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 |
deleted file mode 100644 |
index 1e1d14cd03c84849399dd7aad27cc4adb775c70e..0000000000000000000000000000000000000000 |
--- a/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc |
+++ /dev/null |
@@ -1,21 +0,0 @@ |
-// 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 |