| /* |
| * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
| * |
| * This file is part of FFmpeg. |
| * |
| * FFmpeg 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. |
| * |
| * FFmpeg 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 FFmpeg; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| |
| #ifndef AVRESAMPLE_AVRESAMPLE_H |
| #define AVRESAMPLE_AVRESAMPLE_H |
| |
| /** |
| * @file |
| * @ingroup lavr |
| * external API header |
| */ |
| |
| /** |
| * @defgroup lavr libavresample |
| * @{ |
| * |
| * Libavresample (lavr) is a library that handles audio resampling, sample |
| * format conversion and mixing. |
| * |
| * Interaction with lavr is done through AVAudioResampleContext, which is |
| * allocated with avresample_alloc_context(). 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 |
| * AVAudioResampleContext *avr = avresample_alloc_context(); |
| * av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0); |
| * av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0); |
| * av_opt_set_int(avr, "in_sample_rate", 48000, 0); |
| * av_opt_set_int(avr, "out_sample_rate", 44100, 0); |
| * av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); |
| * av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); |
| * @endcode |
| * |
| * Once the context is initialized, it must be opened with avresample_open(). If |
| * you need to change the conversion parameters, you must close the context with |
| * avresample_close(), change the parameters as described above, then reopen it |
| * again. |
| * |
| * The conversion itself is done by repeatedly calling avresample_convert(). |
| * Note that the samples may get buffered in two places in lavr. The first one |
| * is the output FIFO, where the samples end up if the output buffer is not |
| * large enough. The data stored in there may be retrieved at any time with |
| * avresample_read(). The second place is the resampling delay buffer, |
| * applicable only when resampling is done. The samples in it require more input |
| * before they can be processed. Their current amount is returned by |
| * avresample_get_delay(). At the end of conversion the resampling buffer can be |
| * flushed by calling avresample_convert() with NULL input. |
| * |
| * 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_linesize, in_samples; |
| * |
| * while (get_input(&input, &in_linesize, &in_samples)) { |
| * uint8_t *output |
| * int out_linesize; |
| * int out_samples = avresample_get_out_samples(avr, in_samples); |
| * |
| * av_samples_alloc(&output, &out_linesize, 2, out_samples, |
| * AV_SAMPLE_FMT_S16, 0); |
| * out_samples = avresample_convert(avr, &output, out_linesize, out_samples, |
| * input, in_linesize, in_samples); |
| * handle_output(output, out_linesize, out_samples); |
| * av_freep(&output); |
| * } |
| * @endcode |
| * |
| * When the conversion is finished and the FIFOs are flushed if required, the |
| * conversion context and everything associated with it must be freed with |
| * avresample_free(). |
| */ |
| |
| #include "libavutil/avutil.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/dict.h" |
| #include "libavutil/frame.h" |
| #include "libavutil/log.h" |
| #include "libavutil/mathematics.h" |
| |
| #include "libavresample/version.h" |
| |
| #define AVRESAMPLE_MAX_CHANNELS 32 |
| |
| typedef struct AVAudioResampleContext AVAudioResampleContext; |
| |
| /** Mixing Coefficient Types */ |
| enum AVMixCoeffType { |
| AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */ |
| AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */ |
| AV_MIX_COEFF_TYPE_FLT, /** floating-point */ |
| AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */ |
| }; |
| |
| /** Resampling Filter Types */ |
| enum AVResampleFilterType { |
| AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */ |
| AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */ |
| AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */ |
| }; |
| |
| enum AVResampleDitherMethod { |
| AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */ |
| AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */ |
| AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/ |
| AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */ |
| AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */ |
| AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */ |
| }; |
| |
| /** |
| * Return the LIBAVRESAMPLE_VERSION_INT constant. |
| */ |
| unsigned avresample_version(void); |
| |
| /** |
| * Return the libavresample build-time configuration. |
| * @return configure string |
| */ |
| const char *avresample_configuration(void); |
| |
| /** |
| * Return the libavresample license. |
| */ |
| const char *avresample_license(void); |
| |
| /** |
| * Get the AVClass for AVAudioResampleContext. |
| * |
| * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options |
| * without allocating a context. |
| * |
| * @see av_opt_find(). |
| * |
| * @return AVClass for AVAudioResampleContext |
| */ |
| const AVClass *avresample_get_class(void); |
| |
| /** |
| * Allocate AVAudioResampleContext and set options. |
| * |
| * @return allocated audio resample context, or NULL on failure |
| */ |
| AVAudioResampleContext *avresample_alloc_context(void); |
| |
| /** |
| * Initialize AVAudioResampleContext. |
| * @note The context must be configured using the AVOption API. |
| * @note The fields "in_channel_layout", "out_channel_layout", |
| * "in_sample_rate", "out_sample_rate", "in_sample_fmt", |
| * "out_sample_fmt" must be set. |
| * |
| * @see av_opt_set_int() |
| * @see av_opt_set_dict() |
| * @see av_get_default_channel_layout() |
| * |
| * @param avr audio resample context |
| * @return 0 on success, negative AVERROR code on failure |
| */ |
| int avresample_open(AVAudioResampleContext *avr); |
| |
| /** |
| * Check whether an AVAudioResampleContext is open or closed. |
| * |
| * @param avr AVAudioResampleContext to check |
| * @return 1 if avr is open, 0 if avr is closed. |
| */ |
| int avresample_is_open(AVAudioResampleContext *avr); |
| |
| /** |
| * Close AVAudioResampleContext. |
| * |
| * This closes the context, but it does not change the parameters. The context |
| * can be reopened with avresample_open(). It does, however, clear the output |
| * FIFO and any remaining leftover samples in the resampling delay buffer. If |
| * there was a custom matrix being used, that is also cleared. |
| * |
| * @see avresample_convert() |
| * @see avresample_set_matrix() |
| * |
| * @param avr audio resample context |
| */ |
| void avresample_close(AVAudioResampleContext *avr); |
| |
| /** |
| * Free AVAudioResampleContext and associated AVOption values. |
| * |
| * This also calls avresample_close() before freeing. |
| * |
| * @param avr audio resample context |
| */ |
| void avresample_free(AVAudioResampleContext **avr); |
| |
| /** |
| * Generate a channel mixing matrix. |
| * |
| * This function is the one used internally by libavresample 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 normalize if 1, coefficients will be normalized to prevent |
| * overflow. if 0, 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) |
| * @return 0 on success, negative AVERROR code on failure |
| */ |
| int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout, |
| double center_mix_level, double surround_mix_level, |
| double lfe_mix_level, int normalize, double *matrix, |
| int stride, enum AVMatrixEncoding matrix_encoding); |
| |
| /** |
| * Get the current channel mixing matrix. |
| * |
| * If no custom matrix has been previously set or the AVAudioResampleContext is |
| * not open, an error is returned. |
| * |
| * @param avr audio resample context |
| * @param 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 |
| * @return 0 on success, negative AVERROR code on failure |
| */ |
| int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, |
| int stride); |
| |
| /** |
| * Set channel mixing matrix. |
| * |
| * Allows for setting a custom mixing matrix, overriding the default matrix |
| * generated internally during avresample_open(). This function can be called |
| * anytime on an allocated context, either before or after calling |
| * avresample_open(), as long as the channel layouts have been set. |
| * avresample_convert() always uses the current matrix. |
| * Calling avresample_close() on the context will clear the current matrix. |
| * |
| * @see avresample_close() |
| * |
| * @param avr audio resample context |
| * @param 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 |
| * @return 0 on success, negative AVERROR code on failure |
| */ |
| int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, |
| int stride); |
| |
| /** |
| * Set a customized input channel mapping. |
| * |
| * This function can only be called when the allocated context is not open. |
| * Also, the input channel layout must have already been set. |
| * |
| * Calling avresample_close() on the context will clear the channel mapping. |
| * |
| * The map for each input channel specifies the channel index in the source to |
| * use for that particular channel, or -1 to mute the channel. Source channels |
| * can be duplicated by using the same index for multiple input channels. |
| * |
| * Examples: |
| * |
| * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs): |
| * { 1, 2, 0, 5, 3, 4 } |
| * |
| * Muting the 3rd channel in 4-channel input: |
| * { 0, 1, -1, 3 } |
| * |
| * Duplicating the left channel of stereo input: |
| * { 0, 0 } |
| * |
| * @param avr audio resample context |
| * @param channel_map customized input channel mapping |
| * @return 0 on success, negative AVERROR code on failure |
| */ |
| int avresample_set_channel_mapping(AVAudioResampleContext *avr, |
| const int *channel_map); |
| |
| /** |
| * Set compensation for resampling. |
| * |
| * This can be called anytime after avresample_open(). If resampling is not |
| * automatically enabled because of a sample rate conversion, the |
| * "force_resampling" option must have been set to 1 when opening the context |
| * in order to use resampling compensation. |
| * |
| * @param avr audio resample context |
| * @param sample_delta compensation delta, in samples |
| * @param compensation_distance compensation distance, in samples |
| * @return 0 on success, negative AVERROR code on failure |
| */ |
| int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta, |
| int compensation_distance); |
| |
| /** |
| * Provide the upper bound on the number of samples the configured |
| * conversion would output. |
| * |
| * @param avr audio resample context |
| * @param in_nb_samples number of input samples |
| * |
| * @return number of samples or AVERROR(EINVAL) if the value |
| * would exceed INT_MAX |
| */ |
| |
| int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples); |
| |
| /** |
| * Convert input samples and write them to the output FIFO. |
| * |
| * The upper bound on the number of output samples can be obtained through |
| * avresample_get_out_samples(). |
| * |
| * The output data 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 avresample_read(). |
| * |
| * If converting sample rate, there may be data remaining in the internal |
| * resampling delay buffer. avresample_get_delay() tells the number of remaining |
| * samples. To get this data as output, call avresample_convert() with NULL |
| * input. |
| * |
| * At the end of the conversion process, there may be data remaining in the |
| * internal FIFO buffer. avresample_available() tells the number of remaining |
| * samples. To get this data as output, either call avresample_convert() with |
| * NULL input or call avresample_read(). |
| * |
| * @see avresample_get_out_samples() |
| * @see avresample_read() |
| * @see avresample_get_delay() |
| * |
| * @param avr audio resample context |
| * @param output output data pointers |
| * @param out_plane_size output plane size, in bytes. |
| * This can be 0 if unknown, but that will lead to |
| * optimized functions not being used directly on the |
| * output, which could slow down some conversions. |
| * @param out_samples maximum number of samples that the output buffer can hold |
| * @param input input data pointers |
| * @param in_plane_size input plane size, in bytes |
| * This can be 0 if unknown, but that will lead to |
| * optimized functions not being used directly on the |
| * input, which could slow down some conversions. |
| * @param in_samples number of input samples to convert |
| * @return number of samples written to the output buffer, |
| * not including converted samples added to the internal |
| * output FIFO |
| */ |
| int avresample_convert(AVAudioResampleContext *avr, uint8_t **output, |
| int out_plane_size, int out_samples, |
| uint8_t * const *input, int in_plane_size, |
| int in_samples); |
| |
| /** |
| * Return the number of samples currently in the resampling delay buffer. |
| * |
| * When resampling, there may be a delay between the input and output. Any |
| * unconverted samples in each call are stored internally in a delay buffer. |
| * This function allows the user to determine the current number of samples in |
| * the delay buffer, which can be useful for synchronization. |
| * |
| * @see avresample_convert() |
| * |
| * @param avr audio resample context |
| * @return number of samples currently in the resampling delay buffer |
| */ |
| int avresample_get_delay(AVAudioResampleContext *avr); |
| |
| /** |
| * Return the number of available samples in the output FIFO. |
| * |
| * During conversion, if the user does not specify an output buffer or |
| * specifies an output buffer that is smaller than what is needed, remaining |
| * samples that are not written to the output are stored to an internal FIFO |
| * buffer. The samples in the FIFO can be read with avresample_read() or |
| * avresample_convert(). |
| * |
| * @see avresample_read() |
| * @see avresample_convert() |
| * |
| * @param avr audio resample context |
| * @return number of samples available for reading |
| */ |
| int avresample_available(AVAudioResampleContext *avr); |
| |
| /** |
| * Read samples from the output FIFO. |
| * |
| * During conversion, if the user does not specify an output buffer or |
| * specifies an output buffer that is smaller than what is needed, remaining |
| * samples that are not written to the output are stored to an internal FIFO |
| * buffer. This function can be used to read samples from that internal FIFO. |
| * |
| * @see avresample_available() |
| * @see avresample_convert() |
| * |
| * @param avr audio resample context |
| * @param output output data pointers. May be NULL, in which case |
| * nb_samples of data is discarded from output FIFO. |
| * @param nb_samples number of samples to read from the FIFO |
| * @return the number of samples written to output |
| */ |
| int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples); |
| |
| /** |
| * 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. |
| * |
| * The upper bound on the number of output samples is obtained through |
| * avresample_get_out_samples(). |
| * |
| * If the output AVFrame does not have the data pointers allocated the nb_samples |
| * field will be set using avresample_get_out_samples() and 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 avresample_convert() or to avresample_read(). |
| * |
| * If converting sample rate, there may be data remaining in the internal |
| * resampling delay buffer. avresample_get_delay() tells the number of |
| * remaining samples. To get this data as output, call this function or |
| * avresample_convert() with NULL input. |
| * |
| * At the end of the conversion process, there may be data remaining in the |
| * internal FIFO buffer. avresample_available() tells the number of remaining |
| * samples. To get this data as output, either call this function or |
| * avresample_convert() with NULL input or call avresample_read(). |
| * |
| * If the AVAudioResampleContext 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 AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned. |
| * |
| * @see avresample_get_out_samples() |
| * @see avresample_available() |
| * @see avresample_convert() |
| * @see avresample_read() |
| * @see avresample_get_delay() |
| * |
| * @param avr audio resample context |
| * @param output output AVFrame |
| * @param input input AVFrame |
| * @return 0 on success, AVERROR on failure or nonmatching |
| * configuration. |
| */ |
| int avresample_convert_frame(AVAudioResampleContext *avr, |
| AVFrame *output, AVFrame *input); |
| |
| /** |
| * Configure or reconfigure the AVAudioResampleContext using the information |
| * provided by the AVFrames. |
| * |
| * The original resampling context is reset even on failure. |
| * The function calls avresample_close() internally if the context is open. |
| * |
| * @see avresample_open(); |
| * @see avresample_close(); |
| * |
| * @param avr audio resample context |
| * @param out output AVFrame |
| * @param in input AVFrame |
| * @return 0 on success, AVERROR on failure. |
| */ |
| int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in); |
| |
| /** |
| * @} |
| */ |
| |
| #endif /* AVRESAMPLE_AVRESAMPLE_H */ |
