| Index: webrtc/call/audio_send_stream.cc | 
| diff --git a/webrtc/call/audio_send_stream.cc b/webrtc/call/audio_send_stream.cc | 
| index 8b6dd9e416fc025f619aefb139e9732f334e2cb5..6091462470dba0786850e12b889577653706b397 100644 | 
| --- a/webrtc/call/audio_send_stream.cc | 
| +++ b/webrtc/call/audio_send_stream.cc | 
| @@ -40,7 +40,7 @@ AudioSendStream::Config::~Config() = default; | 
| std::string AudioSendStream::Config::ToString() const { | 
| std::stringstream ss; | 
| ss << "{rtp: " << rtp.ToString(); | 
| -  ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr"); | 
| +  ss << ", send_transport: " << (send_transport ? "(Transport)" : "null"); | 
| ss << ", voe_channel_id: " << voe_channel_id; | 
| ss << ", min_bitrate_bps: " << min_bitrate_bps; | 
| ss << ", max_bitrate_bps: " << max_bitrate_bps; | 
|  |