2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 * This file is part of Libav.
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.
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.
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
21 #ifndef AVRESAMPLE_AVRESAMPLE_H
22 #define AVRESAMPLE_AVRESAMPLE_H
31 * @defgroup lavr Libavresample
34 * Libavresample (lavr) is a library that handles audio resampling, sample
35 * format conversion and mixing.
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.
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
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);
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
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.
70 * The following code demonstrates the conversion loop assuming the parameters
71 * from above and caller-defined functions get_input() and handle_output():
74 * int in_linesize, in_samples;
76 * while (get_input(&input, &in_linesize, &in_samples)) {
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);
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
96 #include "libavutil/audioconvert.h"
97 #include "libavutil/avutil.h"
98 #include "libavutil/dict.h"
99 #include "libavutil/log.h"
101 #include "libavresample/version.h"
103 #define AVRESAMPLE_MAX_CHANNELS 32
105 typedef struct AVAudioResampleContext AVAudioResampleContext;
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 */
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 */
123 * Return the LIBAVRESAMPLE_VERSION_INT constant.
125 unsigned avresample_version(void);
128 * Return the libavresample build-time configuration.
129 * @return configure string
131 const char *avresample_configuration(void);
134 * Return the libavresample license.
136 const char *avresample_license(void);
139 * Get the AVClass for AVAudioResampleContext.
141 * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
142 * without allocating a context.
144 * @see av_opt_find().
146 * @return AVClass for AVAudioResampleContext
148 const AVClass *avresample_get_class(void);
151 * Allocate AVAudioResampleContext and set options.
153 * @return allocated audio resample context, or NULL on failure
155 AVAudioResampleContext *avresample_alloc_context(void);
158 * Initialize AVAudioResampleContext.
160 * @param avr audio resample context
161 * @return 0 on success, negative AVERROR code on failure
163 int avresample_open(AVAudioResampleContext *avr);
166 * Close AVAudioResampleContext.
168 * This closes the context, but it does not change the parameters. The context
169 * can be reopened with avresample_open(). It does, however, clear the output
170 * FIFO and any remaining leftover samples in the resampling delay buffer. If
171 * there was a custom matrix being used, that is also cleared.
173 * @see avresample_convert()
174 * @see avresample_set_matrix()
176 * @param avr audio resample context
178 void avresample_close(AVAudioResampleContext *avr);
181 * Free AVAudioResampleContext and associated AVOption values.
183 * This also calls avresample_close() before freeing.
185 * @param avr audio resample context
187 void avresample_free(AVAudioResampleContext **avr);
190 * Generate a channel mixing matrix.
192 * This function is the one used internally by libavresample for building the
193 * default mixing matrix. It is made public just as a utility function for
194 * building custom matrices.
196 * @param in_layout input channel layout
197 * @param out_layout output channel layout
198 * @param center_mix_level mix level for the center channel
199 * @param surround_mix_level mix level for the surround channel(s)
200 * @param lfe_mix_level mix level for the low-frequency effects channel
201 * @param normalize if 1, coefficients will be normalized to prevent
202 * overflow. if 0, coefficients will not be
204 * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
205 * the weight of input channel i in output channel o.
206 * @param stride distance between adjacent input channels in the
208 * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
209 * @return 0 on success, negative AVERROR code on failure
211 int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
212 double center_mix_level, double surround_mix_level,
213 double lfe_mix_level, int normalize, double *matrix,
214 int stride, enum AVMatrixEncoding matrix_encoding);
217 * Get the current channel mixing matrix.
219 * @param avr audio resample context
220 * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
221 * input channel i in output channel o.
222 * @param stride distance between adjacent input channels in the matrix array
223 * @return 0 on success, negative AVERROR code on failure
225 int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
229 * Set channel mixing matrix.
231 * Allows for setting a custom mixing matrix, overriding the default matrix
232 * generated internally during avresample_open(). This function can be called
233 * anytime on an allocated context, either before or after calling
234 * avresample_open(). avresample_convert() always uses the current matrix.
235 * Calling avresample_close() on the context will clear the current matrix.
237 * @see avresample_close()
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
245 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
249 * Set compensation for resampling.
251 * This can be called anytime after avresample_open(). If resampling was not
252 * being done previously, the AVAudioResampleContext is closed and reopened
253 * with resampling enabled. In this case, any samples remaining in the output
254 * FIFO and the current channel mixing matrix will be restored after reopening
257 * @param avr audio resample context
258 * @param sample_delta compensation delta, in samples
259 * @param compensation_distance compensation distance, in samples
260 * @return 0 on success, negative AVERROR code on failure
262 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
263 int compensation_distance);
266 * Convert input samples and write them to the output FIFO.
268 * The upper bound on the number of output samples is given by
269 * avresample_available() + (avresample_get_delay() + number of input samples) *
270 * output sample rate / input sample rate.
272 * The output data can be NULL or have fewer allocated samples than required.
273 * In this case, any remaining samples not written to the output will be added
274 * to an internal FIFO buffer, to be returned at the next call to this function
275 * or to avresample_read().
277 * If converting sample rate, there may be data remaining in the internal
278 * resampling delay buffer. avresample_get_delay() tells the number of remaining
279 * samples. To get this data as output, call avresample_convert() with NULL
282 * At the end of the conversion process, there may be data remaining in the
283 * internal FIFO buffer. avresample_available() tells the number of remaining
284 * samples. To get this data as output, either call avresample_convert() with
285 * NULL input or call avresample_read().
287 * @see avresample_available()
288 * @see avresample_read()
289 * @see avresample_get_delay()
291 * @param avr audio resample context
292 * @param output output data pointers
293 * @param out_plane_size output plane size, in bytes.
294 * This can be 0 if unknown, but that will lead to
295 * optimized functions not being used directly on the
296 * output, which could slow down some conversions.
297 * @param out_samples maximum number of samples that the output buffer can hold
298 * @param input input data pointers
299 * @param in_plane_size input plane size, in bytes
300 * This can be 0 if unknown, but that will lead to
301 * optimized functions not being used directly on the
302 * input, which could slow down some conversions.
303 * @param in_samples number of input samples to convert
304 * @return number of samples written to the output buffer,
305 * not including converted samples added to the internal
308 int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
309 int out_plane_size, int out_samples, uint8_t **input,
310 int in_plane_size, int in_samples);
313 * Return the number of samples currently in the resampling delay buffer.
315 * When resampling, there may be a delay between the input and output. Any
316 * unconverted samples in each call are stored internally in a delay buffer.
317 * This function allows the user to determine the current number of samples in
318 * the delay buffer, which can be useful for synchronization.
320 * @see avresample_convert()
322 * @param avr audio resample context
323 * @return number of samples currently in the resampling delay buffer
325 int avresample_get_delay(AVAudioResampleContext *avr);
328 * Return the number of available samples in the output FIFO.
330 * During conversion, if the user does not specify an output buffer or
331 * specifies an output buffer that is smaller than what is needed, remaining
332 * samples that are not written to the output are stored to an internal FIFO
333 * buffer. The samples in the FIFO can be read with avresample_read() or
334 * avresample_convert().
336 * @see avresample_read()
337 * @see avresample_convert()
339 * @param avr audio resample context
340 * @return number of samples available for reading
342 int avresample_available(AVAudioResampleContext *avr);
345 * Read samples from the output FIFO.
347 * During conversion, if the user does not specify an output buffer or
348 * specifies an output buffer that is smaller than what is needed, remaining
349 * samples that are not written to the output are stored to an internal FIFO
350 * buffer. This function can be used to read samples from that internal FIFO.
352 * @see avresample_available()
353 * @see avresample_convert()
355 * @param avr audio resample context
356 * @param output output data pointers. May be NULL, in which case
357 * nb_samples of data is discarded from output FIFO.
358 * @param nb_samples number of samples to read from the FIFO
359 * @return the number of samples written to output
361 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
367 #endif /* AVRESAMPLE_AVRESAMPLE_H */