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-rw-r--r--src/inputPlugins/_flac_common.c6
-rw-r--r--src/inputPlugins/audiofile_plugin.c8
2 files changed, 7 insertions, 7 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 0a2adf6b7..3b351d3a7 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -66,7 +66,7 @@ static int flacFindVorbisCommentFloat(const FLAC__StreamMetadata * block,
comments[offset].entry[pos]);
tmp = p[len];
p[len] = '\0';
- *fl = atof((char *)p);
+ *fl = (float)atof((char *)p);
p[len] = tmp;
return 1;
@@ -170,9 +170,9 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- dc->audioFormat.bits = si->bits_per_sample;
+ dc->audioFormat.bits = (mpd_sint8)si->bits_per_sample;
dc->audioFormat.sampleRate = si->sample_rate;
- dc->audioFormat.channels = si->channels;
+ dc->audioFormat.channels = (mpd_sint8)si->channels;
dc->totalTime = ((float)si->total_samples) / (si->sample_rate);
getOutputAudioFormat(&(dc->audioFormat),
&(data->cb->audioFormat));
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",