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422b9e0982
The audio processing in the band-split domain on ARM platforms operate at a sampling frequency of 32 kHz. This CL upsamples the signal to fullband before the "fullband processing" if an output rate of 48 kHz is chosen. Change-Id: I268acd33aff1fcfa4f75ba8c0fb3e16abb9f74e8 Bug: b/130016532 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155640 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29415}
91 lines
2.8 KiB
C++
91 lines
2.8 KiB
C++
/*
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* Copyright (c) 2016 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 "modules/audio_processing/audio_buffer.h"
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#include <cmath>
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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const size_t kSampleRateHz = 48000u;
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const size_t kStereo = 2u;
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const size_t kMono = 1u;
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void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
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EXPECT_EQ(ab.num_channels(), num_channels);
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}
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} // namespace
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TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
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AudioBuffer ab(kSampleRateHz, kStereo, kSampleRateHz, kStereo, kSampleRateHz,
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kStereo);
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ExpectNumChannels(ab, kStereo);
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ab.set_num_channels(1);
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ExpectNumChannels(ab, kMono);
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ab.RestoreNumChannels();
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ExpectNumChannels(ab, kStereo);
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}
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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
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TEST(AudioBufferTest, SetNumChannelsDeathTest) {
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AudioBuffer ab(kSampleRateHz, kMono, kSampleRateHz, kMono, kSampleRateHz,
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kMono);
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EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
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}
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#endif
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TEST(AudioBufferTest, CopyWithoutResampling) {
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AudioBuffer ab1(32000, 2, 32000, 2, 32000, 2);
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AudioBuffer ab2(32000, 2, 32000, 2, 32000, 2);
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// Fill first buffer.
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for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
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for (size_t i = 0; i < ab1.num_frames(); ++i) {
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ab1.channels()[ch][i] = i + ch;
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}
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}
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// Copy to second buffer.
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ab1.CopyTo(&ab2);
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// Verify content of second buffer.
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for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
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for (size_t i = 0; i < ab2.num_frames(); ++i) {
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EXPECT_EQ(ab2.channels()[ch][i], i + ch);
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}
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}
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}
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TEST(AudioBufferTest, CopyWithResampling) {
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AudioBuffer ab1(32000, 2, 32000, 2, 48000, 2);
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AudioBuffer ab2(48000, 2, 48000, 2, 48000, 2);
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float energy_ab1 = 0.f;
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float energy_ab2 = 0.f;
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const float pi = std::acos(-1.f);
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// Put a sine and compute energy of first buffer.
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for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
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for (size_t i = 0; i < ab1.num_frames(); ++i) {
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ab1.channels()[ch][i] = std::sin(2 * pi * 100.f / 32000.f * i);
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energy_ab1 += ab1.channels()[ch][i] * ab1.channels()[ch][i];
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}
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}
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// Copy to second buffer.
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ab1.CopyTo(&ab2);
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// Compute energy of second buffer.
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for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
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for (size_t i = 0; i < ab2.num_frames(); ++i) {
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energy_ab2 += ab2.channels()[ch][i] * ab2.channels()[ch][i];
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}
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}
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// Verify that energies match.
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EXPECT_NEAR(energy_ab1, energy_ab2 * 32000.f / 48000.f, .01f * energy_ab1);
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}
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} // namespace webrtc
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