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author | Reimar Döffinger <Reimar.Doeffinger@gmx.de> | 2010-07-24 13:59:49 +0000 |
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committer | Reimar Döffinger <Reimar.Doeffinger@gmx.de> | 2010-07-24 13:59:49 +0000 |
commit | edac49daf5f703aa4e742ecdd747658e82d91b33 (patch) | |
tree | 25e3407964f6b8f65c1ed2a20e55c72bb7266870 /libavcodec/alacenc.c | |
parent | 6f2c523c85c53cce2bda22a585b07a807255eadd (diff) | |
download | ffmpeg-edac49daf5f703aa4e742ecdd747658e82d91b33.tar.gz |
Use "const" qualifier for pointers that point to input data of
audio encoders.
This is purely for clarity/documentation purposes.
Originally committed as revision 24481 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/alacenc.c')
-rw-r--r-- | libavcodec/alacenc.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index ee6acc0715..0fad99febd 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -75,12 +75,12 @@ typedef struct AlacEncodeContext { } AlacEncodeContext; -static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples) +static void init_sample_buffers(AlacEncodeContext *s, const int16_t *input_samples) { int ch, i; for(ch=0;ch<s->avctx->channels;ch++) { - int16_t *sptr = input_samples + ch; + const int16_t *sptr = input_samples + ch; for(i=0;i<s->avctx->frame_size;i++) { s->sample_buf[ch][i] = *sptr; sptr += s->avctx->channels; @@ -482,7 +482,7 @@ verbatim: if((s->compression_level == 0) || verbatim_flag) { // Verbatim mode - int16_t *samples = data; + const int16_t *samples = data; write_frame_header(s, 1); for(i=0; i<avctx->frame_size*avctx->channels; i++) { put_sbits(pb, 16, *samples++); |