[rtp_rtcp] Lint build/header_guard errors fixed
BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1506043003 Cr-Commit-Position: refs/heads/master@{#10949}
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@ -8,8 +8,8 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
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// This file contains a set of packets masks for the FEC code. The masks in
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// this table are specifically designed to favor recovery to random loss.
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@ -24519,4 +24519,4 @@ const uint8_t** kPacketMaskRandomTbl[48] = {
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};
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} // namespace
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
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@ -8,8 +8,8 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
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#include "testing/gmock/include/gmock/gmock.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
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@ -40,4 +40,4 @@ class MockRTPPayloadStrategy : public RTPPayloadStrategy {
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
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@ -9,8 +9,8 @@
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*
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
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#define WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#include <map>
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#include <string>
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@ -819,4 +819,4 @@ class RawPacket {
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} // namespace rtcp
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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@ -8,8 +8,8 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
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#define WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
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#include <map>
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@ -114,4 +114,4 @@ class RtpHeaderExtensionMap {
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};
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}
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#endif // WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
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@ -10,8 +10,8 @@
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* Class for storing RTP packets.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
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#define WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
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#include <vector>
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@ -101,4 +101,4 @@ class RTPPacketHistory {
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std::vector<StoredPacket> stored_packets_ GUARDED_BY(critsect_);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
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@ -7,6 +7,8 @@
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
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#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/scoped_ptr.h"
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@ -68,3 +70,4 @@ class TestRtpReceiver : public NullRtpData {
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
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@ -7,6 +7,8 @@
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
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#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
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namespace {
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@ -186,3 +188,4 @@ const float kMaxResidualLossBurstyMask[kNumberCodes] = {
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};
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} // namespace
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#endif // WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
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