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preparation for libavcodec
author zipi
date Tue, 31 Dec 2013 14:52:14 +0000
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2133:e378232bfd36 2134:992d2e6f907d
1 /*
2 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3 *
4 * This file is part of libswresample
5 *
6 * libswresample 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 * libswresample 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 libswresample; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #ifndef SWRESAMPLE_SWRESAMPLE_H
22 #define SWRESAMPLE_SWRESAMPLE_H
23
24 /**
25 * @file
26 * @ingroup lswr
27 * libswresample public header
28 */
29
30 /**
31 * @defgroup lswr Libswresample
32 * @{
33 *
34 * Libswresample (lswr) is a library that handles audio resampling, sample
35 * format conversion and mixing.
36 *
37 * Interaction with lswr is done through SwrContext, which is
38 * allocated with swr_alloc() or swr_alloc_set_opts(). 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 * SwrContext *swr = swr_alloc();
47 * av_opt_set_int(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
48 * av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
49 * av_opt_set_int(swr, "in_sample_rate", 48000, 0);
50 * av_opt_set_int(swr, "out_sample_rate", 44100, 0);
51 * av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
52 * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
53 * @endcode
54 *
55 * Once all values have been set, it must be initialized with swr_init(). If
56 * you need to change the conversion parameters, you can change the parameters
57 * as described above, or by using swr_alloc_set_opts(), then call swr_init()
58 * again.
59 *
60 * The conversion itself is done by repeatedly calling swr_convert().
61 * Note that the samples may get buffered in swr if you provide insufficient
62 * output space or if sample rate conversion is done, which requires "future"
63 * samples. Samples that do not require future input can be retrieved at any
64 * time by using swr_convert() (in_count can be set to 0).
65 * At the end of conversion the resampling buffer can be flushed by calling
66 * swr_convert() with NULL in and 0 in_count.
67 *
68 * The delay between input and output, can at any time be found by using
69 * swr_get_delay().
70 *
71 * The following code demonstrates the conversion loop assuming the parameters
72 * from above and caller-defined functions get_input() and handle_output():
73 * @code
74 * uint8_t **input;
75 * int in_samples;
76 *
77 * while (get_input(&input, &in_samples)) {
78 * uint8_t *output;
79 * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
80 * in_samples, 44100, 48000, AV_ROUND_UP);
81 * av_samples_alloc(&output, NULL, 2, out_samples,
82 * AV_SAMPLE_FMT_S16, 0);
83 * out_samples = swr_convert(swr, &output, out_samples,
84 * input, in_samples);
85 * handle_output(output, out_samples);
86 * av_freep(&output);
87 * }
88 * @endcode
89 *
90 * When the conversion is finished, the conversion
91 * context and everything associated with it must be freed with swr_free().
92 * There will be no memory leak if the data is not completely flushed before
93 * swr_free().
94 */
95
96 #include <stdint.h>
97 #include "lib/libavutil/samplefmt.h"
98
99 #include "lib/libswresample/version.h"
100
101 #if LIBSWRESAMPLE_VERSION_MAJOR < 1
102 #define SWR_CH_MAX 32 ///< Maximum number of channels
103 #endif
104
105 #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
106 //TODO use int resample ?
107 //long term TODO can we enable this dynamically?
108
109 enum SwrDitherType {
110 SWR_DITHER_NONE = 0,
111 SWR_DITHER_RECTANGULAR,
112 SWR_DITHER_TRIANGULAR,
113 SWR_DITHER_TRIANGULAR_HIGHPASS,
114
115 SWR_DITHER_NS = 64, ///< not part of API/ABI
116 SWR_DITHER_NS_LIPSHITZ,
117 SWR_DITHER_NS_F_WEIGHTED,
118 SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
119 SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
120 SWR_DITHER_NS_SHIBATA,
121 SWR_DITHER_NS_LOW_SHIBATA,
122 SWR_DITHER_NS_HIGH_SHIBATA,
123 SWR_DITHER_NB, ///< not part of API/ABI
124 };
125
126 /** Resampling Engines */
127 enum SwrEngine {
128 SWR_ENGINE_SWR, /**< SW Resampler */
129 SWR_ENGINE_SOXR, /**< SoX Resampler */
130 SWR_ENGINE_NB, ///< not part of API/ABI
131 };
132
133 /** Resampling Filter Types */
134 enum SwrFilterType {
135 SWR_FILTER_TYPE_CUBIC, /**< Cubic */
136 SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
137 SWR_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
138 };
139
140 typedef struct SwrContext SwrContext;
141
142 /**
143 * Get the AVClass for swrContext. It can be used in combination with
144 * AV_OPT_SEARCH_FAKE_OBJ for examining options.
145 *
146 * @see av_opt_find().
147 */
148 const AVClass *swr_get_class(void);
149
150 /**
151 * Allocate SwrContext.
152 *
153 * If you use this function you will need to set the parameters (manually or
154 * with swr_alloc_set_opts()) before calling swr_init().
155 *
156 * @see swr_alloc_set_opts(), swr_init(), swr_free()
157 * @return NULL on error, allocated context otherwise
158 */
159 struct SwrContext *swr_alloc(void);
160
161 /**
162 * Initialize context after user parameters have been set.
163 *
164 * @return AVERROR error code in case of failure.
165 */
166 int swr_init(struct SwrContext *s);
167
168 /**
169 * Allocate SwrContext if needed and set/reset common parameters.
170 *
171 * This function does not require s to be allocated with swr_alloc(). On the
172 * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
173 * on the allocated context.
174 *
175 * @param s Swr context, can be NULL
176 * @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
177 * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
178 * @param out_sample_rate output sample rate (frequency in Hz)
179 * @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
180 * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
181 * @param in_sample_rate input sample rate (frequency in Hz)
182 * @param log_offset logging level offset
183 * @param log_ctx parent logging context, can be NULL
184 *
185 * @see swr_init(), swr_free()
186 * @return NULL on error, allocated context otherwise
187 */
188 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
189 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
190 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
191 int log_offset, void *log_ctx);
192
193 /**
194 * Free the given SwrContext and set the pointer to NULL.
195 */
196 void swr_free(struct SwrContext **s);
197
198 /**
199 * Convert audio.
200 *
201 * in and in_count can be set to 0 to flush the last few samples out at the
202 * end.
203 *
204 * If more input is provided than output space then the input will be buffered.
205 * You can avoid this buffering by providing more output space than input.
206 * Convertion will run directly without copying whenever possible.
207 *
208 * @param s allocated Swr context, with parameters set
209 * @param out output buffers, only the first one need be set in case of packed audio
210 * @param out_count amount of space available for output in samples per channel
211 * @param in input buffers, only the first one need to be set in case of packed audio
212 * @param in_count number of input samples available in one channel
213 *
214 * @return number of samples output per channel, negative value on error
215 */
216 int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
217 const uint8_t **in , int in_count);
218
219 /**
220 * Convert the next timestamp from input to output
221 * timestamps are in 1/(in_sample_rate * out_sample_rate) units.
222 *
223 * @note There are 2 slightly differently behaving modes.
224 * First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
225 * in this case timestamps will be passed through with delays compensated
226 * Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX)
227 * in this case the output timestamps will match output sample numbers
228 *
229 * @param pts timestamp for the next input sample, INT64_MIN if unknown
230 * @return the output timestamp for the next output sample
231 */
232 int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
233
234 /**
235 * Activate resampling compensation.
236 */
237 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
238
239 /**
240 * Set a customized input channel mapping.
241 *
242 * @param s allocated Swr context, not yet initialized
243 * @param channel_map customized input channel mapping (array of channel
244 * indexes, -1 for a muted channel)
245 * @return AVERROR error code in case of failure.
246 */
247 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
248
249 /**
250 * Set a customized remix matrix.
251 *
252 * @param s allocated Swr context, not yet initialized
253 * @param matrix remix coefficients; matrix[i + stride * o] is
254 * the weight of input channel i in output channel o
255 * @param stride offset between lines of the matrix
256 * @return AVERROR error code in case of failure.
257 */
258 int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
259
260 /**
261 * Drops the specified number of output samples.
262 */
263 int swr_drop_output(struct SwrContext *s, int count);
264
265 /**
266 * Injects the specified number of silence samples.
267 */
268 int swr_inject_silence(struct SwrContext *s, int count);
269
270 /**
271 * Gets the delay the next input sample will experience relative to the next output sample.
272 *
273 * Swresample can buffer data if more input has been provided than available
274 * output space, also converting between sample rates needs a delay.
275 * This function returns the sum of all such delays.
276 * The exact delay is not necessarily an integer value in either input or
277 * output sample rate. Especially when downsampling by a large value, the
278 * output sample rate may be a poor choice to represent the delay, similarly
279 * for upsampling and the input sample rate.
280 *
281 * @param s swr context
282 * @param base timebase in which the returned delay will be
283 * if its set to 1 the returned delay is in seconds
284 * if its set to 1000 the returned delay is in milli seconds
285 * if its set to the input sample rate then the returned delay is in input samples
286 * if its set to the output sample rate then the returned delay is in output samples
287 * an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
288 * @returns the delay in 1/base units.
289 */
290 int64_t swr_get_delay(struct SwrContext *s, int64_t base);
291
292 /**
293 * Return the LIBSWRESAMPLE_VERSION_INT constant.
294 */
295 unsigned swresample_version(void);
296
297 /**
298 * Return the swr build-time configuration.
299 */
300 const char *swresample_configuration(void);
301
302 /**
303 * Return the swr license.
304 */
305 const char *swresample_license(void);
306
307 /**
308 * @}
309 */
310
311 #endif /* SWRESAMPLE_SWRESAMPLE_H */