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| author | Anton Khirnov <anton@khirnov.net> | 2012-05-06 14:10:38 +0200 |
|---|---|---|
| committer | Anton Khirnov <anton@khirnov.net> | 2012-05-09 17:43:26 +0200 |
| commit | 142e740d1ecc6059556f2748a18757d399ee061f (patch) | |
| tree | 556cb9a5adaf71e1dec853849c54615ecfff9328 | |
| parent | 9684341346fd5aad436325529cade47966c4731b (diff) | |
| download | ffmpeg-142e740d1ecc6059556f2748a18757d399ee061f.tar.gz | |
samplefmt: add a function for copying audio samples.
| -rw-r--r-- | libavutil/samplefmt.c | 19 | ||||
| -rw-r--r-- | libavutil/samplefmt.h | 15 |
2 files changed, 34 insertions, 0 deletions
diff --git a/libavutil/samplefmt.c b/libavutil/samplefmt.c index 711afac287..4d94fa69be 100644 --- a/libavutil/samplefmt.c +++ b/libavutil/samplefmt.c @@ -185,3 +185,22 @@ int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, } return 0; } + +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) +{ + int planar = av_sample_fmt_is_planar(sample_fmt); + int planes = planar ? nb_channels : 1; + int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels); + int data_size = nb_samples * block_align; + int i; + + dst_offset *= block_align; + src_offset *= block_align; + + for (i = 0; i < planes; i++) + memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size); + + return 0; +} diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h index 1cb01a357f..9011889e68 100644 --- a/libavutil/samplefmt.h +++ b/libavutil/samplefmt.h @@ -194,4 +194,19 @@ int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, 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); + #endif /* AVUTIL_SAMPLEFMT_H */ |