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665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
123 lines
4.2 KiB
C++
123 lines
4.2 KiB
C++
/*
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* Copyright (c) 2014 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_processing_impl.h"
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#include "modules/audio_processing/test/test_utils.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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using ::testing::Invoke;
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namespace webrtc {
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namespace {
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class MockInitialize : public AudioProcessingImpl {
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public:
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explicit MockInitialize(const webrtc::Config& config)
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: AudioProcessingImpl(config) {}
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MOCK_METHOD0(InitializeLocked, int());
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int RealInitializeLocked() RTC_NO_THREAD_SAFETY_ANALYSIS {
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return AudioProcessingImpl::InitializeLocked();
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}
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MOCK_CONST_METHOD0(AddRef, void());
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MOCK_CONST_METHOD0(Release, rtc::RefCountReleaseStatus());
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};
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void GenerateFixedFrame(int16_t audio_level,
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size_t input_rate,
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size_t num_channels,
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AudioFrame* fixed_frame) {
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const size_t samples_per_input_channel = rtc::CheckedDivExact(
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input_rate, static_cast<size_t>(rtc::CheckedDivExact(
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1000, AudioProcessing::kChunkSizeMs)));
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fixed_frame->samples_per_channel_ = samples_per_input_channel;
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fixed_frame->sample_rate_hz_ = input_rate;
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fixed_frame->num_channels_ = num_channels;
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RTC_DCHECK_LE(samples_per_input_channel * num_channels,
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AudioFrame::kMaxDataSizeSamples);
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for (size_t i = 0; i < samples_per_input_channel * num_channels; ++i) {
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fixed_frame->mutable_data()[i] = audio_level;
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}
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}
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} // namespace
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TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
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webrtc::Config config;
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MockInitialize mock(config);
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ON_CALL(mock, InitializeLocked())
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.WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
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EXPECT_CALL(mock, InitializeLocked()).Times(1);
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mock.Initialize();
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AudioFrame frame;
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// Call with the default parameters; there should be an init.
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frame.num_channels_ = 1;
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SetFrameSampleRate(&frame, 16000);
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EXPECT_CALL(mock, InitializeLocked()).Times(0);
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EXPECT_NOERR(mock.ProcessStream(&frame));
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EXPECT_NOERR(mock.ProcessReverseStream(&frame));
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// New sample rate. (Only impacts ProcessStream).
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SetFrameSampleRate(&frame, 32000);
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EXPECT_CALL(mock, InitializeLocked()).Times(1);
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EXPECT_NOERR(mock.ProcessStream(&frame));
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// New number of channels.
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// TODO(peah): Investigate why this causes 2 inits.
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frame.num_channels_ = 2;
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EXPECT_CALL(mock, InitializeLocked()).Times(2);
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EXPECT_NOERR(mock.ProcessStream(&frame));
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// ProcessStream sets num_channels_ == num_output_channels.
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frame.num_channels_ = 2;
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EXPECT_NOERR(mock.ProcessReverseStream(&frame));
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// A new sample rate passed to ProcessReverseStream should cause an init.
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SetFrameSampleRate(&frame, 16000);
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EXPECT_CALL(mock, InitializeLocked()).Times(1);
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EXPECT_NOERR(mock.ProcessReverseStream(&frame));
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}
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TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
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std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
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webrtc::AudioProcessing::Config apm_config;
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apm_config.pre_amplifier.enabled = true;
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apm_config.pre_amplifier.fixed_gain_factor = 1.f;
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apm->ApplyConfig(apm_config);
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AudioFrame frame;
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constexpr int16_t audio_level = 10000;
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constexpr size_t input_rate = 48000;
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constexpr size_t num_channels = 2;
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GenerateFixedFrame(audio_level, input_rate, num_channels, &frame);
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apm->ProcessStream(&frame);
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EXPECT_EQ(frame.data()[100], audio_level)
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<< "With factor 1, frame shouldn't be modified.";
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constexpr float gain_factor = 2.f;
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCapturePreGain(gain_factor));
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// Process for two frames to have time to ramp up gain.
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for (int i = 0; i < 2; ++i) {
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GenerateFixedFrame(audio_level, input_rate, num_channels, &frame);
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apm->ProcessStream(&frame);
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}
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EXPECT_EQ(frame.data()[100], gain_factor * audio_level)
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<< "Frame should be amplified.";
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}
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} // namespace webrtc
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