| 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 a84b4be018cfc06a8ff6fef35f30c07ec8d8f119..12be6d622e7e4b1f8ccd7530d063a87224f08b9f 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -578,7 +578,7 @@ size_t RTPSender::SendPadData(size_t bytes,
|
| pacing_info);
|
| }
|
|
|
| - if (!SendPacketToNetwork(padding_packet, options))
|
| + if (!SendPacketToNetwork(padding_packet, options, pacing_info))
|
| break;
|
|
|
| bytes_sent += padding_bytes_in_packet;
|
| @@ -630,7 +630,8 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
|
| }
|
|
|
| bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
|
| - const PacketOptions& options) {
|
| + const PacketOptions& options,
|
| + const PacedPacketInfo& pacing_info) {
|
| int bytes_sent = -1;
|
| if (transport_) {
|
| UpdateRtpOverhead(packet);
|
| @@ -639,7 +640,7 @@ bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
|
| : -1;
|
| if (event_log_ && bytes_sent > 0) {
|
| event_log_->LogRtpHeader(kOutgoingPacket, MediaType::ANY, packet.data(),
|
| - packet.size());
|
| + packet.size(), pacing_info.probe_cluster_id);
|
| }
|
| }
|
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
| @@ -760,7 +761,7 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
|
| packet->Ssrc());
|
| }
|
|
|
| - if (!SendPacketToNetwork(*packet_to_send, options))
|
| + if (!SendPacketToNetwork(*packet_to_send, options, pacing_info))
|
| return false;
|
|
|
| {
|
| @@ -890,7 +891,7 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
|
| UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
|
| packet->Ssrc());
|
|
|
| - bool sent = SendPacketToNetwork(*packet, options);
|
| + bool sent = SendPacketToNetwork(*packet, options, PacedPacketInfo());
|
|
|
| if (sent) {
|
| {
|
|
|