| Index: media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc
|
| diff --git a/media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc b/media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc
|
| deleted file mode 100644
|
| index 5fe3a92b61b915720d91f4c6039d25b3760d3b51..0000000000000000000000000000000000000000
|
| --- a/media/cast/net/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/net/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
|
|
|