diff --git a/miniaudio.h b/miniaudio.h
index aa14a5e4..7e09f08d 100644
--- a/miniaudio.h
+++ b/miniaudio.h
@@ -59280,47 +59280,6 @@ static MA_INLINE ma_int16 ma_linear_resampler_mix_s16(ma_int16 x, ma_int16 y, ma
     return (ma_int16)(x + (n >> MA_LINEAR_RESAMPLER_LERP_SHIFT));
 }
 
-static MA_INLINE void ma_linear_resampler_interpolate_frame_s16(ma_linear_resampler* pResampler, ma_uint32 invSampleRateOut, ma_int16* MA_RESTRICT pFrameOut)
-{
-    ma_uint32 c;
-    ma_uint32 a;
-    const ma_uint32 channels = pResampler->channels;
-
-    MA_ASSERT(pResampler != NULL);
-    MA_ASSERT(pFrameOut  != NULL);
-
-    /*
-    The fractional component (inTimeFrac) will be between 0 and the output sample rate. We need to apply a scaling
-    factor (invSampleRateOut). It is assumed invSampleRateOut has been shifted by MA_LINEAR_RESAMPLER_LERP_SHIFT.
-
-    The lerp below is based on the ma_mix_f32_fast(), but with fixed point math.
-    */
-    a = pResampler->inTimeFrac * invSampleRateOut;
-
-    MA_ASSUME(channels > 0);
-    for (c = 0; c < channels; c += 1) {
-        pFrameOut[c] = ma_linear_resampler_mix_s16(pResampler->x0.s16[c], pResampler->x1.s16[c], a);
-    }
-}
-
-static MA_INLINE void ma_linear_resampler_interpolate_frame_f32(ma_linear_resampler* pResampler, float invSampleRateOut, float* MA_RESTRICT pFrameOut)
-{
-    ma_uint32 c;
-    float a;
-    const ma_uint32 channels = pResampler->channels;
-
-    MA_ASSERT(pResampler != NULL);
-    MA_ASSERT(pFrameOut  != NULL);
-
-    a = pResampler->inTimeFrac * invSampleRateOut;
-
-    MA_ASSUME(channels > 0);
-    for (c = 0; c < channels; c += 1) {
-        float s = ma_mix_f32_fast(pResampler->x0.f32[c], pResampler->x1.f32[c], a);
-        pFrameOut[c] = s;
-    }
-}
-
 static MA_INLINE ma_result ma_linear_resampler_process_pcm_frames_s16_no_lpf(ma_linear_resampler* pResampler, const ma_int16* pFramesInS16, ma_uint64* pFrameCountIn, ma_int16* pFramesOutS16, ma_uint64* pFrameCountOut, ma_uint32 invSampleRateOut)
 {
     ma_uint64 frameCountIn;