[FFmpeg-cvslog] samplefmt: add a function for copying audio samples.

[Anton Khirnov]

ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Sun May  6 14:10:38 2012 +0200| [142e740d1ecc6059556f2748a18757d399ee061f] | committer: Anton Khirnov

samplefmt: add a function for copying audio samples.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=142e740d1ecc6059556f2748a18757d399ee061f
---

 libavutil/samplefmt.c |   19 +++++++++++++++++++
 libavutil/samplefmt.h |   15 +++++++++++++++
 2 files changed, 34 insertions(+), 0 deletions(-)

diff --git a/libavutil/samplefmt.c b/libavutil/samplefmt.c
index 711afac..4d94fa6 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 1cb01a3..9011889 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 */