avresample: Add avresample_get_out_samples
[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/log.h"
99 #include "libavutil/mathematics.h"
100
101 #include "libavresample/version.h"
102
103 #define AVRESAMPLE_MAX_CHANNELS 32
104
105 typedef struct AVAudioResampleContext AVAudioResampleContext;
106
107 /** Mixing Coefficient Types */
108 enum AVMixCoeffType {
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 */
116 enum AVResampleFilterType {
117 AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
118 AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
119 AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
120 };
121
122 enum AVResampleDitherMethod {
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 */
164 AVAudioResampleContext *avresample_alloc_context(void);
165
166 /**
167 * Initialize AVAudioResampleContext.
168 *
169 * @param avr audio resample context
170 * @return 0 on success, negative AVERROR code on failure
171 */
172 int avresample_open(AVAudioResampleContext *avr);
173
174 /**
175 * Check whether an AVAudioResampleContext is open or closed.
176 *
177 * @param avr AVAudioResampleContext to check
178 * @return 1 if avr is open, 0 if avr is closed.
179 */
180 int avresample_is_open(AVAudioResampleContext *avr);
181
182 /**
183 * Close AVAudioResampleContext.
184 *
185 * This closes the context, but it does not change the parameters. The context
186 * can be reopened with avresample_open(). It does, however, clear the output
187 * FIFO and any remaining leftover samples in the resampling delay buffer. If
188 * there was a custom matrix being used, that is also cleared.
189 *
190 * @see avresample_convert()
191 * @see avresample_set_matrix()
192 *
193 * @param avr audio resample context
194 */
195 void avresample_close(AVAudioResampleContext *avr);
196
197 /**
198 * Free AVAudioResampleContext and associated AVOption values.
199 *
200 * This also calls avresample_close() before freeing.
201 *
202 * @param avr audio resample context
203 */
204 void avresample_free(AVAudioResampleContext **avr);
205
206 /**
207 * Generate a channel mixing matrix.
208 *
209 * This function is the one used internally by libavresample for building the
210 * default mixing matrix. It is made public just as a utility function for
211 * building custom matrices.
212 *
213 * @param in_layout input channel layout
214 * @param out_layout output channel layout
215 * @param center_mix_level mix level for the center channel
216 * @param surround_mix_level mix level for the surround channel(s)
217 * @param lfe_mix_level mix level for the low-frequency effects channel
218 * @param normalize if 1, coefficients will be normalized to prevent
219 * overflow. if 0, coefficients will not be
220 * normalized.
221 * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
222 * the weight of input channel i in output channel o.
223 * @param stride distance between adjacent input channels in the
224 * matrix array
225 * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
226 * @return 0 on success, negative AVERROR code on failure
227 */
228 int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
229 double center_mix_level, double surround_mix_level,
230 double lfe_mix_level, int normalize, double *matrix,
231 int stride, enum AVMatrixEncoding matrix_encoding);
232
233 /**
234 * Get the current channel mixing matrix.
235 *
236 * If no custom matrix has been previously set or the AVAudioResampleContext is
237 * not open, an error is returned.
238 *
239 * @param avr audio resample context
240 * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
241 * input channel i in output channel o.
242 * @param stride distance between adjacent input channels in the matrix array
243 * @return 0 on success, negative AVERROR code on failure
244 */
245 int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
246 int stride);
247
248 /**
249 * Set channel mixing matrix.
250 *
251 * Allows for setting a custom mixing matrix, overriding the default matrix
252 * generated internally during avresample_open(). This function can be called
253 * anytime on an allocated context, either before or after calling
254 * avresample_open(), as long as the channel layouts have been set.
255 * avresample_convert() always uses the current matrix.
256 * Calling avresample_close() on the context will clear the current matrix.
257 *
258 * @see avresample_close()
259 *
260 * @param avr audio resample context
261 * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
262 * input channel i in output channel o.
263 * @param stride distance between adjacent input channels in the matrix array
264 * @return 0 on success, negative AVERROR code on failure
265 */
266 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
267 int stride);
268
269 /**
270 * Set a customized input channel mapping.
271 *
272 * This function can only be called when the allocated context is not open.
273 * Also, the input channel layout must have already been set.
274 *
275 * Calling avresample_close() on the context will clear the channel mapping.
276 *
277 * The map for each input channel specifies the channel index in the source to
278 * use for that particular channel, or -1 to mute the channel. Source channels
279 * can be duplicated by using the same index for multiple input channels.
280 *
281 * Examples:
282 *
283 * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to Libav order (L,R,C,LFE,Ls,Rs):
284 * { 1, 2, 0, 5, 3, 4 }
285 *
286 * Muting the 3rd channel in 4-channel input:
287 * { 0, 1, -1, 3 }
288 *
289 * Duplicating the left channel of stereo input:
290 * { 0, 0 }
291 *
292 * @param avr audio resample context
293 * @param channel_map customized input channel mapping
294 * @return 0 on success, negative AVERROR code on failure
295 */
296 int avresample_set_channel_mapping(AVAudioResampleContext *avr,
297 const int *channel_map);
298
299 /**
300 * Set compensation for resampling.
301 *
302 * This can be called anytime after avresample_open(). If resampling is not
303 * automatically enabled because of a sample rate conversion, the
304 * "force_resampling" option must have been set to 1 when opening the context
305 * in order to use resampling compensation.
306 *
307 * @param avr audio resample context
308 * @param sample_delta compensation delta, in samples
309 * @param compensation_distance compensation distance, in samples
310 * @return 0 on success, negative AVERROR code on failure
311 */
312 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
313 int compensation_distance);
314
315 /**
316 * Provide the upper bound on the number of samples the configured
317 * conversion would output.
318 *
319 * @param avr audio resample context
320 * @param in_nb_samples number of input samples
321 *
322 * @return number of samples or AVERROR(EINVAL) if the value
323 * would exceed INT_MAX
324 */
325
326 int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
327
328 /**
329 * Convert input samples and write them to the output FIFO.
330 *
331 * The upper bound on the number of output samples can be obtained through
332 * avresample_get_out_samples().
333 *
334 * The output data can be NULL or have fewer allocated samples than required.
335 * In this case, any remaining samples not written to the output will be added
336 * to an internal FIFO buffer, to be returned at the next call to this function
337 * or to avresample_read().
338 *
339 * If converting sample rate, there may be data remaining in the internal
340 * resampling delay buffer. avresample_get_delay() tells the number of remaining
341 * samples. To get this data as output, call avresample_convert() with NULL
342 * input.
343 *
344 * At the end of the conversion process, there may be data remaining in the
345 * internal FIFO buffer. avresample_available() tells the number of remaining
346 * samples. To get this data as output, either call avresample_convert() with
347 * NULL input or call avresample_read().
348 *
349 * @see avresample_get_out_samples()
350 * @see avresample_read()
351 * @see avresample_get_delay()
352 *
353 * @param avr audio resample context
354 * @param output output data pointers
355 * @param out_plane_size output plane size, in bytes.
356 * This can be 0 if unknown, but that will lead to
357 * optimized functions not being used directly on the
358 * output, which could slow down some conversions.
359 * @param out_samples maximum number of samples that the output buffer can hold
360 * @param input input data pointers
361 * @param in_plane_size input plane size, in bytes
362 * This can be 0 if unknown, but that will lead to
363 * optimized functions not being used directly on the
364 * input, which could slow down some conversions.
365 * @param in_samples number of input samples to convert
366 * @return number of samples written to the output buffer,
367 * not including converted samples added to the internal
368 * output FIFO
369 */
370 int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
371 int out_plane_size, int out_samples, uint8_t **input,
372 int in_plane_size, int in_samples);
373
374 /**
375 * Return the number of samples currently in the resampling delay buffer.
376 *
377 * When resampling, there may be a delay between the input and output. Any
378 * unconverted samples in each call are stored internally in a delay buffer.
379 * This function allows the user to determine the current number of samples in
380 * the delay buffer, which can be useful for synchronization.
381 *
382 * @see avresample_convert()
383 *
384 * @param avr audio resample context
385 * @return number of samples currently in the resampling delay buffer
386 */
387 int avresample_get_delay(AVAudioResampleContext *avr);
388
389 /**
390 * Return the number of available samples in the output FIFO.
391 *
392 * During conversion, if the user does not specify an output buffer or
393 * specifies an output buffer that is smaller than what is needed, remaining
394 * samples that are not written to the output are stored to an internal FIFO
395 * buffer. The samples in the FIFO can be read with avresample_read() or
396 * avresample_convert().
397 *
398 * @see avresample_read()
399 * @see avresample_convert()
400 *
401 * @param avr audio resample context
402 * @return number of samples available for reading
403 */
404 int avresample_available(AVAudioResampleContext *avr);
405
406 /**
407 * Read samples from the output FIFO.
408 *
409 * During conversion, if the user does not specify an output buffer or
410 * specifies an output buffer that is smaller than what is needed, remaining
411 * samples that are not written to the output are stored to an internal FIFO
412 * buffer. This function can be used to read samples from that internal FIFO.
413 *
414 * @see avresample_available()
415 * @see avresample_convert()
416 *
417 * @param avr audio resample context
418 * @param output output data pointers. May be NULL, in which case
419 * nb_samples of data is discarded from output FIFO.
420 * @param nb_samples number of samples to read from the FIFO
421 * @return the number of samples written to output
422 */
423 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
424
425 /**
426 * @}
427 */
428
429 #endif /* AVRESAMPLE_AVRESAMPLE_H */