Index: media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc |
diff --git a/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc b/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..1e1d14cd03c84849399dd7aad27cc4adb775c70e |
--- /dev/null |
+++ b/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc |
@@ -0,0 +1,21 @@ |
+// Copyright 2013 The Chromium Authors. All rights reserved. |
+// Use of this source code is governed by a BSD-style license that can be |
+// found in the LICENSE file. |
+ |
+#include "media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.h" |
+ |
+namespace media { |
+namespace cast { |
+ |
+RtpPacketizerConfig::RtpPacketizerConfig() |
+ : ssrc(0), |
+ max_payload_length(kIpPacketSize - 28), // Default is IP-v4/UDP. |
+ audio(false), |
+ frequency(8000), |
+ payload_type(-1), |
+ sequence_number(0), |
+ rtp_timestamp(0) { |
+} |
+ |
+} // namespace cast |
+} // namespace media |