Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index e9659c67e8c48ecd3bf09369d27562322b1e5be3..ae5d8c1ed5f04db81c3d8a48c123de4e5f6b3454 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -39,6 +39,7 @@ std::string AudioSendStream::Config::Rtp::ToString() const { |
} |
} |
ss << ']'; |
+ ss << ", nack: " << nack.ToString(); |
ss << ", c_name: " << c_name; |
ss << '}'; |
return ss.str(); |
@@ -75,6 +76,10 @@ AudioSendStream::AudioSendStream( |
channel_proxy_->SetRTCPStatus(true); |
channel_proxy_->SetLocalSSRC(config.rtp.ssrc); |
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); |
+ // TODO(solenberg): Config NACK history window (which is a packet count), |
+ // using the actual packet size for the configured codec. |
+ channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, |
+ config_.rtp.nack.rtp_history_ms / 20); |
channel_proxy_->RegisterExternalTransport(config.send_transport); |