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audiocommon: remove 1:1 interpolation
The usual one is 32->48 khz interpolation. So there is no need in a special 1:1 interpolation only for performance.
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parent
4a5ef06f71
commit
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@ -56,33 +56,6 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
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if (m_AIplaying) {
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numLeft = (numLeft > numSamples) ? numSamples : numLeft;
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if (AudioInterface::GetAIDSampleRate() == m_sampleRate) // (1:1)
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{
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#if _M_SSE >= 0x301
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if (cpu_info.bSSSE3 && !((numLeft * 2) % 8))
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{
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static const __m128i sr_mask =
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_mm_set_epi32(0x0C0D0E0FL, 0x08090A0BL,
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0x04050607L, 0x00010203L);
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for (unsigned int i = 0; i < numLeft * 2; i += 8)
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{
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_mm_storeu_si128((__m128i *)&samples[i], _mm_shuffle_epi8(_mm_loadu_si128((__m128i *)&m_buffer[(indexR + i) & INDEX_MASK]), sr_mask));
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}
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}
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else
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#endif
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{
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for (unsigned int i = 0; i < numLeft * 2; i+=2)
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{
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samples[i] = Common::swap16(m_buffer[(indexR + i + 1) & INDEX_MASK]);
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samples[i+1] = Common::swap16(m_buffer[(indexR + i) & INDEX_MASK]);
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}
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}
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indexR += numLeft * 2;
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}
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else //linear interpolation
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{
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//render numleft sample pairs to samples[]
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//advance indexR with sample position
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//remember fractional offset
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@ -109,7 +82,6 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
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indexR += 2 * (u16)(frac >> 16);
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frac &= 0xffff;
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}
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}
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} else {
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numLeft = 0;
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