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c054e78f4e
This CL includes the following changes: - APM runtime setting (ID + float payload) and unit tests - Swap queue of APM runtime settings used in AudioProcessingImpl - runtime settings handler that forwards the settings to APM sub-modules when a message is retrieved from the queue - Unit test placeholder to check that the pre-gain update message is correctly delivered Bug: webrtc:9138 Change-Id: Id22704af15fde2b87a4431f5ce64ad1aeafc5280 Reviewed-on: https://webrtc-review.googlesource.com/69320 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22873}
83 lines
2.6 KiB
C++
83 lines
2.6 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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} // 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())
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.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())
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.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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// TODO(bugs.chromium.org/9138): Implement this test as soon as the pre-gain
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// sub-module is implemented and it is notified by HandleRuntimeSettings()
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// when the gain changes.
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}
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} // namespace webrtc
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