Fix for "Android audio playout doesn't support non-call media stream"

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 20 matching lines @@
     if (err != SL_RESULT_SUCCESS) {     \
       ALOGE("%s failed: %d", #op, err); \
       return __VA_ARGS__;               \
     }                                   \
   } while (0)
 
 namespace webrtc {
 
 OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
     : audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
+      stream_type_(audio_manager->OutputStreamType()),
       audio_device_buffer_(NULL),
       initialized_(false),
       playing_(false),
       bytes_per_buffer_(0),
       buffer_index_(0),
       engine_(nullptr),
       player_(nullptr),
       simple_buffer_queue_(nullptr),
       volume_(nullptr) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
+  RTC_DCHECK(stream_type_ == SL_ANDROID_STREAM_VOICE ||
+             stream_type_ == SL_ANDROID_STREAM_RING ||
+             stream_type_ == SL_ANDROID_STREAM_MEDIA) << stream_type_;
   // Use native audio output parameters provided by the audio manager and
   // define the PCM format structure.
   pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
                                        audio_parameters_.sample_rate(),
                                        audio_parameters_.bits_per_sample());
   // Detach from this thread since we want to use the checker to verify calls
   // from the internal  audio thread.
   thread_checker_opensles_.DetachFromThread();
 }
 
@@ skipping 279 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;
+  SLint32 stream_type = stream_type_;
   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 76 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