diff options
Diffstat (limited to 'src/inputPlugins/audiofile_plugin.c')
| -rw-r--r-- | src/inputPlugins/audiofile_plugin.c | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c index 6fcc98239..858b71229 100644 --- a/src/inputPlugins/audiofile_plugin.c +++ b/src/inputPlugins/audiofile_plugin.c @@ -45,7 +45,7 @@ static int audiofile_decode(char *path) int fs, frame_count; AFfilehandle af_fp; int bits; - mpd_uint16 bitRate; + uint16_t bitRate; struct stat st; int ret, current = 0; char chunk[CHUNK_SIZE]; @@ -64,18 +64,18 @@ static int audiofile_decode(char *path) afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16); afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); - dc.audio_format.bits = (mpd_uint8)bits; + dc.audio_format.bits = (uint8_t)bits; dc.audio_format.sampleRate = (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); dc.audio_format.channels = - (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); + (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); dc.total_time = ((float)frame_count / (float)dc.audio_format.sampleRate); - bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.total_time / 1000.0 + 0.5); + bitRate = (uint16_t)(st.st_size * 8.0 / dc.total_time / 1000.0 + 0.5); if (dc.audio_format.bits != 8 && dc.audio_format.bits != 16) { ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n", |