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270 lines
10 KiB
C
270 lines
10 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVUTIL_SAMPLEFMT_H
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#define AVUTIL_SAMPLEFMT_H
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#include <stdint.h>
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/**
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* @addtogroup lavu_audio
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* @{
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*
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* @defgroup lavu_sampfmts Audio sample formats
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*
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* Audio sample format enumeration and related convenience functions.
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* @{
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*/
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/**
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* Audio sample formats
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*
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* - The data described by the sample format is always in native-endian order.
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* Sample values can be expressed by native C types, hence the lack of a signed
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* 24-bit sample format even though it is a common raw audio data format.
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*
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* - The floating-point formats are based on full volume being in the range
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* [-1.0, 1.0]. Any values outside this range are beyond full volume level.
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*
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* - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
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* (such as AVFrame in libavcodec) is as follows:
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*
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* @par
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* For planar sample formats, each audio channel is in a separate data plane,
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* and linesize is the buffer size, in bytes, for a single plane. All data
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* planes must be the same size. For packed sample formats, only the first data
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* plane is used, and samples for each channel are interleaved. In this case,
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* linesize is the buffer size, in bytes, for the 1 plane.
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*
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*/
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enum AVSampleFormat {
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AV_SAMPLE_FMT_NONE = -1,
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AV_SAMPLE_FMT_U8, ///< unsigned 8 bits
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AV_SAMPLE_FMT_S16, ///< signed 16 bits
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AV_SAMPLE_FMT_S32, ///< signed 32 bits
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AV_SAMPLE_FMT_FLT, ///< float
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AV_SAMPLE_FMT_DBL, ///< double
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AV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planar
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AV_SAMPLE_FMT_S16P, ///< signed 16 bits, planar
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AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar
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AV_SAMPLE_FMT_FLTP, ///< float, planar
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AV_SAMPLE_FMT_DBLP, ///< double, planar
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AV_SAMPLE_FMT_S64, ///< signed 64 bits
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AV_SAMPLE_FMT_S64P, ///< signed 64 bits, planar
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AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically
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};
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/**
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* Return the name of sample_fmt, or NULL if sample_fmt is not
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* recognized.
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*/
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const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
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/**
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* Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE
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* on error.
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*/
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enum AVSampleFormat av_get_sample_fmt(const char *name);
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/**
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* Return the planar<->packed alternative form of the given sample format, or
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* AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the
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* requested planar/packed format, the format returned is the same as the
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* input.
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*/
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enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar);
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/**
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* Get the packed alternative form of the given sample format.
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*
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* If the passed sample_fmt is already in packed format, the format returned is
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* the same as the input.
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*
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* @return the packed alternative form of the given sample format or
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AV_SAMPLE_FMT_NONE on error.
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*/
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enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt);
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/**
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* Get the planar alternative form of the given sample format.
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*
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* If the passed sample_fmt is already in planar format, the format returned is
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* the same as the input.
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*
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* @return the planar alternative form of the given sample format or
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AV_SAMPLE_FMT_NONE on error.
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*/
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enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
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/**
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* Generate a string corresponding to the sample format with
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* sample_fmt, or a header if sample_fmt is negative.
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*
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* @param buf the buffer where to write the string
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* @param buf_size the size of buf
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* @param sample_fmt the number of the sample format to print the
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* corresponding info string, or a negative value to print the
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* corresponding header.
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* @return the pointer to the filled buffer or NULL if sample_fmt is
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* unknown or in case of other errors
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*/
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char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
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/**
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* Return number of bytes per sample.
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*
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* @param sample_fmt the sample format
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* @return number of bytes per sample or zero if unknown for the given
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* sample format
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*/
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int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
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/**
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* Check if the sample format is planar.
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*
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* @param sample_fmt the sample format to inspect
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* @return 1 if the sample format is planar, 0 if it is interleaved
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*/
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int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
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/**
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* Get the required buffer size for the given audio parameters.
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*
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* @param[out] linesize calculated linesize, may be NULL
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* @param nb_channels the number of channels
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* @param nb_samples the number of samples in a single channel
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* @param sample_fmt the sample format
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* @param align buffer size alignment (0 = default, 1 = no alignment)
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* @return required buffer size, or negative error code on failure
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*/
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int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
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enum AVSampleFormat sample_fmt, int align);
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/**
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* @}
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*
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* @defgroup lavu_sampmanip Samples manipulation
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*
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* Functions that manipulate audio samples
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* @{
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*/
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/**
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* Fill plane data pointers and linesize for samples with sample
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* format sample_fmt.
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*
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* The audio_data array is filled with the pointers to the samples data planes:
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* for planar, set the start point of each channel's data within the buffer,
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* for packed, set the start point of the entire buffer only.
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*
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* The value pointed to by linesize is set to the aligned size of each
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* channel's data buffer for planar layout, or to the aligned size of the
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* buffer for all channels for packed layout.
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*
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* The buffer in buf must be big enough to contain all the samples
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* (use av_samples_get_buffer_size() to compute its minimum size),
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* otherwise the audio_data pointers will point to invalid data.
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*
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* @see enum AVSampleFormat
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* The documentation for AVSampleFormat describes the data layout.
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*
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* @param[out] audio_data array to be filled with the pointer for each channel
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* @param[out] linesize calculated linesize, may be NULL
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* @param buf the pointer to a buffer containing the samples
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* @param nb_channels the number of channels
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* @param nb_samples the number of samples in a single channel
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* @param sample_fmt the sample format
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* @param align buffer size alignment (0 = default, 1 = no alignment)
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* @return minimum size in bytes required for the buffer on success,
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* or a negative error code on failure
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*/
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int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
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const uint8_t *buf,
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int nb_channels, int nb_samples,
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enum AVSampleFormat sample_fmt, int align);
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/**
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* Allocate a samples buffer for nb_samples samples, and fill data pointers and
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* linesize accordingly.
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* The allocated samples buffer can be freed by using av_freep(&audio_data[0])
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* Allocated data will be initialized to silence.
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*
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* @see enum AVSampleFormat
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* The documentation for AVSampleFormat describes the data layout.
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*
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* @param[out] audio_data array to be filled with the pointer for each channel
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* @param[out] linesize aligned size for audio buffer(s), may be NULL
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* @param nb_channels number of audio channels
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* @param nb_samples number of samples per channel
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* @param sample_fmt the sample format
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* @param align buffer size alignment (0 = default, 1 = no alignment)
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* @return >=0 on success or a negative error code on failure
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* @todo return the size of the allocated buffer in case of success at the next bump
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* @see av_samples_fill_arrays()
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* @see av_samples_alloc_array_and_samples()
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*/
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int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
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int nb_samples, enum AVSampleFormat sample_fmt, int align);
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/**
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* Allocate a data pointers array, samples buffer for nb_samples
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* samples, and fill data pointers and linesize accordingly.
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*
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* This is the same as av_samples_alloc(), but also allocates the data
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* pointers array.
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*
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* @see av_samples_alloc()
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*/
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int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
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int nb_samples, enum AVSampleFormat sample_fmt, int align);
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/**
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* Copy samples from src to dst.
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*
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* @param dst destination array of pointers to data planes
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* @param src source array of pointers to data planes
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* @param dst_offset offset in samples at which the data will be written to dst
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* @param src_offset offset in samples at which the data will be read from src
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* @param nb_samples number of samples to be copied
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* @param nb_channels number of audio channels
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* @param sample_fmt audio sample format
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*/
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int av_samples_copy(uint8_t * const *dst, uint8_t * const *src, int dst_offset,
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int src_offset, int nb_samples, int nb_channels,
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enum AVSampleFormat sample_fmt);
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/**
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* Fill an audio buffer with silence.
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*
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* @param audio_data array of pointers to data planes
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* @param offset offset in samples at which to start filling
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* @param nb_samples number of samples to fill
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* @param nb_channels number of audio channels
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* @param sample_fmt audio sample format
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*/
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int av_samples_set_silence(uint8_t * const *audio_data, int offset, int nb_samples,
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int nb_channels, enum AVSampleFormat sample_fmt);
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/**
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* @}
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* @}
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*/
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#endif /* AVUTIL_SAMPLEFMT_H */
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