audiofile_plugin: use afSetVirtualSampleFormat, too

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

audiofile_plugin: use afSetVirtualSampleFormat, too
64a4c635 · Eric Wong · b8fe818a · 64a4c635
Commit 64a4c635 authored Aug 24, 2006 by Eric Wong
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audiofile_plugin.c src/inputPlugins/audiofile_plugin.c +2 -0

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--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -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);