From cbf5b73658d36d411f7dd1b99dde339272197a34 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Fri, 8 Dec 2017 14:05:20 +0100 Subject: [PATCH] Explicitly convert size_t to int in Call::DeliverPacket to avoid breaking windows compiler TBR=stefan@webrtc.org Bug: None Change-Id: Idd6de316ddad76968283133982561b32292b3ad8 Reviewed-on: https://webrtc-review.googlesource.com/31400 Commit-Queue: Danil Chapovalov Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/master@{#21163} --- call/call.cc | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/call/call.cc b/call/call.cc index 99903d74b0..6a97d64c64 100644 --- a/call/call.cc +++ b/call/call.cc @@ -1323,7 +1323,6 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type, PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, rtc::CopyOnWriteBuffer packet, const PacketTime& packet_time) { - int length = packet.size(); TRACE_EVENT0("webrtc", "Call::DeliverRtp"); RtpPacketReceived parsed_packet; @@ -1361,6 +1360,9 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, NotifyBweOfReceivedPacket(parsed_packet, media_type); + // RateCounters expect input parameter as int, save it as int, + // instead of converting each time it is passed to RateCounter::Add below. + int length = static_cast(parsed_packet.size()); if (media_type == MediaType::AUDIO) { if (audio_receiver_controller_.OnRtpPacket(parsed_packet)) { received_bytes_per_second_counter_.Add(length);