External APM usage downstream dependency support cleanup

webrtc/voice_engine/voe_base_impl.cc

 /*
  *  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 207 matching lines @@
   for (voe::ChannelManager::Iterator it(&shared_->channel_manager());
        it.IsValid(); it.Increment()) {
     it.GetChannel()->DeRegisterVoiceEngineObserver();
   }
 
   return 0;
 }
 
 int VoEBaseImpl::Init(
     AudioDeviceModule* external_adm,
-    AudioProcessing* external_apm,
+    AudioProcessing* audio_processing,
     const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
-  // TODO(peah): Add a DCHECK for external_apm when downstream dependencies
-  // have properly been resolved.
+  RTC_DCHECK(audio_processing);
   rtc::CritScope cs(shared_->crit_sec());
   WebRtcSpl_Init();
   if (shared_->statistics().Initialized()) {
     return 0;
   }
   if (shared_->process_thread()) {
     shared_->process_thread()->Start();
   }
 
   // Create an internal ADM if the user has not added an external
@@ skipping 90 matching lines @@
   // determine the actual number of channels anyway.
   //
   // These functions may be changed; tracked here:
   // http://code.google.com/p/webrtc/issues/detail?id=204
   shared_->audio_device()->StereoRecordingIsAvailable(&available);
   if (shared_->audio_device()->SetStereoRecording(available) != 0) {
     shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
                           "Init() failed to set mono/stereo recording mode");
   }
 
-  // TODO(peah): Remove this when upstream dependencies have properly been
-  // resolved.
-  AudioProcessing* apm = nullptr;
-  if (!external_apm) {
-    audio_processing_ = AudioProcessing::Create();
-    if (!audio_processing_) {
-      // This can only happen if there are problems allocating the dynamic
-      // memory in the Create() call.
-      LOG(LS_ERROR) << "Failed to create AudioProcessing.";
-      shared_->SetLastError(VE_NO_MEMORY);
-      return -1;
-    }
-    apm = audio_processing_.get();
-  } else {
-    apm = external_apm;
-  }
-
-  shared_->set_audio_processing(apm);
+  shared_->set_audio_processing(audio_processing);
 
   // Set the error state for any failures in this block.
   shared_->SetLastError(VE_APM_ERROR);
   // Configure AudioProcessing components.
   // TODO(peah): Move this initialization to webrtcvoiceengine.cc.
-  if (apm->high_pass_filter()->Enable(true) != 0) {
+  if (audio_processing->high_pass_filter()->Enable(true) != 0) {
     LOG_F(LS_ERROR) << "Failed to enable high pass filter.";
     return -1;
   }
-  if (apm->echo_cancellation()->enable_drift_compensation(false) != 0) {
+  if (audio_processing->echo_cancellation()->enable_drift_compensation(false) !=
+      0) {
     LOG_F(LS_ERROR) << "Failed to disable drift compensation.";
     return -1;
   }
-  if (apm->noise_suppression()->set_level(kDefaultNsMode) != 0) {
+  if (audio_processing->noise_suppression()->set_level(kDefaultNsMode) != 0) {
     LOG_F(LS_ERROR) << "Failed to set noise suppression level: "
         << kDefaultNsMode;
     return -1;
   }
-  GainControl* agc = apm->gain_control();
+  GainControl* agc = audio_processing->gain_control();
   if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
     LOG_F(LS_ERROR) << "Failed to set analog level limits with minimum: "
         << kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel;
     return -1;
   }
   if (agc->set_mode(kDefaultAgcMode) != 0) {
     LOG_F(LS_ERROR) << "Failed to set mode: " << kDefaultAgcMode;
     return -1;
   }
   if (agc->Enable(kDefaultAgcState) != 0) {
@@ skipping 367 matching lines @@
     shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
         "AssociateSendChannel() failed to locate accociate_send_channel");
     return -1;
   }
 
   channel_ptr->set_associate_send_channel(ch);
   return 0;
 }
 
 }  // namespace webrtc