| /* |
| * 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 |
| * @{ |
| * |
| * Libswresample (lswr) is a library that handles audio resampling, sample |
| * format conversion and mixing. |
| * |
| * Interaction with lswr is done through SwrContext, which is |
| * allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters |
| * must be set with the @ref avoptions API. |
| * |
| * 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): |
| * @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 |
| * |
| * 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 |
| * as described above, or by using swr_alloc_set_opts(), 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 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(). |
| * There will be no memory leak if the data is not completely flushed before |
| * swr_free(). |
| */ |
| |
| #include <stdint.h> |
| #include "libavutil/samplefmt.h" |
| |
| #include "libswresample/version.h" |
| |
| #if LIBSWRESAMPLE_VERSION_MAJOR < 1 |
| #define SWR_CH_MAX 32 ///< Maximum number of channels |
| #endif |
| |
| #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate |
| //TODO use int resample ? |
| //long term TODO can we enable this dynamically? |
| |
| 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 */ |
| }; |
| |
| 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(). |
| */ |
| const AVClass *swr_get_class(void); |
| |
| /** |
| * Allocate SwrContext. |
| * |
| * If you use this function you will need to set the parameters (manually or |
| * with swr_alloc_set_opts()) before calling swr_init(). |
| * |
| * @see swr_alloc_set_opts(), swr_init(), swr_free() |
| * @return NULL on error, allocated context otherwise |
| */ |
| struct SwrContext *swr_alloc(void); |
| |
| /** |
| * Initialize context after user parameters have been set. |
| * |
| * @return AVERROR error code in case of failure. |
| */ |
| int swr_init(struct SwrContext *s); |
| |
| /** |
| * Check whether an swr context has been initialized or not. |
| * |
| * @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 s to be allocated with swr_alloc(). On the |
| * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters |
| * on the allocated context. |
| * |
| * @param s Swr context, can be NULL |
| * @param out_ch_layout output channel layout (AV_CH_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 (AV_CH_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 NULL on error, allocated context otherwise |
| */ |
| struct SwrContext *swr_alloc_set_opts(struct SwrContext *s, |
| int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, |
| int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, |
| int log_offset, void *log_ctx); |
| |
| /** |
| * Free the given SwrContext and set the pointer to NULL. |
| */ |
| void swr_free(struct SwrContext **s); |
| |
| /** |
| * 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 providing more output space than input. |
| * Convertion 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 **out, int out_count, |
| const uint8_t **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. |
| * First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX) |
| * in this case timestamps will be passed through with delays compensated |
| * Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX) |
| * in this case the output timestamps will match output sample numbers |
| * |
| * @param pts timestamp for the next input sample, INT64_MIN if unknown |
| * @return the output timestamp for the next output sample |
| */ |
| int64_t swr_next_pts(struct SwrContext *s, int64_t pts); |
| |
| /** |
| * Activate resampling compensation. |
| */ |
| int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance); |
| |
| /** |
| * Set a customized input channel mapping. |
| * |
| * @param s allocated Swr context, not yet initialized |
| * @param channel_map customized input channel mapping (array of channel |
| * indexes, -1 for a muted channel) |
| * @return AVERROR error code in case of failure. |
| */ |
| int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map); |
| |
| /** |
| * 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 AVERROR error code in case of failure. |
| */ |
| int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride); |
| |
| /** |
| * Drops the specified number of output samples. |
| */ |
| int swr_drop_output(struct SwrContext *s, int count); |
| |
| /** |
| * Injects the specified number of silence samples. |
| */ |
| 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 |
| * if its set to 1 the returned delay is in seconds |
| * if its set to 1000 the returned delay is in milli seconds |
| * if its set to the input sample rate then the returned delay is in input samples |
| * if its set to the output sample rate then the returned delay is in output samples |
| * an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate) |
| * @returns the delay in 1/base units. |
| */ |
| int64_t swr_get_delay(struct SwrContext *s, int64_t base); |
| |
| /** |
| * Return the LIBSWRESAMPLE_VERSION_INT constant. |
| */ |
| unsigned swresample_version(void); |
| |
| /** |
| * Return the swr build-time configuration. |
| */ |
| const char *swresample_configuration(void); |
| |
| /** |
| * Return the swr license. |
| */ |
| const char *swresample_license(void); |
| |
| /** |
| * @} |
| */ |
| |
| #endif /* SWRESAMPLE_SWRESAMPLE_H */ |