| 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 cdd707978165d3395a993389502c1eaa7935da0b..0dfa062e3c0749ea14fba7eee7c9b2d74ad198bb 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -743,8 +743,11 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
|
| packet_to_send->SetExtension<AbsoluteSendTime>(
|
| AbsoluteSendTime::MsTo24Bits(now_ms));
|
|
|
| - if (packet_to_send->HasExtension<VideoTimingExtension>())
|
| - packet_to_send->set_pacer_exit_time_ms(now_ms);
|
| + // TODO(ilnik): (webrtc:7859) For now we can't modify pacer exit timestamp in
|
| + // video timing extension because only some packets have it and it will break
|
| + // FEC recovered packets, which will lead to corruptions. Ideally, here
|
| + // |packet->set_pacer_exit_time_ms(now_ms)| should be called if
|
| + // |VideoTimingExtension| is present.
|
|
|
| PacketOptions options;
|
| if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
|
| @@ -833,8 +836,11 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
|
| if (packet->capture_time_ms() > 0) {
|
| packet->SetExtension<TransmissionOffset>(
|
| kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
|
| - if (packet->HasExtension<VideoTimingExtension>())
|
| - packet->set_pacer_exit_time_ms(now_ms);
|
| + // TODO(ilnik): (webrtc:7859) For now we can't modify pacer exit timestamp
|
| + // in video timing extension because only some packets have it an it will
|
| + // break FEC recovered packets, which will lead to corruptions. Ideally,
|
| + // here |packet->set_pacer_exit_time_ms(now_ms)| should be called if
|
| + // |VideoTimingExtension| is present.
|
| }
|
| packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
|
|
|
|
|