diff options
Diffstat (limited to 'libavutil/samplefmt.h')
-rw-r--r-- | libavutil/samplefmt.h | 60 |
1 files changed, 48 insertions, 12 deletions
diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h index 33cbdedf5f..db17d43bcf 100644 --- a/libavutil/samplefmt.h +++ b/libavutil/samplefmt.h @@ -1,18 +1,18 @@ /* - * 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 */ @@ -37,7 +37,7 @@ * [-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 + * The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg * (such as AVFrame in libavcodec) is as follows: * * For planar sample formats, each audio channel is in a separate data plane, @@ -76,6 +76,14 @@ const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt); 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. * * If the passed sample_fmt is already in packed format, the format returned is @@ -111,6 +119,14 @@ enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt); */ 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 number of bytes per sample. * @@ -142,16 +158,20 @@ 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 + * Fill plane data pointers and linesize for samples with sample * format sample_fmt. * - * The pointers array is filled with the pointers to the samples data: + * The audio_data array is filled with the pointers to the samples data planes: * 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. + * The value pointed to by linesize is set to the aligned size of each + * channel's data buffer for planar layout, or to the aligned size of the + * buffer for all channels for packed layout. + * + * The buffer in buf must be big enough to contain all the samples + * (use av_samples_get_buffer_size() to compute its minimum size), + * otherwise the audio_data pointers will point to invalid data. * * @see enum AVSampleFormat * The documentation for AVSampleFormat describes the data layout. @@ -163,7 +183,9 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, * @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 + * @return >=0 on success or a negative error code on failure + * @todo return minimum size in bytes required for the buffer in case + * of success at the next bump */ int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, const uint8_t *buf, @@ -184,13 +206,27 @@ int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, * @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 + * @return >=0 on success or a negative error code on failure + * @todo return the size of the allocated buffer in case of success at the next bump * @see av_samples_fill_arrays() + * @see av_samples_alloc_array_and_samples() */ int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align); /** + * Allocate a data pointers array, samples buffer for nb_samples + * samples, and fill data pointers and linesize accordingly. + * + * This is the same as av_samples_alloc(), but also allocates the data + * pointers array. + * + * @see av_samples_alloc() + */ +int av_samples_alloc_array_and_samples(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 |