| Index: webrtc/audio/audio_receive_stream.cc
|
| diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
|
| index e391df471ed8c83ff228bacdc36b0db11da91a5f..aa6fde2bf1a95303b45d22be19fba7f6cc5ac471 100644
|
| --- a/webrtc/audio/audio_receive_stream.cc
|
| +++ b/webrtc/audio/audio_receive_stream.cc
|
| @@ -50,6 +50,8 @@ std::string AudioReceiveStream::Config::Rtp::ToString() const {
|
| std::stringstream ss;
|
| ss << "{remote_ssrc: " << remote_ssrc;
|
| ss << ", local_ssrc: " << local_ssrc;
|
| + ss << ", transport_cc: " << (transport_cc ? "on" : "off");
|
| + ss << ", nack: " << nack.ToString();
|
| ss << ", extensions: [";
|
| for (size_t i = 0; i < extensions.size(); ++i) {
|
| ss << extensions[i].ToString();
|
| @@ -58,7 +60,6 @@ std::string AudioReceiveStream::Config::Rtp::ToString() const {
|
| }
|
| }
|
| ss << ']';
|
| - ss << ", transport_cc: " << (transport_cc ? "on" : "off");
|
| ss << '}';
|
| return ss.str();
|
| }
|
| @@ -93,6 +94,10 @@ AudioReceiveStream::AudioReceiveStream(
|
| VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
|
| channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
|
| channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
|
| + // 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.rtcp_send_transport);
|
|
|
|
|