2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 * This file is part of FFmpeg.
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.
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.
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
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_get_out_samples(avr, in_samples);
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);
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
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"
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 */
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. */
132 * Return the LIBAVRESAMPLE_VERSION_INT constant.
134 unsigned avresample_version(void);
137 * Return the libavresample build-time configuration.
138 * @return configure string
140 const char *avresample_configuration(void);
143 * Return the libavresample license.
145 const char *avresample_license(void);
148 * Get the AVClass for AVAudioResampleContext.
150 * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
151 * without allocating a context.
153 * @see av_opt_find().
155 * @return AVClass for AVAudioResampleContext
157 const AVClass *avresample_get_class(void);
160 * Allocate AVAudioResampleContext and set options.
162 * @return allocated audio resample context, or NULL on failure
164 AVAudioResampleContext *avresample_alloc_context(void);
167 * Initialize AVAudioResampleContext.
169 * @param avr audio resample context
170 * @return 0 on success, negative AVERROR code on failure
172 int avresample_open(AVAudioResampleContext *avr);
175 * Check whether an AVAudioResampleContext is open or closed.
177 * @param avr AVAudioResampleContext to check
178 * @return 1 if avr is open, 0 if avr is closed.
180 int avresample_is_open(AVAudioResampleContext *avr);
183 * Close AVAudioResampleContext.
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.
190 * @see avresample_convert()
191 * @see avresample_set_matrix()
193 * @param avr audio resample context
195 void avresample_close(AVAudioResampleContext *avr);
198 * Free AVAudioResampleContext and associated AVOption values.
200 * This also calls avresample_close() before freeing.
202 * @param avr audio resample context
204 void avresample_free(AVAudioResampleContext **avr);
207 * Generate a channel mixing matrix.
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.
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
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
225 * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
226 * @return 0 on success, negative AVERROR code on failure
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);
234 * Get the current channel mixing matrix.
236 * If no custom matrix has been previously set or the AVAudioResampleContext is
237 * not open, an error is returned.
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_get_matrix(AVAudioResampleContext *avr, double *matrix,
249 * Set channel mixing matrix.
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.
258 * @see avresample_close()
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
266 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
270 * Set a customized input channel mapping.
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.
275 * Calling avresample_close() on the context will clear the channel mapping.
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.
283 * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
284 * { 1, 2, 0, 5, 3, 4 }
286 * Muting the 3rd channel in 4-channel input:
289 * Duplicating the left channel of stereo input:
292 * @param avr audio resample context
293 * @param channel_map customized input channel mapping
294 * @return 0 on success, negative AVERROR code on failure
296 int avresample_set_channel_mapping(AVAudioResampleContext *avr,
297 const int *channel_map);
300 * Set compensation for resampling.
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.
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
312 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
313 int compensation_distance);
316 * Provide the upper bound on the number of samples the configured
317 * conversion would output.
319 * @param avr audio resample context
320 * @param in_nb_samples number of input samples
322 * @return number of samples or AVERROR(EINVAL) if the value
323 * would exceed INT_MAX
326 int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
329 * Convert input samples and write them to the output FIFO.
331 * The upper bound on the number of output samples can be obtained through
332 * avresample_get_out_samples().
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().
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
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().
349 * @see avresample_get_out_samples()
350 * @see avresample_read()
351 * @see avresample_get_delay()
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
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);
375 * Return the number of samples currently in the resampling delay buffer.
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.
382 * @see avresample_convert()
384 * @param avr audio resample context
385 * @return number of samples currently in the resampling delay buffer
387 int avresample_get_delay(AVAudioResampleContext *avr);
390 * Return the number of available samples in the output FIFO.
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().
398 * @see avresample_read()
399 * @see avresample_convert()
401 * @param avr audio resample context
402 * @return number of samples available for reading
404 int avresample_available(AVAudioResampleContext *avr);
407 * Read samples from the output FIFO.
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.
414 * @see avresample_available()
415 * @see avresample_convert()
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
423 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
429 #endif /* AVRESAMPLE_AVRESAMPLE_H */