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authorLukasz Marek <lukasz.m.luki2@gmail.com>2014-05-11 05:40:25 +0200
committerLukasz Marek <lukasz.m.luki2@gmail.com>2014-05-20 00:00:45 +0200
commit37fa4b9bb9672520fa901b69234ac611bbbd133f (patch)
tree9cf2af45e7d054fed807dee1c70b4a0bee076dfb /libavformat/audiointerleave.c
parent2ed9e17ed1793b3b66ed27c0a113676a46eb9871 (diff)
downloadffmpeg-37fa4b9bb9672520fa901b69234ac611bbbd133f.tar.gz
lavf/audiointerleave: return more meaningful error codes
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
Diffstat (limited to 'libavformat/audiointerleave.c')
-rw-r--r--libavformat/audiointerleave.c8
1 files changed, 4 insertions, 4 deletions
diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index 97bd801aa7..7f580b970d 100644
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -45,11 +45,11 @@ int ff_audio_interleave_init(AVFormatContext *s,
int i;
if (!samples_per_frame)
- return -1;
+ return AVERROR(EINVAL);
if (!time_base.num) {
av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
- return -1;
+ return AVERROR(EINVAL);
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
@@ -60,7 +60,7 @@ int ff_audio_interleave_init(AVFormatContext *s,
av_get_bits_per_sample(st->codec->codec_id)) / 8;
if (!aic->sample_size) {
av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
- return -1;
+ return AVERROR(EINVAL);
}
aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
@@ -114,7 +114,7 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
if (new_size > aic->fifo_size) {
if (av_fifo_realloc2(aic->fifo, new_size) < 0)
- return -1;
+ return AVERROR(ENOMEM);
aic->fifo_size = new_size;
}
av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);