ffplay: dont leave swresampler in half initialized state
authorMarton Balint <cus@passwd.hu>
Wed, 8 Oct 2014 21:36:11 +0000 (23:36 +0200)
committerMarton Balint <cus@passwd.hu>
Thu, 9 Oct 2014 21:18:37 +0000 (23:18 +0200)
On init failure, let's just free it, so next time it will be recreated from
start.

Also fixes Coverity CID 1241515.

Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
ffplay.c

index 8fa5ca3..37983e9 100644 (file)
--- a/ffplay.c
+++ b/ffplay.c
@@ -2419,6 +2419,7 @@ static int audio_decode_frame(VideoState *is)
                            "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
                             is->frame->sample_rate, av_get_sample_fmt_name(is->frame->format), av_frame_get_channels(is->frame),
                             is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
+                    swr_free(&is->swr_ctx);
                     break;
                 }
                 is->audio_src.channel_layout = dec_channel_layout;
@@ -2454,7 +2455,8 @@ static int audio_decode_frame(VideoState *is)
                 }
                 if (len2 == out_count) {
                     av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
-                    swr_init(is->swr_ctx);
+                    if (swr_init(is->swr_ctx) < 0)
+                        swr_free(&is->swr_ctx);
                 }
                 is->audio_buf = is->audio_buf1;
                 resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);