| Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| index 351056e216c02458d35fde5a619301c6178368d9..94503dcdf1c8c9464a0ffb1b1b67edd689233713 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| @@ -304,7 +304,6 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
|
| auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
|
|
|
| size_t fec_packet_overhead;
|
| - bool is_timing_frame = false;
|
| bool red_enabled;
|
| int32_t retransmission_settings;
|
| {
|
| @@ -336,7 +335,6 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
|
| if (video_header->video_timing.is_timing_frame) {
|
| last_packet->SetExtension<VideoTimingExtension>(
|
| video_header->video_timing);
|
| - is_timing_frame = true;
|
| }
|
| }
|
|
|
| @@ -396,7 +394,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
|
|
|
| bool protect_packet = (packetizer->GetProtectionType() == kProtectedPacket);
|
| // Put packetization finish timestamp into extension.
|
| - if (last && is_timing_frame) {
|
| + if (packet->HasExtension<VideoTimingExtension>()) {
|
| packet->set_packetization_finish_time_ms(clock_->TimeInMilliseconds());
|
| // TODO(ilnik): Due to webrtc:7859, packets with timing extensions are not
|
| // protected by FEC. It reduces FEC efficiency a bit. When FEC is moved
|
|
|