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_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0); 48 * av_opt_set_channel_layout(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 "libavutil/samplefmt.h" 98 99 #include "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 * Check whether an swr context has been initialized or not. 170 * 171 * @return positive if it has been initialized, 0 if not initialized 172 */ 173 int swr_is_initialized(struct SwrContext *s); 174 175 /** 176 * Allocate SwrContext if needed and set/reset common parameters. 177 * 178 * This function does not require s to be allocated with swr_alloc(). On the 179 * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters 180 * on the allocated context. 181 * 182 * @param s Swr context, can be NULL 183 * @param out_ch_layout output channel layout (AV_CH_LAYOUT_*) 184 * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*). 185 * @param out_sample_rate output sample rate (frequency in Hz) 186 * @param in_ch_layout input channel layout (AV_CH_LAYOUT_*) 187 * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*). 188 * @param in_sample_rate input sample rate (frequency in Hz) 189 * @param log_offset logging level offset 190 * @param log_ctx parent logging context, can be NULL 191 * 192 * @see swr_init(), swr_free() 193 * @return NULL on error, allocated context otherwise 194 */ 195 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s, 196 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, 197 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, 198 int log_offset, void *log_ctx); 199 200 /** 201 * Free the given SwrContext and set the pointer to NULL. 202 */ 203 void swr_free(struct SwrContext **s); 204 205 /** 206 * Convert audio. 207 * 208 * in and in_count can be set to 0 to flush the last few samples out at the 209 * end. 210 * 211 * If more input is provided than output space then the input will be buffered. 212 * You can avoid this buffering by providing more output space than input. 213 * Convertion will run directly without copying whenever possible. 214 * 215 * @param s allocated Swr context, with parameters set 216 * @param out output buffers, only the first one need be set in case of packed audio 217 * @param out_count amount of space available for output in samples per channel 218 * @param in input buffers, only the first one need to be set in case of packed audio 219 * @param in_count number of input samples available in one channel 220 * 221 * @return number of samples output per channel, negative value on error 222 */ 223 int swr_convert(struct SwrContext *s, uint8_t **out, int out_count, 224 const uint8_t **in , int in_count); 225 226 /** 227 * Convert the next timestamp from input to output 228 * timestamps are in 1/(in_sample_rate * out_sample_rate) units. 229 * 230 * @note There are 2 slightly differently behaving modes. 231 * First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX) 232 * in this case timestamps will be passed through with delays compensated 233 * Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX) 234 * in this case the output timestamps will match output sample numbers 235 * 236 * @param pts timestamp for the next input sample, INT64_MIN if unknown 237 * @return the output timestamp for the next output sample 238 */ 239 int64_t swr_next_pts(struct SwrContext *s, int64_t pts); 240 241 /** 242 * Activate resampling compensation. 243 */ 244 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance); 245 246 /** 247 * Set a customized input channel mapping. 248 * 249 * @param s allocated Swr context, not yet initialized 250 * @param channel_map customized input channel mapping (array of channel 251 * indexes, -1 for a muted channel) 252 * @return AVERROR error code in case of failure. 253 */ 254 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map); 255 256 /** 257 * Set a customized remix matrix. 258 * 259 * @param s allocated Swr context, not yet initialized 260 * @param matrix remix coefficients; matrix[i + stride * o] is 261 * the weight of input channel i in output channel o 262 * @param stride offset between lines of the matrix 263 * @return AVERROR error code in case of failure. 264 */ 265 int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride); 266 267 /** 268 * Drops the specified number of output samples. 269 */ 270 int swr_drop_output(struct SwrContext *s, int count); 271 272 /** 273 * Injects the specified number of silence samples. 274 */ 275 int swr_inject_silence(struct SwrContext *s, int count); 276 277 /** 278 * Gets the delay the next input sample will experience relative to the next output sample. 279 * 280 * Swresample can buffer data if more input has been provided than available 281 * output space, also converting between sample rates needs a delay. 282 * This function returns the sum of all such delays. 283 * The exact delay is not necessarily an integer value in either input or 284 * output sample rate. Especially when downsampling by a large value, the 285 * output sample rate may be a poor choice to represent the delay, similarly 286 * for upsampling and the input sample rate. 287 * 288 * @param s swr context 289 * @param base timebase in which the returned delay will be 290 * if its set to 1 the returned delay is in seconds 291 * if its set to 1000 the returned delay is in milli seconds 292 * if its set to the input sample rate then the returned delay is in input samples 293 * if its set to the output sample rate then the returned delay is in output samples 294 * an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate) 295 * @returns the delay in 1/base units. 296 */ 297 int64_t swr_get_delay(struct SwrContext *s, int64_t base); 298 299 /** 300 * Return the LIBSWRESAMPLE_VERSION_INT constant. 301 */ 302 unsigned swresample_version(void); 303 304 /** 305 * Return the swr build-time configuration. 306 */ 307 const char *swresample_configuration(void); 308 309 /** 310 * Return the swr license. 311 */ 312 const char *swresample_license(void); 313 314 /** 315 * @} 316 */ 317 318 #endif /* SWRESAMPLE_SWRESAMPLE_H */ 319