allocate AVAudioResampleContext and set options.
161  *
162  * @return allocated audio resample context, or NULL on failure
163  */
165 
166 /**
167  * Initialize AVAudioResampleContext.
168  *
169  * @param avr audio resample context
170  * @return 0 on success, negative AVERROR code on failure
171  */
173 
174 /**
175  * Close AVAudioResampleContext.
176  *
177  * This closes the context, but it does not change the parameters. The context
178  * can be reopened with avresample_open(). It does, however, clear the output
179  * FIFO and any remaining leftover samples in the resampling delay buffer. If
180  * there was a custom matrix being used, that is also cleared.
181  *
182  * @see avresample_convert()
183  * @see avresample_set_matrix()
184  *
185  * @param avr audio resample context
186  */
188 
189 /**
190  * Free AVAudioResampleContext and associated AVOption values.
191  *
192  * This also calls avresample_close() before freeing.
193  *
194  * @param avr audio resample context
195  */
197 
198 /**
199  * Generate a channel mixing matrix.
200  *
201  * This function is the one used internally by libavresample for building the
202  * default mixing matrix. It is made public just as a utility function for
203  * building custom matrices.
204  *
205  * @param in_layout input channel layout
206  * @param out_layout output channel layout
207  * @param center_mix_level mix level for the center channel
208  * @param surround_mix_level mix level for the surround channel(s)
209  * @param lfe_mix_level mix level for the low-frequency effects channel
210  * @param normalize if 1, coefficients will be normalized to prevent
211  * overflow. if 0, coefficients will not be
212  * normalized.
213  * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
214  * the weight of input channel i in output channel o.
215  * @param stride distance between adjacent input channels in the
216  * matrix array
217  * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
218  * @return 0 on success, negative AVERROR code on failure
219  */
220 int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
221  double center_mix_level, double surround_mix_level,
222  double lfe_mix_level, int normalize, double *matrix,
224 
225 /**
226  * Get the current channel mixing matrix.
227  *
228  * If no custom matrix has been previously set or the AVAudioResampleContext is
229  * not open, an error is returned.
230  *
231  * @param avr audio resample context
232  * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
233  * input channel i in output channel o.
234  * @param stride distance between adjacent input channels in the matrix array
235  * @return 0 on success, negative AVERROR code on failure
236  */
237 int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
238  int stride);
239 
240 /**
241  * Set channel mixing matrix.
242  *
243  * Allows for setting a custom mixing matrix, overriding the default matrix
244  * generated internally during avresample_open(). This function can be called
245  * anytime on an allocated context, either before or after calling
246  * avresample_open(), as long as the channel layouts have been set.
247  * avresample_convert() always uses the current matrix.
248  * Calling avresample_close() on the context will clear the current matrix.
249  *
250  * @see avresample_close()
251  *
252  * @param avr audio resample context
253  * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
254  * input channel i in output channel o.
255  * @param stride distance between adjacent input channels in the matrix array
256  * @return 0 on success, negative AVERROR code on failure
257  */
258 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
259  int stride);
260 
261 /**
262  * Set compensation for resampling.
263  *
264  * This can be called anytime after avresample_open(). If resampling is not
265  * automatically enabled because of a sample rate conversion, the
266  * "force_resampling" option must have been set to 1 when opening the context
267  * in order to use resampling compensation.
268  *
269  * @param avr audio resample context
270  * @param sample_delta compensation delta, in samples
271  * @param compensation_distance compensation distance, in samples
272  * @return 0 on success, negative AVERROR code on failure
273  */
274 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
275  int compensation_distance);
276 
277 /**
278  * Convert input samples and write them to the output FIFO.
279  *
280  * The upper bound on the number of output samples is given by
281  * avresample_available() + (avresample_get_delay() + number of input samples) *
282  * output sample rate / input sample rate.
283  *
284  * The output data can be NULL or have fewer allocated samples than required.
285  * In this case, any remaining samples not written to the output will be added
286  * to an internal FIFO buffer, to be returned at the next call to this function
287  * or to avresample_read().
288  *
289  * If converting sample rate, there may be data remaining in the internal
290  * resampling delay buffer. avresample_get_delay() tells the number of remaining
291  * samples. To get this data as output, call avresample_convert() with NULL
292  * input.
293  *
294  * At the end of the conversion process, there may be data remaining in the
295  * internal FIFO buffer. avresample_available() tells the number of remaining
296  * samples. To get this data as output, either call avresample_convert() with
297  * NULL input or call avresample_read().
298  *
299  * @see avresample_available()
300  * @see avresample_read()
301  * @see avresample_get_delay()
302  *
303  * @param avr audio resample context
304  * @param output output data pointers
305  * @param out_plane_size output plane size, in bytes.
306  * This can be 0 if unknown, but that will lead to
307  * optimized functions not being used directly on the
308  * output, which could slow down some conversions.
309  * @param out_samples maximum number of samples that the output buffer can hold
310  * @param input input data pointers
311  * @param in_plane_size input plane size, in bytes
312  * This can be 0 if unknown, but that will lead to
313  * optimized functions not being used directly on the
314  * input, which could slow down some conversions.
315  * @param in_samples number of input samples to convert
316  * @return number of samples written to the output buffer,
317  * not including converted samples added to the internal
318  * output FIFO
319  */
321  int out_plane_size, int out_samples, uint8_t **input,
322  int in_plane_size, int in_samples);
323 
324 /**
325  * Return the number of samples currently in the resampling delay buffer.
326  *
327  * When resampling, there may be a delay between the input and output. Any
328  * unconverted samples in each call are stored internally in a delay buffer.
329  * This function allows the user to determine the current number of samples in
330  * the delay buffer, which can be useful for synchronization.
331  *
332  * @see avresample_convert()
333  *
334  * @param avr audio resample context
335  * @return number of samples currently in the resampling delay buffer
336  */
338 
339 /**
340  * Return the number of available samples in the output FIFO.
341  *
342  * During conversion, if the user does not specify an output buffer or
343  * specifies an output buffer that is smaller than what is needed, remaining
344  * samples that are not written to the output are stored to an internal FIFO
345  * buffer. The samples in the FIFO can be read with avresample_read() or
346  * avresample_convert().
347  *
348  * @see avresample_read()
349  * @see avresample_convert()
350  *
351  * @param avr audio resample context
352  * @return number of samples available for reading
353  */
355 
356 /**
357  * Read samples from the output FIFO.
358  *
359  * During conversion, if the user does not specify an output buffer or
360  * specifies an output buffer that is smaller than what is needed, remaining
361  * samples that are not written to the output are stored to an internal FIFO
362  * buffer. This function can be used to read samples from that internal FIFO.
363  *
364  * @see avresample_available()
365  * @see avresample_convert()
366  *
367  * @param avr audio resample context
368  * @param output output data pointers. May be NULL, in which case
369  * nb_samples of data is discarded from output FIFO.
370  * @param nb_samples number of samples to read from the FIFO
371  * @return the number of samples written to output
372  */
374 
375 /**
376  * @}
377  */
378 
379 #endif /* AVRESAMPLE_AVRESAMPLE_H */