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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"
96	#include "libavutil/channel_layout.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
106	typedef struct AVAudioResampleContext AVAudioResampleContext;
107
108	/** Mixing Coefficient Types */
109	enum AVMixCoeffType {
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 */
117	enum 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
123	enum 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
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	*/
165	AVAudioResampleContext *avresample_alloc_context(void);
166
167	/**
168	* Initialize AVAudioResampleContext.
169	* @note The context must be configured using the AVOption API.
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.
173	*
174	* @see av_opt_set_int()
175	* @see av_opt_set_dict()
176	* @see av_get_default_channel_layout()
177	*
178	* @param avr audio resample context
179	* @return 0 on success, negative AVERROR code on failure
180	*/
181	int avresample_open(AVAudioResampleContext *avr);
182
183	/**
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	*/
189	int avresample_is_open(AVAudioResampleContext *avr);
190
191	/**
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	*/
204	void 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	*/
213	void 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
234	* @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
235	* @return 0 on success, negative AVERROR code on failure
236	*/
237	int 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,
240	int stride, enum AVMatrixEncoding matrix_encoding);
241
242	/**
243	* Get the current channel mixing matrix.
244	*
245	* If no custom matrix has been previously set or the AVAudioResampleContext is
246	* not open, an error is returned.
247	*
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	*/
254	int 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
263	* avresample_open(), as long as the channel layouts have been set.
264	* avresample_convert() always uses the current matrix.
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	*/
275	int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
276	int stride);
277
278	/**
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 FFmpeg 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	*/
305	int avresample_set_channel_mapping(AVAudioResampleContext *avr,
306	const int *channel_map);
307
308	/**
309	* Set compensation for resampling.
310	*
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.
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	*/
321	int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
322	int compensation_distance);
323
324	/**
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
335	int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
336
337	/**
338	* Convert input samples and write them to the output FIFO.
339	*
340	* The upper bound on the number of output samples can be obtained through
341	* avresample_get_out_samples().
342	*
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	*
358	* @see avresample_get_out_samples()
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	*/
379	int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
380	int out_plane_size, int out_samples,
381	uint8_t * const *input, int in_plane_size,
382	int in_samples);
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	*/
397	int 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	*/
414	int 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
428	* @param output output data pointers. May be NULL, in which case
429	* nb_samples of data is discarded from output FIFO.
430	* @param nb_samples number of samples to read from the FIFO
431	* @return the number of samples written to output
432	*/
433	int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
434
435	/**
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	*/
479	int 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
493	* @param output output AVFrame
494	* @param input input AVFrame
495	* @return 0 on success, AVERROR on failure.
496	*/
497	int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in);
498
499	/**
500	* @}
501	*/
502
503	#endif /* AVRESAMPLE_AVRESAMPLE_H */
504