Allow an external audio processing module to be used in WebRTC
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index f3c5557..e151c43 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -321,7 +321,8 @@
AudioProcessing* AudioProcessing::Create(const webrtc::Config& config,
NonlinearBeamformer* beamformer) {
- AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
+ AudioProcessingImpl* apm =
+ new rtc::RefCountedObject<AudioProcessingImpl>(config, beamformer);
if (apm->Initialize() != kNoError) {
delete apm;
apm = nullptr;