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Unified Diff: webrtc/call/call_unittest.cc

Issue 1403363003: Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: missing file Created 5 years, 2 months ago
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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));
}

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