Upstream version 10.39.225.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / video / video_receive_stream.cc

diff --git a/src/third_party/webrtc/video/video_receive_stream.cc b/src/third_party/webrtc/video/video_receive_stream.cc
index 012ca68..41a800f 100644 (file)
--- a/src/third_party/webrtc/video/video_receive_stream.cc
+++ b/src/third_party/webrtc/video/video_receive_stream.cc
@@ -52,14 +52,7 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
   // TODO(pbos): This is not fine grained enough...
   rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
   rtp_rtcp_->SetKeyFrameRequestMethod(channel_, kViEKeyFrameRequestPliRtcp);
-  switch (config_.rtp.rtcp_mode) {
-    case newapi::kRtcpCompound:
-      rtp_rtcp_->SetRTCPStatus(channel_, kRtcpCompound_RFC4585);
-      break;
-    case newapi::kRtcpReducedSize:
-      rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNonCompound_RFC5506);
-      break;
-  }
+  SetRtcpMode(config_.rtp.rtcp_mode);
 
   assert(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
@@ -150,13 +143,13 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
 
   external_codec_ = ViEExternalCodec::GetInterface(video_engine);
   for (size_t i = 0; i < config_.external_decoders.size(); ++i) {
-    ExternalVideoDecoder* decoder = &config_.external_decoders[i];
+    const ExternalVideoDecoder& decoder = config_.external_decoders[i];
     if (external_codec_->RegisterExternalReceiveCodec(
             channel_,
-            decoder->payload_type,
-            decoder->decoder,
-            decoder->renderer,
-            decoder->expected_delay_ms) != 0) {
+            decoder.payload_type,
+            decoder.decoder,
+            decoder.renderer,
+            decoder.expected_delay_ms) != 0) {
       // TODO(pbos): Abort gracefully? Can this be a runtime error?
       abort();
     }
@@ -264,5 +257,24 @@ int32_t VideoReceiveStream::RenderFrame(const uint32_t stream_id,
 
   return 0;
 }
+
+void VideoReceiveStream::SignalNetworkState(Call::NetworkState state) {
+  if (state == Call::kNetworkUp)
+    SetRtcpMode(config_.rtp.rtcp_mode);
+  network_->SetNetworkTransmissionState(channel_, state == Call::kNetworkUp);
+  if (state == Call::kNetworkDown)
+    rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNone);
+}
+
+void VideoReceiveStream::SetRtcpMode(newapi::RtcpMode mode) {
+  switch (mode) {
+    case newapi::kRtcpCompound:
+      rtp_rtcp_->SetRTCPStatus(channel_, kRtcpCompound_RFC4585);
+      break;
+    case newapi::kRtcpReducedSize:
+      rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNonCompound_RFC5506);
+      break;
+  }
+}
 }  // namespace internal
 }  // namespace webrtc