| 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);
|
|
|
|
|