Move frame_type member from RtpDepacketizer::ParsedPayload to RTPVideoHeader
The latter is also a member of the former. This cleanup is also
a preparation for dropping WebRtcRTPHeader::frameType (or deleting
WebRtcRTPHeader right away), now that it's a video-specific member.
Tbr: kwiberg@webrtc.org # Comment change in modules/include/
Bug: None
Change-Id: I5c1f3f981f0d750713fc9b9b145278150fe32b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133024
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27740}
diff --git a/modules/video_coding/jitter_buffer.cc b/modules/video_coding/jitter_buffer.cc
index 51e828f..3d3b967 100644
--- a/modules/video_coding/jitter_buffer.cc
+++ b/modules/video_coding/jitter_buffer.cc
@@ -413,7 +413,7 @@
// Empty packets may bias the jitter estimate (lacking size component),
// therefore don't let empty packet trigger the following updates:
- if (packet.frameType != VideoFrameType::kEmptyFrame) {
+ if (packet.video_header.frame_type != VideoFrameType::kEmptyFrame) {
if (waiting_for_completion_.timestamp == packet.timestamp) {
// This can get bad if we have a lot of duplicate packets,
// we will then count some packet multiple times.
@@ -446,7 +446,7 @@
frame->IncrementNackCount();
}
if (!UpdateNackList(packet.seqNum) &&
- packet.frameType != VideoFrameType::kVideoFrameKey) {
+ packet.video_header.frame_type != VideoFrameType::kVideoFrameKey) {
buffer_state = kFlushIndicator;
}