mirror of
https://github.com/danog/libtgvoip.git
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68 lines
2.3 KiB
C++
68 lines
2.3 KiB
C++
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/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Modified from the Chromium original:
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// src/media/base/simd/sinc_resampler_sse.cc
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#include "rtc_base/system/arch.h"
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#ifdef WEBRTC_ARCH_X86_FAMILY
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#include <stddef.h>
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#include <stdint.h>
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#include <xmmintrin.h>
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#include "common_audio/resampler/sinc_resampler.h"
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namespace webrtc {
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float SincResampler::Convolve_SSE(const float* input_ptr,
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const float* k1,
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const float* k2,
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double kernel_interpolation_factor) {
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__m128 m_input;
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__m128 m_sums1 = _mm_setzero_ps();
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__m128 m_sums2 = _mm_setzero_ps();
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// Based on |input_ptr| alignment, we need to use loadu or load. Unrolling
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// these loops hurt performance in local testing.
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if (reinterpret_cast<uintptr_t>(input_ptr) & 0x0F) {
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for (size_t i = 0; i < kKernelSize; i += 4) {
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m_input = _mm_loadu_ps(input_ptr + i);
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m_sums1 = _mm_add_ps(m_sums1, _mm_mul_ps(m_input, _mm_load_ps(k1 + i)));
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m_sums2 = _mm_add_ps(m_sums2, _mm_mul_ps(m_input, _mm_load_ps(k2 + i)));
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}
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} else {
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for (size_t i = 0; i < kKernelSize; i += 4) {
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m_input = _mm_load_ps(input_ptr + i);
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m_sums1 = _mm_add_ps(m_sums1, _mm_mul_ps(m_input, _mm_load_ps(k1 + i)));
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m_sums2 = _mm_add_ps(m_sums2, _mm_mul_ps(m_input, _mm_load_ps(k2 + i)));
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}
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}
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// Linearly interpolate the two "convolutions".
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m_sums1 = _mm_mul_ps(
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m_sums1,
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_mm_set_ps1(static_cast<float>(1.0 - kernel_interpolation_factor)));
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m_sums2 = _mm_mul_ps(
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m_sums2, _mm_set_ps1(static_cast<float>(kernel_interpolation_factor)));
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m_sums1 = _mm_add_ps(m_sums1, m_sums2);
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// Sum components together.
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float result;
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m_sums2 = _mm_add_ps(_mm_movehl_ps(m_sums1, m_sums1), m_sums1);
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_mm_store_ss(&result,
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_mm_add_ss(m_sums2, _mm_shuffle_ps(m_sums2, m_sums2, 1)));
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return result;
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}
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} // namespace webrtc
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#endif
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