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author | wm4 <wm4@nowhere> | 2015-04-01 21:39:40 +0200 |
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committer | wm4 <wm4@nowhere> | 2015-04-01 21:39:40 +0200 |
commit | f5603cba2359767a1c4be2713a9acd031e021d02 (patch) | |
tree | cb4f7970b26038ee1cb069e2d07bb76a3cfee218 /audio/filter | |
parent | 4987c1906d26745cb6a69449f6aa579387e1d9fe (diff) | |
download | mpv-f5603cba2359767a1c4be2713a9acd031e021d02.tar.bz2 mpv-f5603cba2359767a1c4be2713a9acd031e021d02.tar.xz |
af: remove unused functions
Diffstat (limited to 'audio/filter')
-rw-r--r-- | audio/filter/af.h | 2 | ||||
-rw-r--r-- | audio/filter/tools.c | 32 |
2 files changed, 0 insertions, 34 deletions
diff --git a/audio/filter/af.h b/audio/filter/af.h index cecc3897b3..4c67208123 100644 --- a/audio/filter/af.h +++ b/audio/filter/af.h @@ -159,9 +159,7 @@ double af_calc_delay(struct af_stream *s); int af_test_output(struct af_instance *af, struct mp_audio *out); int af_from_dB(int n, float *in, float *out, float k, float mi, float ma); -int af_to_dB(int n, float *in, float *out, float k); int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma); -int af_to_ms(int n, int *in, float *out, int rate); float af_softclip(float a); #endif /* MPLAYER_AF_H */ diff --git a/audio/filter/tools.c b/audio/filter/tools.c index 231477927f..8943f29f8b 100644 --- a/audio/filter/tools.c +++ b/audio/filter/tools.c @@ -40,24 +40,6 @@ int af_from_dB(int n, float* in, float* out, float k, float mi, float ma) return AF_OK; } -/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if - * fail. gain=0 will become -200 dB. k is just a multiplier. */ -int af_to_dB(int n, float* in, float* out, float k) -{ - int i = 0; - // Sanity check - if(!in || !out) - return AF_ERROR; - - for(i=0;i<n;i++){ - if(in[i] == 0.0) - out[i]=-200.0; - else - out[i]=k*log10(in[i]); - } - return AF_OK; -} - /* Convert from ms to sample time */ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma) { @@ -72,20 +54,6 @@ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma) return AF_OK; } -/* Convert from sample time to ms */ -int af_to_ms(int n, int* in, float* out, int rate) -{ - int i = 0; - // Sanity check - if(!in || !out || !rate) - return AF_ERROR; - - for(i=0;i<n;i++) - out[i]=1000.0 * (float)in[i]/((float)rate); - - return AF_OK; -} - /* * test if output format matches * af: audio filter |