/* * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) * * This file is part of libswresample * * libswresample is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * libswresample is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with libswresample; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #ifndef SWRESAMPLE_SWRESAMPLE_H #define SWRESAMPLE_SWRESAMPLE_H /** * @file * @ingroup lswr * libswresample public header */ /** * @defgroup lswr libswresample * @{ * * Audio resampling, sample format conversion and mixing library. * * Interaction with lswr is done through SwrContext, which is * allocated with swr_alloc() or swr_alloc_set_opts2(). It is opaque, so all parameters * must be set with the @ref avoptions API. * * The first thing you will need to do in order to use lswr is to allocate * SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts2(). If you * are using the former, you must set options through the @ref avoptions API. * The latter function provides the same feature, but it allows you to set some * common options in the same statement. * * For example the following code will setup conversion from planar float sample * format to interleaved signed 16-bit integer, downsampling from 48kHz to * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing * matrix). This is using the swr_alloc() function. * @code * SwrContext *swr = swr_alloc(); * av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0); * av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0); * av_opt_set_int(swr, "in_sample_rate", 48000, 0); * av_opt_set_int(swr, "out_sample_rate", 44100, 0); * av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); * @endcode * * The same job can be done using swr_alloc_set_opts2() as well: * @code * SwrContext *swr = NULL; * int ret = swr_alloc_set_opts2(&swr, // we're allocating a new context * &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO, // out_ch_layout * AV_SAMPLE_FMT_S16, // out_sample_fmt * 44100, // out_sample_rate * &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1, // in_ch_layout * AV_SAMPLE_FMT_FLTP, // in_sample_fmt * 48000, // in_sample_rate * 0, // log_offset * NULL); // log_ctx * @endcode * * Once all values have been set, it must be initialized with swr_init(). If * you need to change the conversion parameters, you can change the parameters * using @ref avoptions, as described above in the first example; or by using * swr_alloc_set_opts2(), but with the first argument the allocated context. * You must then call swr_init() again. * * The conversion itself is done by repeatedly calling swr_convert(). * Note that the samples may get buffered in swr if you provide insufficient * output space or if sample rate conversion is done, which requires "future" * samples. Samples that do not require future input can be retrieved at any * time by using swr_convert() (in_count can be set to 0). * At the end of conversion the resampling buffer can be flushed by calling * swr_convert() with NULL in and 0 in_count. * * The samples used in the conversion process can be managed with the libavutil * @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc() * function used in the following example. * * The delay between input and output, can at any time be found by using * swr_get_delay(). * * The following code demonstrates the conversion loop assuming the parameters * from above and caller-defined functions get_input() and handle_output(): * @code * uint8_t **input; * int in_samples; * * while (get_input(&input, &in_samples)) { * uint8_t *output; * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) + * in_samples, 44100, 48000, AV_ROUND_UP); * av_samples_alloc(&output, NULL, 2, out_samples, * AV_SAMPLE_FMT_S16, 0); * out_samples = swr_convert(swr, &output, out_samples, * input, in_samples); * handle_output(output, out_samples); * av_freep(&output); * } * @endcode * * When the conversion is finished, the conversion * context and everything associated with it must be freed with swr_free(). * A swr_close() function is also available, but it exists mainly for * compatibility with libavresample, and is not required to be called. * * There will be no memory leak if the data is not completely flushed before * swr_free(). */ #include #include "libavutil/channel_layout.h" #include "libavutil/frame.h" #include "libavutil/samplefmt.h" #include "libswresample/version_major.h" #ifndef HAVE_AV_CONFIG_H /* When included as part of the ffmpeg build, only include the major version * to avoid unnecessary rebuilds. When included externally, keep including * the full version information. */ #include "libswresample/version.h" #endif /** * @name Option constants * These constants are used for the @ref avoptions interface for lswr. * @{ * */ #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate //TODO use int resample ? //long term TODO can we enable this dynamically? /** Dithering algorithms */ enum SwrDitherType { SWR_DITHER_NONE = 0, SWR_DITHER_RECTANGULAR, SWR_DITHER_TRIANGULAR, SWR_DITHER_TRIANGULAR_HIGHPASS, SWR_DITHER_NS = 64, ///< not part of API/ABI SWR_DITHER_NS_LIPSHITZ, SWR_DITHER_NS_F_WEIGHTED, SWR_DITHER_NS_MODIFIED_E_WEIGHTED, SWR_DITHER_NS_IMPROVED_E_WEIGHTED, SWR_DITHER_NS_SHIBATA, SWR_DITHER_NS_LOW_SHIBATA, SWR_DITHER_NS_HIGH_SHIBATA, SWR_DITHER_NB, ///< not part of API/ABI }; /** Resampling Engines */ enum SwrEngine { SWR_ENGINE_SWR, /**< SW Resampler */ SWR_ENGINE_SOXR, /**< SoX Resampler */ SWR_ENGINE_NB, ///< not part of API/ABI }; /** Resampling Filter Types */ enum SwrFilterType { SWR_FILTER_TYPE_CUBIC, /**< Cubic */ SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall windowed sinc */ SWR_FILTER_TYPE_KAISER, /**< Kaiser windowed sinc */ }; /** * @} */ /** * The libswresample context. Unlike libavcodec and libavformat, this structure * is opaque. This means that if you would like to set options, you must use * the @ref avoptions API and cannot directly set values to members of the * structure. */ typedef struct SwrContext SwrContext; /** * Get the AVClass for SwrContext. It can be used in combination with * AV_OPT_SEARCH_FAKE_OBJ for examining options. * * @see av_opt_find(). * @return the AVClass of SwrContext */ const AVClass *swr_get_class(void); /** * @name SwrContext constructor functions * @{ */ /** * Allocate SwrContext. * * If you use this function you will need to set the parameters (manually or * with swr_alloc_set_opts2()) before calling swr_init(). * * @see swr_alloc_set_opts2(), swr_init(), swr_free() * @return NULL on error, allocated context otherwise */ struct SwrContext *swr_alloc(void); /** * Initialize context after user parameters have been set. * @note The context must be configured using the AVOption API. * * @see av_opt_set_int() * @see av_opt_set_dict() * * @param[in,out] s Swr context to initialize * @return AVERROR error code in case of failure. */ int swr_init(struct SwrContext *s); /** * Check whether an swr context has been initialized or not. * * @param[in] s Swr context to check * @see swr_init() * @return positive if it has been initialized, 0 if not initialized */ int swr_is_initialized(struct SwrContext *s); /** * Allocate SwrContext if needed and set/reset common parameters. * * This function does not require *ps to be allocated with swr_alloc(). On the * other hand, swr_alloc() can use swr_alloc_set_opts2() to set the parameters * on the allocated context. * * @param ps Pointer to an existing Swr context if available, or to NULL if not. * On success, *ps will be set to the allocated context. * @param out_ch_layout output channel layout (e.g. AV_CHANNEL_LAYOUT_*) * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*). * @param out_sample_rate output sample rate (frequency in Hz) * @param in_ch_layout input channel layout (e.g. AV_CHANNEL_LAYOUT_*) * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*). * @param in_sample_rate input sample rate (frequency in Hz) * @param log_offset logging level offset * @param log_ctx parent logging context, can be NULL * * @see swr_init(), swr_free() * @return 0 on success, a negative AVERROR code on error. * On error, the Swr context is freed and *ps set to NULL. */ int swr_alloc_set_opts2(struct SwrContext **ps, const AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, const AVChannelLayout *in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx); /** * @} * * @name SwrContext destructor functions * @{ */ /** * Free the given SwrContext and set the pointer to NULL. * * @param[in] s a pointer to a pointer to Swr context */ void swr_free(struct SwrContext **s); /** * Closes the context so that swr_is_initialized() returns 0. * * The context can be brought back to life by running swr_init(), * swr_init() can also be used without swr_close(). * This function is mainly provided for simplifying the usecase * where one tries to support libavresample and libswresample. * * @param[in,out] s Swr context to be closed */ void swr_close(struct SwrContext *s); /** * @} * * @name Core conversion functions * @{ */ /** Convert audio. * * in and in_count can be set to 0 to flush the last few samples out at the * end. * * If more input is provided than output space, then the input will be buffered. * You can avoid this buffering by using swr_get_out_samples() to retrieve an * upper bound on the required number of output samples for the given number of * input samples. Conversion will run directly without copying whenever possible. * * @param s allocated Swr context, with parameters set * @param out output buffers, only the first one need be set in case of packed audio * @param out_count amount of space available for output in samples per channel * @param in input buffers, only the first one need to be set in case of packed audio * @param in_count number of input samples available in one channel * * @return number of samples output per channel, negative value on error */ int swr_convert(struct SwrContext *s, uint8_t * const *out, int out_count, const uint8_t * const *in , int in_count); /** * Convert the next timestamp from input to output * timestamps are in 1/(in_sample_rate * out_sample_rate) units. * * @note There are 2 slightly differently behaving modes. * @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX) * in this case timestamps will be passed through with delays compensated * @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX) * in this case the output timestamps will match output sample numbers. * See ffmpeg-resampler(1) for the two modes of compensation. * * @param[in] s initialized Swr context * @param[in] pts timestamp for the next input sample, INT64_MIN if unknown * @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are * function used internally for timestamp compensation. * @return the output timestamp for the next output sample */ int64_t swr_next_pts(struct SwrContext *s, int64_t pts); /** * @} * * @name Low-level option setting functions * These functons provide a means to set low-level options that is not possible * with the AVOption API. * @{ */ /** * Activate resampling compensation ("soft" compensation). This function is * internally called when needed in swr_next_pts(). * * @param[in,out] s allocated Swr context. If it is not initialized, * or SWR_FLAG_RESAMPLE is not set, swr_init() is * called with the flag set. * @param[in] sample_delta delta in PTS per sample * @param[in] compensation_distance number of samples to compensate for * @return >= 0 on success, AVERROR error codes if: * @li @c s is NULL, * @li @c compensation_distance is less than 0, * @li @c compensation_distance is 0 but sample_delta is not, * @li compensation unsupported by resampler, or * @li swr_init() fails when called. */ int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance); /** * Set a customized input channel mapping. * * @param[in,out] s allocated Swr context, not yet initialized * @param[in] channel_map customized input channel mapping (array of channel * indexes, -1 for a muted channel) * @return >= 0 on success, or AVERROR error code in case of failure. */ int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map); /** * Generate a channel mixing matrix. * * This function is the one used internally by libswresample for building the * default mixing matrix. It is made public just as a utility function for * building custom matrices. * * @param in_layout input channel layout * @param out_layout output channel layout * @param center_mix_level mix level for the center channel * @param surround_mix_level mix level for the surround channel(s) * @param lfe_mix_level mix level for the low-frequency effects channel * @param rematrix_maxval if 1.0, coefficients will be normalized to prevent * overflow. if INT_MAX, coefficients will not be * normalized. * @param[out] matrix mixing coefficients; matrix[i + stride * o] is * the weight of input channel i in output channel o. * @param stride distance between adjacent input channels in the * matrix array * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii) * @param log_ctx parent logging context, can be NULL * @return 0 on success, negative AVERROR code on failure */ int swr_build_matrix2(const AVChannelLayout *in_layout, const AVChannelLayout *out_layout, double center_mix_level, double surround_mix_level, double lfe_mix_level, double maxval, double rematrix_volume, double *matrix, ptrdiff_t stride, enum AVMatrixEncoding matrix_encoding, void *log_context); /** * Set a customized remix matrix. * * @param s allocated Swr context, not yet initialized * @param matrix remix coefficients; matrix[i + stride * o] is * the weight of input channel i in output channel o * @param stride offset between lines of the matrix * @return >= 0 on success, or AVERROR error code in case of failure. */ int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride); /** * @} * * @name Sample handling functions * @{ */ /** * Drops the specified number of output samples. * * This function, along with swr_inject_silence(), is called by swr_next_pts() * if needed for "hard" compensation. * * @param s allocated Swr context * @param count number of samples to be dropped * * @return >= 0 on success, or a negative AVERROR code on failure */ int swr_drop_output(struct SwrContext *s, int count); /** * Injects the specified number of silence samples. * * This function, along with swr_drop_output(), is called by swr_next_pts() * if needed for "hard" compensation. * * @param s allocated Swr context * @param count number of samples to be dropped * * @return >= 0 on success, or a negative AVERROR code on failure */ int swr_inject_silence(struct SwrContext *s, int count); /** * Gets the delay the next input sample will experience relative to the next output sample. * * Swresample can buffer data if more input has been provided than available * output space, also converting between sample rates needs a delay. * This function returns the sum of all such delays. * The exact delay is not necessarily an integer value in either input or * output sample rate. Especially when downsampling by a large value, the * output sample rate may be a poor choice to represent the delay, similarly * for upsampling and the input sample rate. * * @param s swr context * @param base timebase in which the returned delay will be: * @li if it's set to 1 the returned delay is in seconds * @li if it's set to 1000 the returned delay is in milliseconds * @li if it's set to the input sample rate then the returned * delay is in input samples * @li if it's set to the output sample rate then the returned * delay is in output samples * @li if it's the least common multiple of in_sample_rate and * out_sample_rate then an exact rounding-free delay will be * returned * @returns the delay in 1 / @c base units. */ int64_t swr_get_delay(struct SwrContext *s, int64_t base); /** * Find an upper bound on the number of samples that the next swr_convert * call will output, if called with in_samples of input samples. This * depends on the internal state, and anything changing the internal state * (like further swr_convert() calls) will may change the number of samples * swr_get_out_samples() returns for the same number of input samples. * * @param in_samples number of input samples. * @note any call to swr_inject_silence(), swr_convert(), swr_next_pts() * or swr_set_compensation() invalidates this limit * @note it is recommended to pass the correct available buffer size * to all functions like swr_convert() even if swr_get_out_samples() * indicates that less would be used. * @returns an upper bound on the number of samples that the next swr_convert * will output or a negative value to indicate an error */ int swr_get_out_samples(struct SwrContext *s, int in_samples); /** * @} * * @name Configuration accessors * @{ */ /** * Return the @ref LIBSWRESAMPLE_VERSION_INT constant. * * This is useful to check if the build-time libswresample has the same version * as the run-time one. * * @returns the unsigned int-typed version */ unsigned swresample_version(void); /** * Return the swr build-time configuration. * * @returns the build-time @c ./configure flags */ const char *swresample_configuration(void); /** * Return the swr license. * * @returns the license of libswresample, determined at build-time */ const char *swresample_license(void); /** * @} * * @name AVFrame based API * @{ */ /** * Convert the samples in the input AVFrame and write them to the output AVFrame. * * Input and output AVFrames must have channel_layout, sample_rate and format set. * * If the output AVFrame does not have the data pointers allocated the nb_samples * field will be set using av_frame_get_buffer() * is called to allocate the frame. * * The output AVFrame can be NULL or have fewer allocated samples than required. * In this case, any remaining samples not written to the output will be added * to an internal FIFO buffer, to be returned at the next call to this function * or to swr_convert(). * * If converting sample rate, there may be data remaining in the internal * resampling delay buffer. swr_get_delay() tells the number of * remaining samples. To get this data as output, call this function or * swr_convert() with NULL input. * * If the SwrContext configuration does not match the output and * input AVFrame settings the conversion does not take place and depending on * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED * or the result of a bitwise-OR of them is returned. * * @see swr_delay() * @see swr_convert() * @see swr_get_delay() * * @param swr audio resample context * @param output output AVFrame * @param input input AVFrame * @return 0 on success, AVERROR on failure or nonmatching * configuration. */ int swr_convert_frame(SwrContext *swr, AVFrame *output, const AVFrame *input); /** * Configure or reconfigure the SwrContext using the information * provided by the AVFrames. * * The original resampling context is reset even on failure. * The function calls swr_close() internally if the context is open. * * @see swr_close(); * * @param swr audio resample context * @param out output AVFrame * @param in input AVFrame * @return 0 on success, AVERROR on failure. */ int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in); /** * @} * @} */ #endif /* SWRESAMPLE_SWRESAMPLE_H */