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Review URL: http://webrtc-codereview.appspot.com/345011

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@1435 4adac7df-926f-26a2-2b94-8c16560cd09d
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asapersson@webrtc.org 2012-01-16 13:49:04 +00:00
parent c12f815de6
commit 43b8fc5c0d
4 changed files with 1 additions and 6 deletions
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2
src/modules/rtp_rtcp/source/rtp_header_extension.h
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@ -11,8 +11,6 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
#define WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
#include <stdint.h>
#include "map_wrapper.h"
#include "rtp_rtcp_defines.h"
#include "typedefs.h"

2
src/modules/rtp_rtcp/source/rtp_packet_history.h
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@ -15,11 +15,11 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
#define WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
#include <stdint.h>
#include <vector>
#include "module_common_types.h"
#include "rtp_rtcp_defines.h"
#include "typedefs.h"
namespace webrtc {

2
src/modules/rtp_rtcp/source/rtp_packet_history_test.cc
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@ -13,11 +13,9 @@
*/
#include <gtest/gtest.h>
#include <stdint.h>
#include "rtp_packet_history.h"
#include "rtp_rtcp_defines.h"
#include "rtp_utility.h"
#include "typedefs.h"
namespace webrtc {

1
src/modules/rtp_rtcp/source/transmission_bucket.h
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@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
#define WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
#include <stdint.h>
#include <vector>
#include "typedefs.h"