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audiofile_plugin: use afSetVirtualSampleFormat, too

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This finally fixes a bug from over two years ago playing a wave file
(oprah.wav) with the following characteristics (from sfinfo):

File Format    Microsoft RIFF WAVE Format (wave)
Data Format    8-bit integer (unsigned, little endian)
Audio Data     986827 bytes begins at offset 58 (3a hex)
1 channel, 986827 frames
Sampling Rate  22050.00 Hz
Duration       44.754 seconds

Of course, this has been regression tested with all the files
that the previous commit got working.  Thanks to Michael Pruett
(audiofile author) for the hint and shame on me for forgetting
about it for over two years :x

git-svn-id: https://svn.musicpd.org/mpd/trunk@4682 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Eric Wong 2006-08-24 21:59:19 +00:00
parent b8fe818ae7
commit 64a4c635de
1 changed files with 2 additions and 0 deletions
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2
src/inputPlugins/audiofile_plugin.c
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@ -70,6 +70,8 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
return -1;
}
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);