Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_state_unittest.cc

Index: webrtc/audio/audio_state_unittest.cc
diff --git a/webrtc/audio/audio_state_unittest.cc b/webrtc/audio/audio_state_unittest.cc
index 38485a88bf14665ce6b15d9a149cefec6513751f..05c5003459e94292af27d8eee45e289efb322e65 100644
--- a/webrtc/audio/audio_state_unittest.cc
+++ b/webrtc/audio/audio_state_unittest.cc
@@ -19,26 +19,65 @@ namespace webrtc {
 namespace test {
 namespace {
 
+const int kSampleRate = 8000;
+const int kNumberOfChannels = 1;
+const int kBytesPerSample = 2;
+
 struct ConfigHelper {
-  ConfigHelper() {
-    EXPECT_CALL(voice_engine_,
-        RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
-    EXPECT_CALL(voice_engine_,
-        DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
-    EXPECT_CALL(voice_engine_, audio_device_module());
-    EXPECT_CALL(voice_engine_, audio_processing());
-    EXPECT_CALL(voice_engine_, audio_transport());
-
-    config_.voice_engine = &voice_engine_;
-    config_.audio_mixer = AudioMixerImpl::Create();
+  ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
+    EXPECT_CALL(mock_voice_engine, RegisterVoiceEngineObserver(testing::_))
+        .WillOnce(testing::Return(0));
+    EXPECT_CALL(mock_voice_engine, DeRegisterVoiceEngineObserver())
+        .WillOnce(testing::Return(0));
+    EXPECT_CALL(mock_voice_engine, audio_device_module())
+        .Times(testing::AtLeast(1));
+    EXPECT_CALL(mock_voice_engine, audio_processing())
+        .Times(testing::AtLeast(1));
+    EXPECT_CALL(mock_voice_engine, audio_transport())
+        .WillRepeatedly(testing::Return(&audio_transport));
+
+    auto device = static_cast<MockAudioDeviceModule*>(
+        voice_engine().audio_device_module());
+
+    // Populate the audio transport proxy pointer to the most recent
+    // transport connected to the Audio Device.
+    ON_CALL(*device, RegisterAudioCallback(testing::_))
+        .WillByDefault(testing::Invoke([this](AudioTransport* transport) {
+          registered_audio_transport = transport;
+          return 0;
+        }));
+
+    audio_state_config.voice_engine = &mock_voice_engine;
+    audio_state_config.audio_mixer = audio_mixer;
   }
-  AudioState::Config& config() { return config_; }
-  MockVoiceEngine& voice_engine() { return voice_engine_; }
+  AudioState::Config& config() { return audio_state_config; }
+  MockVoiceEngine& voice_engine() { return mock_voice_engine; }
+  rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
+  MockAudioTransport& original_audio_transport() { return audio_transport; }
+  AudioTransport* audio_transport_proxy() { return registered_audio_transport; }
 
  private:
-  testing::StrictMock<MockVoiceEngine> voice_engine_;
-  AudioState::Config config_;
+  testing::StrictMock<MockVoiceEngine> mock_voice_engine;
+  AudioState::Config audio_state_config;
+  rtc::scoped_refptr<AudioMixer> audio_mixer;
+  MockAudioTransport audio_transport;
+  AudioTransport* registered_audio_transport = nullptr;
+};
+
+class FakeAudioSource : public AudioMixer::Source {
+ public:
+  // TODO(aleloi): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
+  int Ssrc() const /*override*/ { return 0; }
+
+  int PreferredSampleRate() const /*override*/ { return kSampleRate; }
+
+  MOCK_METHOD2(GetAudioFrameWithInfo,
+               AudioFrameInfo(int sample_rate_hz, AudioFrame* audio_frame));
 };
+
 }  // namespace
 
 TEST(AudioStateTest, Create) {
@@ -84,38 +123,52 @@ TEST(AudioStateTest, TypingNoiseDetected) {
 }
 
 // Test that RecordedDataIsAvailable calls get to the original transport.
-TEST(AudioStateTest, RecordedAudioArrivesAtOriginalTransport) {
-  using testing::_;
+TEST(AudioStateAudioPathTest, RecordedAudioArrivesAtOriginalTransport) {
+  ConfigHelper helper;
 
+  rtc::scoped_refptr<AudioState> audio_state =
+      AudioState::Create(helper.config());
+
+  // Setup completed. Ensure call of original transport is forwarded to new.
+  uint32_t new_mic_level;
+  EXPECT_CALL(
+      helper.original_audio_transport(),
+      RecordedDataIsAvailable(nullptr, kSampleRate / 100, kBytesPerSample,
+                              kNumberOfChannels, kSampleRate, 0, 0, 0, false,
+                              testing::Ref(new_mic_level)));
+
+  helper.audio_transport_proxy()->RecordedDataIsAvailable(
+      nullptr, kSampleRate / 100, kBytesPerSample, kNumberOfChannels,
+      kSampleRate, 0, 0, 0, false, new_mic_level);
+}
+
+TEST(AudioStateAudioPathTest,
+     QueryingProxyForAudioShouldResultInGetAudioCallOnMixerSource) {
   ConfigHelper helper;
-  auto& voice_engine = helper.voice_engine();
-  auto device =
-      static_cast<MockAudioDeviceModule*>(voice_engine.audio_device_module());
-
-  AudioTransport* audio_transport_proxy = nullptr;
-  ON_CALL(*device, RegisterAudioCallback(_))
-      .WillByDefault(
-          testing::Invoke([&audio_transport_proxy](AudioTransport* transport) {
-            audio_transport_proxy = transport;
-            return 0;
-          }));
 
-  MockAudioTransport original_audio_transport;
-  ON_CALL(voice_engine, audio_transport())
-      .WillByDefault(testing::Return(&original_audio_transport));
+  rtc::scoped_refptr<AudioState> audio_state =
+      AudioState::Create(helper.config());
+
+  FakeAudioSource fake_source;
 
-  EXPECT_CALL(voice_engine, audio_device_module());
-  std::unique_ptr<internal::AudioState> audio_state(
-      new internal::AudioState(helper.config()));
+  helper.mixer()->AddSource(&fake_source);
 
-  // Setup completed. Ensure call of old transport is forwarded to new.
-  uint32_t new_mic_level;
-  EXPECT_CALL(original_audio_transport,
-              RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0, 0, false,
-                                      testing::Ref(new_mic_level)));
+  EXPECT_CALL(fake_source, GetAudioFrameWithInfo(testing::_, testing::_))
+      .WillOnce(
+          testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
+            audio_frame->sample_rate_hz_ = sample_rate_hz;
+            audio_frame->samples_per_channel_ = sample_rate_hz / 100;
+            audio_frame->num_channels_ = kNumberOfChannels;
+            return AudioMixer::Source::AudioFrameInfo::kNormal;
+          }));
 
-  audio_transport_proxy->RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0,
-                                                 0, false, new_mic_level);
+  int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
+  size_t n_samples_out;
+  int64_t elapsed_time_ms;
+  int64_t ntp_time_ms;
+  helper.audio_transport_proxy()->NeedMorePlayData(
+      kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate,
+      audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc