/*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * 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 "avutil.h"
+
/**
- * all in native-endian format
+ * 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_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
};
*/
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.
*/
char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
+#if FF_API_GET_BITS_PER_SAMPLE_FMT
+/**
+ * @deprecated Use av_get_bytes_per_sample() instead.
+ */
+attribute_deprecated
+int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt);
+#endif
+
/**
- * Return sample format bits per sample.
+ * Return number of bytes per sample.
*
* @param sample_fmt the sample format
- * @return number of bits per sample or zero if unknown for the given
+ * @return number of bytes per sample or zero if unknown for the given
* sample format
*/
-int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt);
+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])
+ *
+ * @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 */
projects / ffmpeg / blobdiff