media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc - Issue 25544003: Fix code style and gyp files in cast to build cast_unittest

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Side by Side Diff: media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.cc

Issue 25544003: Fix code style and gyp files in cast to build cast_unittest (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src

Patch Set: fixed gyp files Created 7 years, 2 months ago

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	1 // Copyright 2013 The Chromium Authors. All rights reserved.

	2 // Use of this source code is governed by a BSD-style license that can be

	3 // found in the LICENSE file.

	4

	5 #include "media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.h"

	6

	7 namespace media {

	8 namespace cast {

	9

	10 RtpPacketizerConfig::RtpPacketizerConfig()

	11 : ssrc(0),

	12 max_payload_length(kIpPacketSize - 28), // Default is IP-v4/UDP.

	13 audio(false),

	14 frequency(8000),

	15 payload_type(-1),

	16 sequence_number(0),

	17 rtp_timestamp(0) {

	18 }

	19

	20 } // namespace cast

	21 } // namespace media

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