diff options
Diffstat (limited to 'libavutil/samplefmt.h')
-rw-r--r-- | libavutil/samplefmt.h | 70 |
1 files changed, 48 insertions, 22 deletions
diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h index 8347fac..6a8a031 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 */ @@ -36,20 +36,17 @@ */ /** - * Audio Sample Formats + * 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. + * - 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. + * - 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: + * - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg + * (such as AVFrame in libavcodec) is as follows: * * @par * For planar sample formats, each audio channel is in a separate data plane, @@ -57,6 +54,7 @@ * 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, @@ -88,6 +86,14 @@ 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 @@ -163,16 +169,20 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, */ /** - * 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. @@ -184,7 +194,9 @@ 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, @@ -205,13 +217,27 @@ 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 |