b159c2e3dd
Ideally, PushSincResampler would have very little overhead on SincResampler. This gets closer to that ideal. Replace std::min/max and floor with inline functions. Add a benchmark test to verify the improvement. On a MacBook Retina, this results in PushSincResampler::Resample() accounting for ~1% of CPU usage on voe_cmd_test vs the earlier ~2% (with ISAC16 and 48 kHz audio devices). Using the new benchmark, this results in a performance improvement of: 16 -> 44.1 : 1.7x 16 -> 48 : 1.9x 32 -> 44.1 : 1.6x 32 -> 48 : 1.7x 44.1 -> 16 : 1.5x 44.1 -> 32 : 1.7x 44.1 -> 48 : 1.7x 48 -> 16 : 1.5x 48 -> 32 : 1.5x 48 -> 44.1 : 1.8x R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2157005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4695 4adac7df-926f-26a2-2b94-8c16560cd09d
70 lines
2.4 KiB
C++
70 lines
2.4 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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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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void ExpectArraysEq(const int16_t* ref, const int16_t* test, int length) {
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for (int i = 0; i < length; ++i) {
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EXPECT_EQ(test[i], ref[i]);
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}
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}
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TEST(AudioUtilTest, Clamp) {
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EXPECT_EQ(1000.f, ClampInt16(1000.f));
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EXPECT_EQ(32767.f, ClampInt16(32767.5f));
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EXPECT_EQ(-32768.f, ClampInt16(-32768.5f));
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}
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TEST(AudioUtilTest, Round) {
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EXPECT_EQ(0, RoundToInt16(0.f));
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EXPECT_EQ(0, RoundToInt16(0.4f));
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EXPECT_EQ(1, RoundToInt16(0.5f));
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EXPECT_EQ(0, RoundToInt16(-0.4f));
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EXPECT_EQ(-1, RoundToInt16(-0.5f));
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}
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TEST(AudioUtilTest, InterleavingStereo) {
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const int16_t kInterleaved[] = {2, 3, 4, 9, 8, 27, 16, 81};
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const int kSamplesPerChannel = 4;
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const int kNumChannels = 2;
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const int kLength = kSamplesPerChannel * kNumChannels;
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int16_t left[kSamplesPerChannel], right[kSamplesPerChannel];
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int16_t* deinterleaved[] = {left, right};
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Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
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const int16_t kRefLeft[] = {2, 4, 8, 16};
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const int16_t kRefRight[] = {3, 9, 27, 81};
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ExpectArraysEq(left, kRefLeft, kSamplesPerChannel);
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ExpectArraysEq(right, kRefRight, kSamplesPerChannel);
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int16_t interleaved[kLength];
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Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
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ExpectArraysEq(interleaved, kInterleaved, kLength);
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}
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TEST(AudioUtilTest, InterleavingMonoIsIdentical) {
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const int16_t kInterleaved[] = {1, 2, 3, 4, 5};
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const int kSamplesPerChannel = 5;
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const int kNumChannels = 1;
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int16_t mono[kSamplesPerChannel];
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int16_t* deinterleaved[] = {mono};
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Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
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ExpectArraysEq(mono, kInterleaved, kSamplesPerChannel);
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int16_t interleaved[kSamplesPerChannel];
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Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
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ExpectArraysEq(interleaved, mono, kSamplesPerChannel);
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}
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} // namespace webrtc
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