1 files changed, 80 insertions, 0 deletions
diff --git a/webrtc/audio/audio_state_unittest.cc b/webrtc/audio/audio_state_unittest.cc
new file mode 100644
index 0000000000..11fbdb4a86
--- /dev/null
+++ b/webrtc/audio/audio_state_unittest.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/test/mock_voice_engine.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+struct ConfigHelper {
+  ConfigHelper() {
+    EXPECT_CALL(voice_engine_,
+        RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
+    EXPECT_CALL(voice_engine_,
+        DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
+    config_.voice_engine = &voice_engine_;
+  }
+  AudioState::Config& config() { return config_; }
+  MockVoiceEngine& voice_engine() { return voice_engine_; }
+
+ private:
+  testing::StrictMock<MockVoiceEngine> voice_engine_;
+  AudioState::Config config_;
+};
+}  // namespace
+
+TEST(AudioStateTest, Create) {
+  ConfigHelper helper;
+  rtc::scoped_refptr<AudioState> audio_state =
+      AudioState::Create(helper.config());
+  EXPECT_TRUE(audio_state.get());
+}
+
+TEST(AudioStateTest, ConstructDestruct) {
+  ConfigHelper helper;
+  rtc::scoped_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+}
+
+TEST(AudioStateTest, GetVoiceEngine) {
+  ConfigHelper helper;
+  rtc::scoped_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+  EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine());
+}
+
+TEST(AudioStateTest, TypingNoiseDetected) {
+  ConfigHelper helper;
+  rtc::scoped_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+  VoiceEngineObserver* voe_observer =
+      static_cast<VoiceEngineObserver*>(audio_state.get());
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+
+  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+
+  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
+  EXPECT_TRUE(audio_state->typing_noise_detected());
+  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+  EXPECT_TRUE(audio_state->typing_noise_detected());
+
+  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+}
+}  // namespace test
+}  // namespace webrtc