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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
52 lines
1.6 KiB
C++
52 lines
1.6 KiB
C++
/*
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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/sinc_resampler.cc
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#if defined(__arm__) || defined(_M_ARM) || defined(__aarch64__)
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#include "common_audio/resampler/sinc_resampler.h"
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#include <arm_neon.h>
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namespace webrtc {
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float SincResampler::Convolve_NEON(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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float32x4_t m_input;
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float32x4_t m_sums1 = vmovq_n_f32(0);
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float32x4_t m_sums2 = vmovq_n_f32(0);
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const float* upper = input_ptr + kKernelSize;
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for (; input_ptr < upper;) {
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m_input = vld1q_f32(input_ptr);
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input_ptr += 4;
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m_sums1 = vmlaq_f32(m_sums1, m_input, vld1q_f32(k1));
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k1 += 4;
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m_sums2 = vmlaq_f32(m_sums2, m_input, vld1q_f32(k2));
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k2 += 4;
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}
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// Linearly interpolate the two "convolutions".
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m_sums1 = vmlaq_f32(
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vmulq_f32(m_sums1, vmovq_n_f32(1.0 - kernel_interpolation_factor)),
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m_sums2, vmovq_n_f32(kernel_interpolation_factor));
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// Sum components together.
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float32x2_t m_half = vadd_f32(vget_high_f32(m_sums1), vget_low_f32(m_sums1));
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return vget_lane_f32(vpadd_f32(m_half, m_half), 0);
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}
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} // namespace webrtc
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#endif
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