| 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
|
|
|