Improves error handling for playout initialization on Android

webrtc/modules/audio_device/android/audio_track_jni.cc

 /*
  *  Copyright (c) 2013 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 15 matching lines @@
 #define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
 #define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
 
 namespace webrtc {
 
 // AudioTrackJni::JavaAudioTrack implementation.
 AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
     NativeRegistration* native_reg,
     std::unique_ptr<GlobalRef> audio_track)
     : audio_track_(std::move(audio_track)),
-      init_playout_(native_reg->GetMethodId("initPlayout", "(II)V")),
+      init_playout_(native_reg->GetMethodId("initPlayout", "(II)Z")),
       start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
       stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
       set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
       get_stream_max_volume_(
           native_reg->GetMethodId("getStreamMaxVolume", "()I")),
       get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {}
 
 AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
 
-void AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
-  audio_track_->CallVoidMethod(init_playout_, sample_rate, channels);
+bool AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
+  return audio_track_->CallBooleanMethod(init_playout_, sample_rate, channels);
 }
 
 bool AudioTrackJni::JavaAudioTrack::StartPlayout() {
   return audio_track_->CallBooleanMethod(start_playout_);
 }
 
 bool AudioTrackJni::JavaAudioTrack::StopPlayout() {
   return audio_track_->CallBooleanMethod(stop_playout_);
 }
 
@@ skipping 58 matching lines @@
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   StopPlayout();
   return 0;
 }
 
 int32_t AudioTrackJni::InitPlayout() {
   ALOGD("InitPlayout%s", GetThreadInfo().c_str());
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(!initialized_);
   RTC_DCHECK(!playing_);
-  j_audio_track_->InitPlayout(
-      audio_parameters_.sample_rate(), audio_parameters_.channels());
+  if (!j_audio_track_->InitPlayout(
+      audio_parameters_.sample_rate(), audio_parameters_.channels())) {
+    ALOGE("InitPlayout failed!");
+    return -1;
+  }
   initialized_ = true;
   return 0;
 }
 
 int32_t AudioTrackJni::StartPlayout() {
   ALOGD("StartPlayout%s", GetThreadInfo().c_str());
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(initialized_);
   RTC_DCHECK(!playing_);
   if (!j_audio_track_->StartPlayout()) {
@@ skipping 113 matching lines @@
     return;
   }
   RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_);
   // Copy decoded data into common byte buffer to ensure that it can be
   // written to the Java based audio track.
   samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
   RTC_DCHECK_EQ(length, kBytesPerFrame * samples);
 }
 
 }  // namespace webrtc