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author | Stefano Sabatini <stefasab@gmail.com> | 2012-06-30 00:02:58 +0200 |
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committer | Stefano Sabatini <stefasab@gmail.com> | 2012-06-30 15:01:51 +0200 |
commit | b12e61ac7f3f1fdf47e5a5083242d2ba4096a1da (patch) | |
tree | 7c9f86a6c0395e3ea80f1b125ceb8641c7f8f61f | |
parent | 6f5e7e8690c9bfd0f4cf9581319adfb7c0983a9f (diff) | |
download | ffmpeg-b12e61ac7f3f1fdf47e5a5083242d2ba4096a1da.tar.gz |
ffplay: factorize expression in audio_decode_frame()
Possibly improve readability.
-rw-r--r-- | ffplay.c | 6 |
1 files changed, 3 insertions, 3 deletions
@@ -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); } |