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 |