Android audio playout now supports non-call media streams

webrtc/modules/audio_device/android/audio_manager_unittest.cc

 /*
  *  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.
  */
 
@@ skipping 17 matching lines @@
   AudioManagerTest() {
     // One-time initialization of JVM and application context. Ensures that we
     // can do calls between C++ and Java.
     webrtc::audiodevicemodule::EnsureInitialized();
     audio_manager_.reset(new AudioManager());
     SetActiveAudioLayer();
     playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
     record_parameters_ = audio_manager()->GetRecordAudioParameters();
   }
 
+  virtual ~AudioManagerTest() {
+    // Always ensure that we restore default/normal mode after the test.
+    audio_manager()->SetCommunicationMode(false);
+  }
+
   AudioManager* audio_manager() const { return audio_manager_.get(); }
 
   // A valid audio layer must always be set before calling Init(), hence we
   // might as well make it a part of the test fixture.
   void SetActiveAudioLayer() {
     EXPECT_EQ(0, audio_manager()->GetDelayEstimateInMilliseconds());
     audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio);
     EXPECT_NE(0, audio_manager()->GetDelayEstimateInMilliseconds());
   }
 
@@ skipping 52 matching lines @@
     // Perform a simple sanity check of the created engine object.
     ValidateSLEngine(engine_object);
   }
 }
 
 TEST_F(AudioManagerTest, InitClose) {
   EXPECT_TRUE(audio_manager()->Init());
   EXPECT_TRUE(audio_manager()->Close());
 }
 
+// Verify communication mode functionality.
+TEST_F(AudioManagerTest, CommunicationMode) {
+  audio_manager()->SetCommunicationMode(true);
+  EXPECT_TRUE(audio_manager()->IsCommunicationModeEnabled());
+  audio_manager()->SetCommunicationMode(false);
+  EXPECT_FALSE(audio_manager()->IsCommunicationModeEnabled());
+}
+
 TEST_F(AudioManagerTest, IsAcousticEchoCancelerSupported) {
   PRINT("%sAcoustic Echo Canceler support: %s\n", kTag,
         audio_manager()->IsAcousticEchoCancelerSupported() ? "Yes" : "No");
 }
 
 TEST_F(AudioManagerTest, IsAutomaticGainControlSupported) {
   EXPECT_FALSE(audio_manager()->IsAutomaticGainControlSupported());
 }
 
 TEST_F(AudioManagerTest, IsNoiseSuppressorSupported) {
@@ skipping 102 matching lines @@
             params.frames_per_10ms_buffer());
   EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
   EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
   EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
             params.GetBytesPer10msBuffer());
   EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
 }
 
 }  // namespace webrtc