webrtc/audio/audio_send_stream_unittest.cc - Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call.

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Unified Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Undid unneccessary changes to rtp_rtcp module. Created 4 years, 9 months ago

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Index: webrtc/audio/audio_send_stream_unittest.cc

diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc

index 6788699cd5003037536ffecc2872583ec908727c..27eb58ebd57cc9e5b5df10358d68df8516c666ba 100644

--- a/webrtc/audio/audio_send_stream_unittest.cc

+++ b/webrtc/audio/audio_send_stream_unittest.cc

@@ -55,7 +55,8 @@ struct ConfigHelper {

stream_config_(nullptr),

congestion_controller_(&simulated_clock_,

&bitrate_observer_,

- &remote_bitrate_observer_) {

+ &remote_bitrate_observer_,

+ nullptr) {

using testing::Invoke;

using testing::StrEq;

« webrtc/api/peerconnection.cc ('K') | « webrtc/audio/audio_receive_stream_unittest.cc ('k') | webrtc/call.h » ('j') | webrtc/call/rtc_event_log.cc » ('J')