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-rw-r--r--src/inputPlugins/audiofile_plugin.c8
1 files changed, 4 insertions, 4 deletions
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 1213d31e5..3ca9a14c3 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -67,9 +67,9 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- dc->audioFormat.bits = bits;
- dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
- dc->audioFormat.channels = afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+ dc->audioFormat.bits = (mpd_uint8)bits;
+ dc->audioFormat.sampleRate = (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
+ dc->audioFormat.channels = (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
@@ -77,7 +77,7 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
dc->totalTime =
((float)frame_count / (float)dc->audioFormat.sampleRate);
- bitRate = st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5;
+ bitRate = (mpd_uint16)(st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5);
if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) {
ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",