Revert of Android audio playout now supports non-call media streams

webrtc/modules/audio_device/android/opensles_player.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 290 matching lines @@
   // Use the Android configuration interface to set platform-specific
   // parameters. Should be done before player is realized.
   SLAndroidConfigurationItf player_config;
   RETURN_ON_ERROR(
       player_object_->GetInterface(player_object_.Get(),
                                    SL_IID_ANDROIDCONFIGURATION, &player_config),
       false);
   // Set audio player configuration to SL_ANDROID_STREAM_VOICE which
   // corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
   SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
-  if (!audio_manager_->IsCommunicationModeEnabled()) {
-    // But use STREAM_MUSIC if the user for some reason wants to run in a mode
-    // other than the preferred communication mode.
-    // SL_ANDROID_STREAM_MEDIA <=> android.media.AudioManager.STREAM_MUSIC.
-    stream_type = SL_ANDROID_STREAM_MEDIA;
-    ALOGW("Using non-default stream type SL_ANDROID_STREAM_MEDIA");
-  }
   RETURN_ON_ERROR(
       (*player_config)
           ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
                              &stream_type, sizeof(SLint32)),
       false);
 
   // Realize the audio player object after configuration has been set.
   RETURN_ON_ERROR(
       player_object_->Realize(player_object_.Get(), SL_BOOLEAN_FALSE), false);
 
@@ skipping 96 matching lines @@
   RTC_DCHECK(player_);
   SLuint32 state;
   SLresult err = (*player_)->GetPlayState(player_, &state);
   if (SL_RESULT_SUCCESS != err) {
     ALOGE("GetPlayState failed: %d", err);
   }
   return state;
 }
 
 }  // namespace webrtc