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author | Andreas Claesson <andreas.claesson@gmail.com> | 2005-05-24 15:06:23 +0000 |
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committer | Andreas Claesson <andreas.claesson@gmail.com> | 2005-05-24 15:06:23 +0000 |
commit | 29cc42bf9781f2407cc7ccfe801329b07434e50b (patch) | |
tree | 7a7b61381e788880c03b99325cb7448d5b7f88cd /src/audio.c | |
parent | 422410c46de81cec8872a9b91d07a7c9eca96b82 (diff) | |
download | mpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.tar.gz mpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.tar.xz mpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.zip |
Adding modifications for internal 1-3-28 bits integer samples. Only mp3 and mpc of the input plugins are modified. Resampling is NOT working!!
git-svn-id: https://svn.musicpd.org/mpd/branches/ancl@3281 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r-- | src/audio.c | 16 |
1 files changed, 5 insertions, 11 deletions
diff --git a/src/audio.c b/src/audio.c index 0fb2b3e2d..1bcd24de6 100644 --- a/src/audio.c +++ b/src/audio.c @@ -111,15 +111,9 @@ void getInternalAudioFormat(AudioFormat * inAudioFormat, /* take the input format... */ copyAudioFormat(outAudioFormat,inAudioFormat); -#ifdef MPD_FIXED_POINT - /* .. change to 16 bit integer */ - outAudioFormat->bits = 16; - outAudioFormat->floatSamples = 0; -#else - /* .. change to 32 bit float */ + /* .. change to 32 bit integers with 28 bit resolution */ outAudioFormat->bits = 32; - outAudioFormat->floatSamples = 1; -#endif + outAudioFormat->fracBits = 28; /* if forced output sample rate - use that as internal sample rate */ if(audio_configFormat && (audio_configFormat->sampleRate != 0)) outAudioFormat->sampleRate = audio_configFormat->sampleRate; @@ -132,7 +126,7 @@ void getOutputAudioFormat(AudioFormat * inAudioFormat, * for plugins who still use this we copy the format from in to out * these plugin don't know about the floatSample flag so we set it * for them - in both in and out */ - inAudioFormat->floatSamples = 0; + inAudioFormat->fracBits = 0; copyAudioFormat(outAudioFormat,inAudioFormat); } @@ -214,8 +208,8 @@ int parseAudioConfig(AudioFormat * audioFormat, char * conf) { return -1; } - /* audioFormat is never float */ - audioFormat->floatSamples = 0; + /* audioFormat is never shifted */ + audioFormat->fracBits = 0; return 0; } |