Chromium Code Reviews| Index: webrtc/call/call.cc | 
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc | 
| index 25dcf32a620c48965c22077d8c6d35cc3b298267..dd5ef9c7a87744cfd67a368b3f2bda719574bf78 100644 | 
| --- a/webrtc/call/call.cc | 
| +++ b/webrtc/call/call.cc | 
| @@ -212,6 +212,7 @@ class Call : public webrtc::Call, | 
| void OnSentPacket(const rtc::SentPacket& sent_packet) override; | 
| + bool SetRtpKeepAliveConfig(const RtpKeepAliveConfig& config) override; | 
| // Implements BitrateObserver. | 
| void OnNetworkChanged(uint32_t bitrate_bps, | 
| @@ -1143,6 +1144,20 @@ void Call::OnSentPacket(const rtc::SentPacket& sent_packet) { | 
| transport_send_->send_side_cc()->OnSentPacket(sent_packet); | 
| } | 
| +bool Call::SetRtpKeepAliveConfig(const RtpKeepAliveConfig& config) { | 
| + // Can be called by RtpTransportController not on the configuration thread. | 
| 
 
stefan-webrtc
2017/08/08 08:13:06
This comment will become confusing. Call currently
 
sprang_webrtc
2017/08/08 08:53:30
OK, so this badly phrased comment referred to the
 
 | 
| + | 
| + ReadLockScoped lock(*send_crit_); | 
| + if (config != config_.keepalive_config && | 
| + (!video_send_streams_.empty() || !audio_send_ssrcs_.empty())) { | 
| + LOG(LS_WARNING) << "RTP keep-alive settings cannot be altered after " | 
| + "creating send streams."; | 
| + return false; | 
| + } | 
| + config_.keepalive_config = config; | 
| + return true; | 
| +} | 
| + | 
| void Call::OnNetworkChanged(uint32_t target_bitrate_bps, | 
| uint8_t fraction_loss, | 
| int64_t rtt_ms, |