Send runtime settings to the Audio Processing Module (APM)
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}
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@ -35,6 +35,7 @@
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/protobuf_utils.h"
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#include "rtc_base/refcountedobject.h"
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#include "rtc_base/swap_queue.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/thread.h"
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#include "system_wrappers/include/event_wrapper.h"
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@ -2819,6 +2820,34 @@ INSTANTIATE_TEST_CASE_P(
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} // namespace
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TEST(RuntimeSettingTest, TestDefaultCtor) {
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auto s = AudioProcessing::RuntimeSetting();
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EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
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}
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TEST(RuntimeSettingTest, TestCapturePreGain) {
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using Type = AudioProcessing::RuntimeSetting::Type;
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{
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auto s = AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.25f);
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EXPECT_EQ(Type::kCapturePreGain, s.type());
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float v;
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s.GetFloat(&v);
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EXPECT_EQ(1.25f, v);
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}
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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
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EXPECT_DEATH(AudioProcessing::RuntimeSetting::CreateCapturePreGain(0.1f), "");
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#endif
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}
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TEST(RuntimeSettingTest, TestUsageWithSwapQueue) {
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SwapQueue<AudioProcessing::RuntimeSetting> q(1);
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auto s = AudioProcessing::RuntimeSetting();
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ASSERT_TRUE(q.Insert(&s));
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ASSERT_TRUE(q.Remove(&s));
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EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
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}
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TEST(ApmConfiguration, EnablePostProcessing) {
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// Verify that apm uses a capture post processing module if one is provided.
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webrtc::Config webrtc_config;
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