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https://github.com/mpv-player/mpv
synced 2024-09-05 02:48:21 +02:00
Remove some pointless 'inline' qualifiers
Most of these functions aren't even used in the same translation unit. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24918 b3059339-0415-0410-9bf9-f77b7e298cf2
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@ -572,7 +572,7 @@ double af_calc_delay(af_stream_t* s)
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/* Helper function called by the macro with the same name this
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function should not be called directly */
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inline int af_resize_local_buffer(af_instance_t* af, af_data_t* data)
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int af_resize_local_buffer(af_instance_t* af, af_data_t* data)
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{
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// Calculate new length
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register int len = af_lencalc(af->mul,data);
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@ -4,7 +4,7 @@
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/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
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fail */
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inline int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
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int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
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{
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int i = 0;
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// Sanity check
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@ -22,7 +22,7 @@ inline int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
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/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
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fail */
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inline int af_to_dB(int n, float* in, float* out, float k)
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int af_to_dB(int n, float* in, float* out, float k)
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{
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int i = 0;
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// Sanity check
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@ -39,7 +39,7 @@ inline int af_to_dB(int n, float* in, float* out, float k)
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}
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/* Convert from ms to sample time */
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inline int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
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int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
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{
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int i = 0;
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// Sanity check
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@ -53,7 +53,7 @@ inline int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
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}
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/* Convert from sample time to ms */
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inline int af_to_ms(int n, int* in, float* out, int rate)
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int af_to_ms(int n, int* in, float* out, int rate)
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{
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int i = 0;
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// Sanity check
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@ -67,7 +67,7 @@ inline int af_to_ms(int n, int* in, float* out, int rate)
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}
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/* Helper function for testing the output format */
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inline int af_test_output(struct af_instance_s* af, af_data_t* out)
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int af_test_output(struct af_instance_s* af, af_data_t* out)
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{
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if((af->data->format != out->format) ||
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(af->data->bps != out->bps) ||
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@ -81,7 +81,7 @@ inline int af_test_output(struct af_instance_s* af, af_data_t* out)
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/* Soft clipping, the sound of a dream, thanks to Jon Wattes
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post to Musicdsp.org */
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inline float af_softclip(float a)
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float af_softclip(float a)
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{
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if (a >= M_PI/2)
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return 1.0;
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@ -46,7 +46,7 @@ inline _ftype_t af_filter_fir(register unsigned int n, _ftype_t* w, _ftype_t* x)
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y output buffer
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s output buffer stride
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*/
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inline _ftype_t* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** w, _ftype_t** x, _ftype_t* y, unsigned int s)
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_ftype_t* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** w, _ftype_t** x, _ftype_t* y, unsigned int s)
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{
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register _ftype_t* xt = *x + xi;
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register _ftype_t* wt = *w;
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@ -65,7 +65,7 @@ inline _ftype_t* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi,
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at the new samples, xi current index in xq and n the length of the
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filter. xq must be n*2 by k big, s is the index for in.
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*/
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inline int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s)
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int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s)
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{
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register _ftype_t* txq = *xq + xi;
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register int nt = n*2;
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@ -62,7 +62,7 @@ int af_str2fmt(const char* str)
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return format;
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}
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inline int af_fmt2bits(int format)
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int af_fmt2bits(int format)
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{
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return (format & AF_FORMAT_BITS_MASK)+8;
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// return (((format & AF_FORMAT_BITS_MASK)>>3)+1) * 8;
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@ -79,7 +79,7 @@ inline int af_fmt2bits(int format)
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return -1;
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}
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inline int af_bits2fmt(int bits)
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int af_bits2fmt(int bits)
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{
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return (bits/8 - 1) << 3;
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}
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@ -2566,7 +2566,7 @@ int vo_xv_init_colorkey(void)
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* It doesn't call XFlush.
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*
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*/
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inline void vo_xv_draw_colorkey( int32_t x, int32_t y,
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void vo_xv_draw_colorkey( int32_t x, int32_t y,
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int32_t w, int32_t h )
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{
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if( xv_ck_info.method == CK_METHOD_MANUALFILL ||
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