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Diffstat (limited to 'src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h')
| -rw-r--r-- | src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h | 220 |
1 files changed, 0 insertions, 220 deletions
diff --git a/src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h b/src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h deleted file mode 100644 index 33cbdedf5..000000000 --- a/src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h +++ /dev/null @@ -1,220 +0,0 @@ -/* - * This file is part of Libav. - * - * Libav 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. - * - * Libav 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 Libav; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#ifndef AVUTIL_SAMPLEFMT_H -#define AVUTIL_SAMPLEFMT_H - -#include <stdint.h> - -#include "avutil.h" -#include "attributes.h" - -/** - * Audio Sample Formats - * - * @par - * The data described by the sample format is always in native-endian order. - * Sample values can be expressed by native C types, hence the lack of a signed - * 24-bit sample format even though it is a common raw audio data format. - * - * @par - * The floating-point formats are based on full volume being in the range - * [-1.0, 1.0]. Any values outside this range are beyond full volume level. - * - * @par - * The data layout as used in av_samples_fill_arrays() and elsewhere in Libav - * (such as AVFrame in libavcodec) is as follows: - * - * For planar sample formats, each audio channel is in a separate data plane, - * and linesize is the buffer size, in bytes, for a single plane. All data - * planes must be the same size. For packed sample formats, only the first data - * plane is used, and samples for each channel are interleaved. In this case, - * linesize is the buffer size, in bytes, for the 1 plane. - */ -enum AVSampleFormat { - AV_SAMPLE_FMT_NONE = -1, - AV_SAMPLE_FMT_U8, ///< unsigned 8 bits - AV_SAMPLE_FMT_S16, ///< signed 16 bits - AV_SAMPLE_FMT_S32, ///< signed 32 bits - AV_SAMPLE_FMT_FLT, ///< float - AV_SAMPLE_FMT_DBL, ///< double - - AV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planar - AV_SAMPLE_FMT_S16P, ///< signed 16 bits, planar - AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar - AV_SAMPLE_FMT_FLTP, ///< float, planar - AV_SAMPLE_FMT_DBLP, ///< double, planar - - AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically -}; - -/** - * Return the name of sample_fmt, or NULL if sample_fmt is not - * recognized. - */ -const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt); - -/** - * Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE - * on error. - */ -enum AVSampleFormat av_get_sample_fmt(const char *name); - -/** - * Get the packed alternative form of the given sample format. - * - * If the passed sample_fmt is already in packed format, the format returned is - * the same as the input. - * - * @return the packed alternative form of the given sample format or - AV_SAMPLE_FMT_NONE on error. - */ -enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt); - -/** - * Get the planar alternative form of the given sample format. - * - * If the passed sample_fmt is already in planar format, the format returned is - * the same as the input. - * - * @return the planar alternative form of the given sample format or - AV_SAMPLE_FMT_NONE on error. - */ -enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt); - -/** - * Generate a string corresponding to the sample format with - * sample_fmt, or a header if sample_fmt is negative. - * - * @param buf the buffer where to write the string - * @param buf_size the size of buf - * @param sample_fmt the number of the sample format to print the - * corresponding info string, or a negative value to print the - * corresponding header. - * @return the pointer to the filled buffer or NULL if sample_fmt is - * unknown or in case of other errors - */ -char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt); - -/** - * Return number of bytes per sample. - * - * @param sample_fmt the sample format - * @return number of bytes per sample or zero if unknown for the given - * sample format - */ -int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt); - -/** - * Check if the sample format is planar. - * - * @param sample_fmt the sample format to inspect - * @return 1 if the sample format is planar, 0 if it is interleaved - */ -int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt); - -/** - * Get the required buffer size for the given audio parameters. - * - * @param[out] linesize calculated linesize, may be NULL - * @param nb_channels the number of channels - * @param nb_samples the number of samples in a single channel - * @param sample_fmt the sample format - * @param align buffer size alignment (0 = default, 1 = no alignment) - * @return required buffer size, or negative error code on failure - */ -int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, - enum AVSampleFormat sample_fmt, int align); - -/** - * Fill channel data pointers and linesize for samples with sample - * format sample_fmt. - * - * The pointers array is filled with the pointers to the samples data: - * for planar, set the start point of each channel's data within the buffer, - * for packed, set the start point of the entire buffer only. - * - * The linesize array is filled with the aligned size of each channel's data - * buffer for planar layout, or the aligned size of the buffer for all channels - * for packed layout. - * - * @see enum AVSampleFormat - * The documentation for AVSampleFormat describes the data layout. - * - * @param[out] audio_data array to be filled with the pointer for each channel - * @param[out] linesize calculated linesize, may be NULL - * @param buf the pointer to a buffer containing the samples - * @param nb_channels the number of channels - * @param nb_samples the number of samples in a single channel - * @param sample_fmt the sample format - * @param align buffer size alignment (0 = default, 1 = no alignment) - * @return 0 on success or a negative error code on failure - */ -int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, - const uint8_t *buf, - int nb_channels, int nb_samples, - enum AVSampleFormat sample_fmt, int align); - -/** - * Allocate a samples buffer for nb_samples samples, and fill data pointers and - * linesize accordingly. - * The allocated samples buffer can be freed by using av_freep(&audio_data[0]) - * Allocated data will be initialized to silence. - * - * @see enum AVSampleFormat - * The documentation for AVSampleFormat describes the data layout. - * - * @param[out] audio_data array to be filled with the pointer for each channel - * @param[out] linesize aligned size for audio buffer(s), may be NULL - * @param nb_channels number of audio channels - * @param nb_samples number of samples per channel - * @param align buffer size alignment (0 = default, 1 = no alignment) - * @return 0 on success or a negative error code on failure - * @see av_samples_fill_arrays() - */ -int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, - int nb_samples, enum AVSampleFormat sample_fmt, int align); - -/** - * Copy samples from src to dst. - * - * @param dst destination array of pointers to data planes - * @param src source array of pointers to data planes - * @param dst_offset offset in samples at which the data will be written to dst - * @param src_offset offset in samples at which the data will be read from src - * @param nb_samples number of samples to be copied - * @param nb_channels number of audio channels - * @param sample_fmt audio sample format - */ -int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset, - int src_offset, int nb_samples, int nb_channels, - enum AVSampleFormat sample_fmt); - -/** - * Fill an audio buffer with silence. - * - * @param audio_data array of pointers to data planes - * @param offset offset in samples at which to start filling - * @param nb_samples number of samples to fill - * @param nb_channels number of audio channels - * @param sample_fmt audio sample format - */ -int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, - int nb_channels, enum AVSampleFormat sample_fmt); - -#endif /* AVUTIL_SAMPLEFMT_H */ |