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_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
100
102
103 #define AVRESAMPLE_MAX_CHANNELS 32
104
106
107 /** Mixing Coefficient Types */
113 };
114
115 /** Resampling Filter Types */
120 };
121
129 };
130
131 /**
132 * Return the LIBAVRESAMPLE_VERSION_INT constant.
133 */
135
136 /**
137 * Return the libavresample build-time configuration.
138 * @return configure string
139 */
141
142 /**
143 * Return the libavresample license.
144 */
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 */
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 */
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 */
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 */
260
261 /**
262 * Set a customized input channel mapping.
263 *
264 * This function can only be called when the allocated context is not open.
265 * Also, the input channel layout must have already been set.
266 *
267 * Calling avresample_close() on the context will clear the channel mapping.
268 *
269 * The map for each input channel specifies the channel index in the source to
270 * use for that particular channel, or -1 to mute the channel. Source channels
271 * can be duplicated by using the same index for multiple input channels.
272 *
273 * Examples:
274 *
275 * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
276 * { 1, 2, 0, 5, 3, 4 }
277 *
278 * Muting the 3rd channel in 4-channel input:
279 * { 0, 1, -1, 3 }
280 *
281 * Duplicating the left channel of stereo input:
282 * { 0, 0 }
283 *
284 * @param avr audio resample context
285 * @param channel_map customized input channel mapping
286 * @return 0 on success, negative AVERROR code on failure
287 */
289 const int *channel_map);
290
291 /**
292 * Set compensation for resampling.
293 *
294 * This can be called anytime after avresample_open(). If resampling is not
295 * automatically enabled because of a sample rate conversion, the
296 * "force_resampling" option must have been set to 1 when opening the context
297 * in order to use resampling compensation.
298 *
299 * @param avr audio resample context
300 * @param sample_delta compensation delta, in samples
301 * @param compensation_distance compensation distance, in samples
302 * @return 0 on success, negative AVERROR code on failure
303 */
305 int compensation_distance);
306
307 /**
308 * Convert input samples and write them to the output FIFO.
309 *
310 * The upper bound on the number of output samples is given by
311 * avresample_available() + (avresample_get_delay() + number of input samples) *
312 * output sample rate / input sample rate.
313 *
314 * The output data can be NULL or have fewer allocated samples than required.
315 * In this case, any remaining samples not written to the output will be added
316 * to an internal FIFO buffer, to be returned at the next call to this function
317 * or to avresample_read().
318 *
319 * If converting sample rate, there may be data remaining in the internal
320 * resampling delay buffer. avresample_get_delay() tells the number of remaining
321 * samples. To get this data as output, call avresample_convert() with NULL
322 * input.
323 *
324 * At the end of the conversion process, there may be data remaining in the
325 * internal FIFO buffer. avresample_available() tells the number of remaining
326 * samples. To get this data as output, either call avresample_convert() with
327 * NULL input or call avresample_read().
328 *
329 * @see avresample_available()
330 * @see avresample_read()
331 * @see avresample_get_delay()
332 *
333 * @param avr audio resample context
334 * @param output output data pointers
335 * @param out_plane_size output plane size, in bytes.
336 * This can be 0 if unknown, but that will lead to
337 * optimized functions not being used directly on the
338 * output, which could slow down some conversions.
339 * @param out_samples maximum number of samples that the output buffer can hold
340 * @param input input data pointers
341 * @param in_plane_size input plane size, in bytes
342 * This can be 0 if unknown, but that will lead to
343 * optimized functions not being used directly on the
344 * input, which could slow down some conversions.
345 * @param in_samples number of input samples to convert
346 * @return number of samples written to the output buffer,
347 * not including converted samples added to the internal
348 * output FIFO
349 */
351 int out_plane_size,
int out_samples,
uint8_t **input,
352 int in_plane_size, int in_samples);
353
354 /**
355 * Return the number of samples currently in the resampling delay buffer.
356 *
357 * When resampling, there may be a delay between the input and output. Any
358 * unconverted samples in each call are stored internally in a delay buffer.
359 * This function allows the user to determine the current number of samples in
360 * the delay buffer, which can be useful for synchronization.
361 *
362 * @see avresample_convert()
363 *
364 * @param avr audio resample context
365 * @return number of samples currently in the resampling delay buffer
366 */
368
369 /**
370 * Return the number of available samples in the output FIFO.
371 *
372 * During conversion, if the user does not specify an output buffer or
373 * specifies an output buffer that is smaller than what is needed, remaining
374 * samples that are not written to the output are stored to an internal FIFO
375 * buffer. The samples in the FIFO can be read with avresample_read() or
376 * avresample_convert().
377 *
378 * @see avresample_read()
379 * @see avresample_convert()
380 *
381 * @param avr audio resample context
382 * @return number of samples available for reading
383 */
385
386 /**
387 * Read samples from the output FIFO.
388 *
389 * During conversion, if the user does not specify an output buffer or
390 * specifies an output buffer that is smaller than what is needed, remaining
391 * samples that are not written to the output are stored to an internal FIFO
392 * buffer. This function can be used to read samples from that internal FIFO.
393 *
394 * @see avresample_available()
395 * @see avresample_convert()
396 *
397 * @param avr audio resample context
398 * @param output output data pointers. May be NULL, in which case
399 * nb_samples of data is discarded from output FIFO.
400 * @param nb_samples number of samples to read from the FIFO
401 * @return the number of samples written to output
402 */
404
405 /**
406 * @}
407 */
408
409 #endif /* AVRESAMPLE_AVRESAMPLE_H */