UMA log for audio_device Init and Start(Playout|Recording). Make Init return a more specific error …

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  Copyright (c) 2012 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 103 matching lines @@
   Terminate();
 }
 
 void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   LOGI() << "AttachAudioBuffer";
   RTC_DCHECK(audioBuffer);
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   audio_device_buffer_ = audioBuffer;
 }
 
-int32_t AudioDeviceIOS::Init() {
+AudioDeviceGeneric::InitStatus AudioDeviceIOS::Init() {
   LOGI() << "Init";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (initialized_) {
-    return 0;
+    return InitStatus::OK;
   }
 #if !defined(NDEBUG)
   LogDeviceInfo();
 #endif
   // Store the preferred sample rate and preferred number of channels already
   // here. They have not been set and confirmed yet since configureForWebRTC
   // is not called until audio is about to start. However, it makes sense to
   // store the parameters now and then verify at a later stage.
   RTCAudioSessionConfiguration* config =
       [RTCAudioSessionConfiguration webRTCConfiguration];
   playout_parameters_.reset(config.sampleRate,
                             config.outputNumberOfChannels);
   record_parameters_.reset(config.sampleRate,
                            config.inputNumberOfChannels);
   // Ensure that the audio device buffer (ADB) knows about the internal audio
   // parameters. Note that, even if we are unable to get a mono audio session,
   // we will always tell the I/O audio unit to do a channel format conversion
   // to guarantee mono on the "input side" of the audio unit.
   UpdateAudioDeviceBuffer();
   initialized_ = true;
-  return 0;
+  return InitStatus::OK;
 }
 
 int32_t AudioDeviceIOS::Terminate() {
   LOGI() << "Terminate";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (!initialized_) {
     return 0;
   }
   StopPlayout();
   StopRecording();
@@ skipping 681 matching lines @@
 
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
   [session lockForConfiguration];
   UnconfigureAudioSession();
   [session endWebRTCSession:nil];
   [session unlockForConfiguration];
 }
 
 }  // namespace webrtc