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Added beginnings of resampling code.
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@@ -142,10 +142,15 @@ typedef struct SDL_AudioCVT |
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SDL_AudioFormat dst_format; /* Target audio format */ |
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double rate_incr; /* Rate conversion increment */ |
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Uint8 *buf; /* Buffer to hold entire audio data */ |
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Uint8 *sinc; /* Windowed sinc filter */ |
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Uint8 *state_buf; /* Sample history for either the FIR or IIR filter */ |
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int state_pos; /* Position in the state */ |
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int len_sinc; /* Length of windowed sinc filter, in appropriate units (not necessarily bytes) */ |
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int len; /* Length of original audio buffer */ |
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int len_cvt; /* Length of converted audio buffer */ |
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int len_mult; /* buffer must be len*len_mult big */ |
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double len_ratio; /* Given len, final size is len*len_ratio */ |
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int len_div; /* desination length = len_mult / len_div * src length */ |
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double len_ratio; /* Given len, final size is len*len_ratio ( len_ratio = len_mult / len_div ) */ |
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SDL_AudioFilter filters[10]; /* Filter list */ |
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int filter_index; /* Current audio conversion function */ |
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} SDL_AudioCVT; |
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