Index: webrtc/call/call_perf_tests.cc |
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc |
index 50e1a62cfb512f10a0b47f93c25631b3906c42af..137bb2bf6d2e05a255474d82991aab9a58f3ec1b 100644 |
--- a/webrtc/call/call_perf_tests.cc |
+++ b/webrtc/call/call_perf_tests.cc |
@@ -158,7 +158,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, |
ASSERT_STRNE("", audio_filename.c_str()); |
FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename, |
audio_rtp_speed); |
- EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr)); |
+ EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_)); |
Config voe_config; |
voe_config.Set<VoicePacing>(new VoicePacing(true)); |
int send_channel_id = voe_base->CreateChannel(voe_config); |
@@ -248,6 +248,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, |
audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; |
audio_recv_config.voe_channel_id = recv_channel_id; |
audio_recv_config.sync_group = kSyncGroup; |
+ audio_recv_config.decoder_factory = decoder_factory_; |
AudioReceiveStream* audio_receive_stream; |