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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc

Issue 2007743003: Add sender controlled playout delay limits (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@cleanup_rtp_hdr_extensions
Patch Set: Created 4 years, 7 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 9d76c1a6163c033fbf275e891fa7ad001ab7c852..491f485392a130435a3ae0cf6e0081ea1a8a10fb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -98,6 +98,11 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
rtp_header->header.extension.videoRotation);
}
+ rtp_header->type.Video.min_playout_delay_ms =
+ rtp_header->header.extension.min_playout_delay_ms;
+ rtp_header->type.Video.max_playout_delay_ms =
+ rtp_header->header.extension.max_playout_delay_ms;
+
return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
parsed_payload.payload_length,
rtp_header) == 0

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