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; |