Allow an external audio processing module to be used in WebRTC

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* audioproc,
+    AudioProcessing* external_apm,
     const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  // TODO(peah): Add a DCHECK for external_apmwhen upstream dependencies have

[Taylor Brandstetter, 2017/06/28 07:19:38]

Here too, and space between "apm" and "when"

[peah-webrtc, 2017/06/29 11:46:31]

Done.

+  // properly been resolved.
   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");
   }
 
-  if (!audioproc) {
-    audioproc = AudioProcessing::Create();
-    if (!audioproc) {
+  // TODO(peah): Remove this when upstream dependencies have properly been
+  // resolved.
+  AudioProcessing* apm = nullptr;
+  if (!external_apm) {
+    audio_processing_ = AudioProcessing::Create();
+    if (!audio_processing_) {
       LOG(LS_ERROR) << "Failed to create AudioProcessing.";
       shared_->SetLastError(VE_NO_MEMORY);

[Taylor Brandstetter, 2017/06/28 07:19:38]

Can you add a comment explaining why failure to create AudioProcessing can be
assumed to be a "no memory" error? Or change the Create method to return an
error?

[peah-webrtc, 2017/06/29 11:46:31]

This makes sense and this is not clear at all. However, this code will be
removed within a month. I added a short comment for now.

Done.

       return -1;
     }
+    apm = audio_processing_.get();
+  } else {
+    apm = external_apm;
   }
-  shared_->set_audio_processing(audioproc);
+
+  shared_->set_audio_processing(apm);
 
   // Set the error state for any failures in this block.
   shared_->SetLastError(VE_APM_ERROR);
   // Configure AudioProcessing components.
-  if (audioproc->high_pass_filter()->Enable(true) != 0) {
+  // TODO(peah): Move this initialization to webrtcvoiceengine.cc.
+  if (apm->high_pass_filter()->Enable(true) != 0) {
     LOG_F(LS_ERROR) << "Failed to enable high pass filter.";
     return -1;
   }
-  if (audioproc->echo_cancellation()->enable_drift_compensation(false) != 0) {
+  if (apm->echo_cancellation()->enable_drift_compensation(false) != 0) {
     LOG_F(LS_ERROR) << "Failed to disable drift compensation.";
     return -1;
   }
-  if (audioproc->noise_suppression()->set_level(kDefaultNsMode) != 0) {
+  if (apm->noise_suppression()->set_level(kDefaultNsMode) != 0) {
     LOG_F(LS_ERROR) << "Failed to set noise suppression level: "
         << kDefaultNsMode;
     return -1;
   }
-  GainControl* agc = audioproc->gain_control();
+  GainControl* agc = apm->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 303 matching lines @@
           "TerminateInternal() failed to de-register audio callback "
           "for the ADM");
     }
     if (shared_->audio_device()->Terminate() != 0) {
       shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
                             "TerminateInternal() failed to terminate the ADM");
     }
     shared_->set_audio_device(nullptr);
   }
 
-  if (shared_->audio_processing()) {
-    shared_->set_audio_processing(nullptr);
-  }
+  shared_->set_audio_processing(nullptr);
 
   return shared_->statistics().SetUnInitialized();
 }
 
 void VoEBaseImpl::GetPlayoutData(int sample_rate, size_t number_of_channels,
                                  size_t number_of_frames, bool feed_data_to_apm,
                                  void* audio_data, int64_t* elapsed_time_ms,
                                  int64_t* ntp_time_ms) {
   assert(shared_->output_mixer() != nullptr);
 
@@ skipping 43 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