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Unified Diff: webrtc/call/call_perf_tests.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: better comments Created 5 years, 2 months ago
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Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index cab39144503e36a098fe7610bbbf4fba9a831c13..63928ab8243091cef335c1ed6fce5337ed7358d9 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -241,8 +241,12 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
voe_sync,
&audio_observer);
+ AudioState::Config audio_state_config;
+ audio_state_config.voice_engine = voice_engine;
+ rtc::scoped_ptr<AudioState> audio_state(
+ AudioState::Create(audio_state_config));
Call::Config receiver_config;
- receiver_config.voice_engine = voice_engine;
+ receiver_config.audio_state = audio_state.get();
CreateCalls(Call::Config(), receiver_config);
CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
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