Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(182)

Unified Diff: media/cast/rtp_receiver/rtp_parser/rtp_parser.cc

Issue 214273003: [Cast] Remove AudioDecoder's dependency on WebRTC, and refactor/clean-up AudioReceiver. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 6 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
diff --git a/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc b/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
index fabc5fd93638ad53ca242a6a52d0939097483dd5..9a06c2b7d2962de80d1369e6d7674485f669fca1 100644
--- a/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
+++ b/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
@@ -112,7 +112,7 @@ bool RtpParser::ParseCast(const uint8* packet,
if (rtp_header->max_packet_id < rtp_header->packet_id)
return false;
- data_callback_->OnReceivedPayloadData(data_ptr, data_length, rtp_header);
+ data_callback_->OnReceivedPayloadData(data_ptr, data_length, *rtp_header);
return true;
}

Powered by Google App Engine
This is Rietveld 408576698