/*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
*/
enum AVSampleFormat av_get_sample_fmt(const char *name);
+/**
+ * Return the planar<->packed alternative form of the given sample format, or
+ * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the
+ * requested planar/packed format, the format returned is the same as the
+ * input.
+ */
+enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar);
+
/**
* Get the packed alternative form of the given sample format.
*
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 */