Commit b12e61ac authored by Stefano Sabatini's avatar Stefano Sabatini

ffplay: factorize expression in audio_decode_frame()

Possibly improve readability.
parent 6f5e7e86
...@@ -1985,6 +1985,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) ...@@ -1985,6 +1985,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
if (is->swr_ctx) { if (is->swr_ctx) {
const uint8_t *in[] = { is->frame->data[0] }; const uint8_t *in[] = { is->frame->data[0] };
uint8_t *out[] = {is->audio_buf2}; 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 (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, 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) { 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) ...@@ -1992,13 +1993,12 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
break; 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), len2 = swr_convert(is->swr_ctx, out, out_count, in, is->frame->nb_samples);
in, is->frame->nb_samples);
if (len2 < 0) { if (len2 < 0) {
fprintf(stderr, "swr_convert() failed\n"); fprintf(stderr, "swr_convert() failed\n");
break; 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"); fprintf(stderr, "warning: audio buffer is probably too small\n");
swr_init(is->swr_ctx); swr_init(is->swr_ctx);
} }
......
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