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authorStefano Sabatini <stefasab@gmail.com>2012-06-30 00:02:58 +0200
committerStefano Sabatini <stefasab@gmail.com>2012-06-30 15:01:51 +0200
commitb12e61ac7f3f1fdf47e5a5083242d2ba4096a1da (patch)
tree7c9f86a6c0395e3ea80f1b125ceb8641c7f8f61f
parent6f5e7e8690c9bfd0f4cf9581319adfb7c0983a9f (diff)
downloadffmpeg-b12e61ac7f3f1fdf47e5a5083242d2ba4096a1da.tar.gz
ffplay: factorize expression in audio_decode_frame()
Possibly improve readability.
-rw-r--r--ffplay.c6
1 files changed, 3 insertions, 3 deletions
diff --git a/ffplay.c b/ffplay.c
index df66a2ecc3..c01740a682 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -1985,6 +1985,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
if (is->swr_ctx) {
const uint8_t *in[] = { is->frame->data[0] };
uint8_t *out[] = {is->audio_buf2};
+ int out_count = sizeof(is->audio_buf2) / is->audio_tgt.channels / av_get_bytes_per_sample(is->audio_tgt.fmt);
if (wanted_nb_samples != is->frame->nb_samples) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt.freq / dec->sample_rate,
wanted_nb_samples * is->audio_tgt.freq / dec->sample_rate) < 0) {
@@ -1992,13 +1993,12 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
break;
}
}
- len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt.channels / av_get_bytes_per_sample(is->audio_tgt.fmt),
- in, is->frame->nb_samples);
+ len2 = swr_convert(is->swr_ctx, out, out_count, in, is->frame->nb_samples);
if (len2 < 0) {
fprintf(stderr, "swr_convert() failed\n");
break;
}
- if (len2 == sizeof(is->audio_buf2) / is->audio_tgt.channels / av_get_bytes_per_sample(is->audio_tgt.fmt)) {
+ if (len2 == out_count) {
fprintf(stderr, "warning: audio buffer is probably too small\n");
swr_init(is->swr_ctx);
}