Index: webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
index 6fbf5f1297f8cc3254cd8c34625b0d00aac869ef..f7510c7972cc76457628255aea87d02a6838cb62 100644 |
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
@@ -55,6 +55,9 @@ ConferenceTransport::ConferenceTransport() |
local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); |
local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_); |
+ local_apm_ = webrtc::AudioProcessing::Create(); |
+ local_base_->Init(nullptr, local_apm_.get(), nullptr); |
+ |
// In principle, we can use one VoiceEngine to achieve the same goal. Well, in |
// here, we use two engines to make it more like reality. |
remote_voe_ = webrtc::VoiceEngine::Create(); |
@@ -64,7 +67,9 @@ ConferenceTransport::ConferenceTransport() |
remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_); |
remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_); |
- EXPECT_EQ(0, local_base_->Init()); |
+ remote_apm_ = webrtc::AudioProcessing::Create(); |
+ remote_base_->Init(nullptr, remote_apm_.get(), nullptr); |
+ |
local_sender_ = local_base_->CreateChannel(); |
static_cast<webrtc::VoiceEngineImpl*>(local_voe_) |
->GetChannelProxy(local_sender_) |
@@ -74,10 +79,8 @@ ConferenceTransport::ConferenceTransport() |
EXPECT_EQ(0, local_rtp_rtcp_-> |
SetSendAudioLevelIndicationStatus(local_sender_, true, |
kAudioLevelHeaderId)); |
- |
EXPECT_EQ(0, local_base_->StartSend(local_sender_)); |
- EXPECT_EQ(0, remote_base_->Init()); |
reflector_ = remote_base_->CreateChannel(); |
static_cast<webrtc::VoiceEngineImpl*>(remote_voe_) |
->GetChannelProxy(reflector_) |