Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
index 9ceebd99e02257027ebf137c7dfd59b8054e89ad..47ec31b9187584c11939ab5cba3e1a8e5bc7daf2 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -622,7 +622,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) { |
bool rtx = (RtxStatus() & kRtxRetransmitted) > 0; |
int32_t packet_size = static_cast<int32_t>(packet->size()); |
if (!PrepareAndSendPacket(std::move(packet), rtx, true, |
- PacketInfo::kNotAProbe)) |
+ PacedPacketInfo::kNotAProbe)) |
return -1; |
return packet_size; |
} |
@@ -880,7 +880,7 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet, |
PacketOptions options; |
if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id)) { |
AddPacketToTransportFeedback(options.packet_id, *packet.get(), |
- PacketInfo::kNotAProbe); |
+ PacedPacketInfo::kNotAProbe); |
} |
UpdateDelayStatistics(packet->capture_time_ms(), now_ms); |