Remove WVoE::SetAudioDeviceModule() - it is now set in ctor.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 501 matching lines @@
   { kRedCodecName,    8000,  1, 127, false, { } },
   { kDtmfCodecName,   8000,  1, 126, false, { } },
 };
 } // namespace {
 
 bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
                                     webrtc::CodecInst* out) {
   return WebRtcVoiceCodecs::ToCodecInst(in, out);
 }
 
-WebRtcVoiceEngine::WebRtcVoiceEngine()
-    : voe_wrapper_(new VoEWrapper()),
-      audio_state_(webrtc::AudioState::Create(MakeAudioStateConfig(voe()))) {
-  Construct();
+WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm)
+    : WebRtcVoiceEngine(adm, new VoEWrapper()) {
+  audio_state_ = webrtc::AudioState::Create(MakeAudioStateConfig(voe()));
 }
 
-WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper)
+WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm,
+                                     VoEWrapper* voe_wrapper)
     : voe_wrapper_(voe_wrapper) {
-  Construct();
-}
-
-void WebRtcVoiceEngine::Construct() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
 
   signal_thread_checker_.DetachFromThread();
+  webrtc::Trace::SetTraceCallback(this);
   std::memset(&default_agc_config_, 0, sizeof(default_agc_config_));
-  voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
-
-  webrtc::Trace::set_level_filter(kDefaultTraceFilter);
-  webrtc::Trace::SetTraceCallback(this);
-
-  // Load our audio codec list.
-  codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
+  adm_ = adm;
+  if (adm_) {
+    adm_->AddRef();

[pthatcher1, 2016/03/25 23:23:21]

Should we use a scoped_refptr for adm_?

[Taylor Brandstetter, 2016/03/29 00:49:23]

+1

[the sun, 2016/03/29 21:46:40]

Done.

[the sun, 2016/03/29 21:46:40]

Done.

+  }
 }
 
 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
   if (adm_) {
-    voe_wrapper_.reset();

[Taylor Brandstetter, 2016/03/29 00:49:23]

Is there a reason this was called here, and if so, why is it no longer
necessary?

[the sun, 2016/03/29 21:46:40]

Looks like it wasn't needed; VoE's ADM reference should already have been
released from VoEBaseImpl::TerminateInternal() [called indirectly from
WVoE::Terminate()]. Note that previously adm_ would only be set if using an
external ADM. I've added a unit test to verify ref counting of the ADM is well
behaved.

     adm_->Release();
-    adm_ = NULL;
   }
   webrtc::Trace::SetTraceCallback(nullptr);
 }
 
-bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
+bool WebRtcVoiceEngine::Init() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  RTC_DCHECK(worker_thread == rtc::Thread::Current());
   LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
-  bool res = InitInternal();
-  if (res) {
-    LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
-  } else {
-    LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
-    Terminate();
-  }
-  return res;
-}
+  LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
 
-bool WebRtcVoiceEngine::InitInternal() {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
+
   // Temporarily turn logging level up for the Init call.
   webrtc::Trace::set_level_filter(kElevatedTraceFilter);
-  LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
   if (voe_wrapper_->base()->Init(adm_) == -1) {
     LOG_RTCERR0_EX(Init, voe_wrapper_->error());
     return false;
   }
   webrtc::Trace::set_level_filter(kDefaultTraceFilter);
 
+  // No ADM supplied? Get the default one from VoE.
+  if (!adm_) {
+    adm_ = voe_wrapper_->base()->audio_device_module();
+    adm_->AddRef();
+  }
+  RTC_DCHECK(adm_);
+
   // Save the default AGC configuration settings. This must happen before
   // calling ApplyOptions or the default will be overwritten.
   if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
     LOG_RTCERR0(GetAgcConfig);
     return false;
   }
 
   // Set default engine options.
   {
     AudioOptions options;
     options.echo_cancellation = rtc::Optional<bool>(true);
     options.auto_gain_control = rtc::Optional<bool>(true);
     options.noise_suppression = rtc::Optional<bool>(true);
     options.highpass_filter = rtc::Optional<bool>(true);
     options.stereo_swapping = rtc::Optional<bool>(false);
     options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
     options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
     options.typing_detection = rtc::Optional<bool>(true);
     options.adjust_agc_delta = rtc::Optional<int>(0);
     options.experimental_agc = rtc::Optional<bool>(false);
     options.extended_filter_aec = rtc::Optional<bool>(false);
     options.delay_agnostic_aec = rtc::Optional<bool>(false);
     options.experimental_ns = rtc::Optional<bool>(false);
     if (!ApplyOptions(options)) {
       return false;
     }
   }
 
   // Print our codec list again for the call diagnostic log.
-  LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
+  LOG(LS_INFO) << "Supported codecs in order of preference:";
+  codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
   for (const AudioCodec& codec : codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
   SetDefaultDevices();
 
-  initialized_ = true;
+  LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
   return true;
 }
 
 void WebRtcVoiceEngine::Terminate() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
-  initialized_ = false;
-
   StopAecDump();
-
   voe_wrapper_->base()->Terminate();
 }
 
 rtc::scoped_refptr<webrtc::AudioState>
     WebRtcVoiceEngine::GetAudioState() const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return audio_state_;
 }
 
 VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
@@ skipping 404 matching lines @@
                << default_agc_config_.targetLeveldBOv << "dB to -"
                << config.targetLeveldBOv << "dB";
 
   if (voe_wrapper_->processing()->SetAgcConfig(config) == -1) {
     LOG_RTCERR1(SetAgcConfig, config.targetLeveldBOv);
     return false;
   }
   return true;
 }
 
-bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  if (initialized_) {
-    LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
-    return false;
-  }
-  if (adm_) {
-    adm_->Release();
-    adm_ = NULL;
-  }
-  if (adm) {
-    adm_ = adm;
-    adm_->AddRef();
-  }
-  return true;
-}
-
 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
                                      int64_t max_size_bytes) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
   if (!aec_dump_file_stream) {
     LOG(LS_ERROR) << "Could not open AEC dump file stream.";
     if (!rtc::ClosePlatformFile(file))
       LOG(LS_WARNING) << "Could not close file.";
     return false;
   }
@@ skipping 1466 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE