Index: webrtc/api/call/audio_send_stream.cc |
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc |
index 06cbc545d9313846b057ed3432e2862c5c8b9b14..020470fa782d0606bbdf118df267539def2e1ad5 100644 |
--- a/webrtc/api/call/audio_send_stream.cc |
+++ b/webrtc/api/call/audio_send_stream.cc |
@@ -39,8 +39,8 @@ std::string AudioSendStream::Config::ToString() const { |
ss << "{rtp: " << rtp.ToString(); |
ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr"); |
ss << ", voe_channel_id: " << voe_channel_id; |
- ss << ", min_bitrate_kbps: " << min_bitrate_kbps; |
- ss << ", max_bitrate_kbps: " << max_bitrate_kbps; |
+ ss << ", min_bitrate_bps: " << min_bitrate_bps; |
+ ss << ", max_bitrate_bps: " << max_bitrate_bps; |
ss << ", send_codec_spec: " << send_codec_spec.ToString(); |
ss << '}'; |
return ss.str(); |