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| author | Eric Wong <normalperson@yhbt.net> | 2006-08-24 20:54:40 +0000 |
|---|---|---|
| committer | Eric Wong <normalperson@yhbt.net> | 2006-08-24 20:54:40 +0000 |
| commit | b8fe818ae744e3656c7f235b23e6bfccf7494d59 (patch) | |
| tree | a565ab31d9cdfcfabc49fb7690f18563b7940ff0 /src | |
| parent | c81f4e2c048e8e9053a2ea1573a9fc3a70c3a9b8 (diff) | |
| download | mpd-b8fe818ae744e3656c7f235b23e6bfccf7494d59.tar.gz mpd-b8fe818ae744e3656c7f235b23e6bfccf7494d59.tar.xz mpd-b8fe818ae744e3656c7f235b23e6bfccf7494d59.zip | |
audiofile_plugin: fix for playing mono .au files with 8000Hz sample rate
Use the 'Virtual' variants of afGetSampleFormat, afGetChannels,
afGetVirtualFrameSize in the audiofile library, since it already does
the necessary abstraction for us.
Of course, I've regression tested these changes against my
standard 44100Hz/2ch/16bit wave files and they continue to play
fine.
Files tested:
english.au (Linus Torvalds pronouncing 'Linux' in English)
B01.Red_Bright_Heart.au (Chinese opera, sounds correct to me even though
I don't actually understand the words)
git-svn-id: https://svn.musicpd.org/mpd/trunk@4681 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src')
| -rw-r--r-- | src/inputPlugins/audiofile_plugin.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c index adec604ae..31c835b8f 100644 --- a/src/inputPlugins/audiofile_plugin.c +++ b/src/inputPlugins/audiofile_plugin.c @@ -70,10 +70,10 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path) return -1; } - afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); + afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); dc->audioFormat.bits = bits; dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK); - dc->audioFormat.channels = afGetChannels(af_fp, AF_DEFAULT_TRACK); + dc->audioFormat.channels = afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat)); frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); @@ -90,7 +90,7 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path) return -1; } - fs = (int)afGetFrameSize(af_fp, AF_DEFAULT_TRACK, 1); + fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); dc->state = DECODE_STATE_DECODE; { |