| Index: webrtc/call/call_unittest.cc
|
| diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
|
| index 9819b538f85c51e66edef33b7b9059a6febf502d..ea28c8ed49c1957328d8bcbde5fe28302cc2697f 100644
|
| --- a/webrtc/call/call_unittest.cc
|
| +++ b/webrtc/call/call_unittest.cc
|
| @@ -12,6 +12,7 @@
|
|
|
| #include "testing/gtest/include/gtest/gtest.h"
|
|
|
| +#include "webrtc/audio_state.h"
|
| #include "webrtc/call.h"
|
| #include "webrtc/test/fake_voice_engine.h"
|
|
|
| @@ -19,8 +20,10 @@ namespace {
|
|
|
| struct CallHelper {
|
| CallHelper() : voice_engine_(new webrtc::test::FakeVoiceEngine()) {
|
| + webrtc::AudioState::Config audio_state_config;
|
| + audio_state_config.voice_engine = voice_engine_.get();
|
| webrtc::Call::Config config;
|
| - config.voice_engine = voice_engine_.get();
|
| + config.audio_state = webrtc::AudioState::Create(audio_state_config);
|
| call_.reset(webrtc::Call::Create(config));
|
| }
|
|
|
|
|