commit c054e78f4e13223766b4bad56c76ac950879c97c
author Alessio Bazzica <alessiob@webrtc.org> Mon Apr 16 12:10:09 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Mon Apr 16 11:11:27 2018 +0000
tree a1e9975242055441f9232e4e321b8f6f95d8dca5
parent 5b07c240560c5dd055034fb84b74abecc9bcd0a1

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}

modules/audio_processing/audio_processing_unittest.cc

diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 6993da5..fbd81a1 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -35,6 +35,7 @@
 #include "rtc_base/numerics/safe_minmax.h"
 #include "rtc_base/protobuf_utils.h"
 #include "rtc_base/refcountedobject.h"
+#include "rtc_base/swap_queue.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread.h"
 #include "system_wrappers/include/event_wrapper.h"
@@ -2819,6 +2820,34 @@
 
 }  // namespace
 
+TEST(RuntimeSettingTest, TestDefaultCtor) {
+  auto s = AudioProcessing::RuntimeSetting();
+  EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
+}
+
+TEST(RuntimeSettingTest, TestCapturePreGain) {
+  using Type = AudioProcessing::RuntimeSetting::Type;
+  {
+    auto s = AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.25f);
+    EXPECT_EQ(Type::kCapturePreGain, s.type());
+    float v;
+    s.GetFloat(&v);
+    EXPECT_EQ(1.25f, v);
+  }
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+  EXPECT_DEATH(AudioProcessing::RuntimeSetting::CreateCapturePreGain(0.1f), "");
+#endif
+}
+
+TEST(RuntimeSettingTest, TestUsageWithSwapQueue) {
+  SwapQueue<AudioProcessing::RuntimeSetting> q(1);
+  auto s = AudioProcessing::RuntimeSetting();
+  ASSERT_TRUE(q.Insert(&s));
+  ASSERT_TRUE(q.Remove(&s));
+  EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
+}
+
 TEST(ApmConfiguration, EnablePostProcessing) {
   // Verify that apm uses a capture post processing module if one is provided.
   webrtc::Config webrtc_config;