Android audio playout now supports non-call media streams

webrtc/modules/audio_device/android/audio_device_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 506 matching lines @@
   AudioDeviceTest()
       : test_is_done_(EventWrapper::Create()) {
     // One-time initialization of JVM and application context. Ensures that we
     // can do calls between C++ and Java. Initializes both Java and OpenSL ES
     // implementations.
     webrtc::audiodevicemodule::EnsureInitialized();
     // Creates an audio device using a default audio layer.
     audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
     EXPECT_NE(audio_device_.get(), nullptr);
     EXPECT_EQ(0, audio_device_->Init());
+    // Set audio mode to MODE_IN_COMMUNICATION.
+    audio_manager()->SetCommunicationMode(true);
     playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
     record_parameters_ = audio_manager()->GetRecordAudioParameters();
     build_info_.reset(new BuildInfo());
   }
   virtual ~AudioDeviceTest() {
     EXPECT_EQ(0, audio_device_->Terminate());
+    // Restore audio mode back to MODE_NORMAL.
+    audio_manager()->SetCommunicationMode(false);
   }
 
   int playout_sample_rate() const {
     return playout_parameters_.sample_rate();
   }
   int record_sample_rate() const {
     return record_parameters_.sample_rate();
   }
   size_t playout_channels() const {
     return playout_parameters_.channels();
@@ skipping 513 matching lines @@
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc