From 66fe58064295f838f894758d92b087100ce022aa Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Wed, 26 Mar 2008 10:37:17 +0000 Subject: explicitly downcast Tools like "sparse" check for missing downcasts, since implicit cast may be dangerous. Although that does not change the compiler result, it may make the code more readable (IMHO), because you always see when there may be data cut off. git-svn-id: https://svn.musicpd.org/mpd/trunk@7196 09075e82-0dd4-0310-85a5-a0d7c8717e4f --- src/inputPlugins/audiofile_plugin.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'src/inputPlugins/audiofile_plugin.c') 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", -- cgit v1.2.3