diff options
Diffstat (limited to 'src/inputPlugins')
| -rw-r--r-- | src/inputPlugins/_flac_common.c | 6 | ||||
| -rw-r--r-- | src/inputPlugins/audiofile_plugin.c | 8 |
2 files changed, 7 insertions, 7 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c index 0a2adf6b..3b351d3a 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 1213d31e..3ca9a14c 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", |