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Include files from webrtc/.. paths in rtp_rtcp/

Browse Source 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1557004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-05-29 14:27:38 +00:00
parent eea2622350
commit a048d7cb0a
75 changed files with 299 additions and 312 deletions
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4
webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
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@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ #define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
#include "typedefs.h" #include "webrtc/modules/interface/module_common_types.h"
#include "module_common_types.h"
#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/typedefs.h"
#ifndef NULL #ifndef NULL
#define NULL 0 #define NULL 0

8
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
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@ -11,11 +11,11 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#include <gmock/gmock.h> #include "testing/gmock/include/gmock/gmock.h"
#include "modules/interface/module.h" #include "webrtc/modules/interface/module.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
namespace webrtc { namespace webrtc {

2
webrtc/modules/rtp_rtcp/source/H264/bitstream_builder.h
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@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
class BitstreamBuilder class BitstreamBuilder

2
webrtc/modules/rtp_rtcp/source/H264/bitstream_parser.h
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@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
class BitstreamParser class BitstreamParser

1
webrtc/modules/rtp_rtcp/source/bitrate.h
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@ -12,6 +12,7 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
#include <stdio.h> #include <stdio.h>
#include <list> #include <list>
#include "webrtc/common_types.h" #include "webrtc/common_types.h"

3
webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
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@ -25,7 +25,7 @@
// (i.e., more packets/symbols in the code, so larger (k,m), i.e., k > 4, // (i.e., more packets/symbols in the code, so larger (k,m), i.e., k > 4,
// m > 3). // m > 3).
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace { namespace {
@ -758,4 +758,3 @@ const uint8_t** kPacketMaskBurstyTbl[12] = {
} // namespace } // namespace
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_ #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_

3
webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
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@ -15,7 +15,7 @@
// this table are specifically designed to favor recovery to random loss. // this table are specifically designed to favor recovery to random loss.
// These packet masks are defined to protect up to maximum of 48 media packets. // These packet masks are defined to protect up to maximum of 48 media packets.
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace { namespace {
@ -24520,4 +24520,3 @@ const uint8_t** kPacketMaskRandomTbl[48] = {
} // namespace } // namespace
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_ #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_

2
webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
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@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
#include "gmock/gmock.h" #include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
namespace webrtc { namespace webrtc {

2
webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
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@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_
#include "modules/rtp_rtcp/source/rtp_receiver_video.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
namespace webrtc { namespace webrtc {

6
webrtc/modules/rtp_rtcp/source/producer_fec.cc
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "modules/rtp_rtcp/source/producer_fec.h" #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
#include "modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc { namespace webrtc {

2
webrtc/modules/rtp_rtcp/source/producer_fec.h
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@ -13,7 +13,7 @@
#include <list> #include <list>
#include "modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
namespace webrtc { namespace webrtc {

8
webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
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@ -10,10 +10,10 @@
#include <list> #include <list>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "modules/rtp_rtcp/source/fec_test_helper.h" #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
#include "modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "modules/rtp_rtcp/source/producer_fec.h" #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
namespace webrtc { namespace webrtc {

7
webrtc/modules/rtp_rtcp/source/receiver_fec_unittest.cc
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@ -8,11 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <list>
#include <string.h> #include <string.h>
#include "gmock/gmock.h" #include <list>
#include "gtest/gtest.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h" #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h" #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h"

18
webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
#include "typedefs.h" #include "webrtc/common_types.h"
#include "common_types.h" #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtp_utility.h" #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "rtcp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "rtcp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "rtp_rtcp_impl.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h" #include "webrtc/typedefs.h"
namespace { namespace {

14
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "rtcp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include <string.h> //memset
#include <cassert> //assert #include <cassert> //assert
#include <string.h> //memset
#include "trace.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "trace_event.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "rtcp_utility.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "rtp_rtcp_impl.h" #include "webrtc/system_wrappers/interface/trace_event.h"
namespace namespace
{ {

12
webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
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@ -14,12 +14,12 @@
#include <map> #include <map>
#include <vector> #include <vector>
#include "typedefs.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
#include "rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "rtcp_receiver_help.h" #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
#include "tmmbr_help.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
class ModuleRtpRtcpImpl; class ModuleRtpRtcpImpl;

2
webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
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@ -10,8 +10,8 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
#include <string.h> // memset
#include <cassert> // assert #include <cassert> // assert
#include <string.h> // memset
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"

18
webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
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@ -12,17 +12,17 @@
/* /*
* This file includes unit tests for the RTCPReceiver. * This file includes unit tests for the RTCPReceiver.
*/ */
#include <gmock/gmock.h> #include "testing/gmock/include/gmock/gmock.h"
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
// Note: This file has no directory. Lint warning must be ignored. // Note: This file has no directory. Lint warning must be ignored.
#include "common_types.h" #include "webrtc/common_types.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtcp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc { namespace webrtc {

12
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
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@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "rtcp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include <algorithm> // min #include <algorithm> // min
#include <cassert> // assert #include <cassert> // assert
#include <cstdlib> // rand #include <cstdlib> // rand
#include <string.h> // memcpy #include <string.h> // memcpy
#include "common_types.h" #include "webrtc/common_types.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "system_wrappers/interface/trace_event.h" #include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc { namespace webrtc {

16
webrtc/modules/rtp_rtcp/source/rtcp_sender.h
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@ -15,14 +15,14 @@
#include <sstream> #include <sstream>
#include <string> #include <string>
#include "typedefs.h" #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtcp_utility.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "rtp_utility.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "tmmbr_help.h" #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

8
webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
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@ -13,17 +13,17 @@
* This file includes unit tests for the RTCPSender. * This file includes unit tests for the RTCPSender.
*/ */
#include <gmock/gmock.h> #include "testing/gmock/include/gmock/gmock.h"
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h" #include "webrtc/common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h" #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc { namespace webrtc {

6
webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include <cstring> // memcpy
#include <cmath> // ceil
#include <cassert> #include <cassert>
#include <cmath> // ceil
#include <cstring> // memcpy
namespace webrtc { namespace webrtc {
// RTCPParserV2 : currently read only // RTCPParserV2 : currently read only

6
webrtc/modules/rtp_rtcp/source/rtcp_utility.h
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@ -13,9 +13,9 @@
#include <cstddef> // size_t, ptrdiff_t #include <cstddef> // size_t, ptrdiff_t
#include "typedefs.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_rtcp_config.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "rtp_rtcp_defines.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
namespace RTCPUtility { namespace RTCPUtility {

2
webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
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@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <gtest/gtest.h>
#include <list> #include <list>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"

4
webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "modules/rtp_rtcp/source/rtp_format_vp8.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include <string.h> // memcpy #include <string.h> // memcpy
#include <cassert> // assert #include <cassert> // assert
#include <vector> #include <vector>
#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h" #include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
namespace webrtc { namespace webrtc {

6
webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
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@ -28,9 +28,9 @@
#include <queue> #include <queue>
#include <vector> #include <vector>
#include "modules/interface/module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"
#include "system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/constructor_magic.h"
#include "typedefs.h" // NOLINT(build/include) #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

4
webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
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@ -9,9 +9,9 @@
*/ */
#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
namespace webrtc { namespace webrtc {

8
webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
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@ -18,10 +18,10 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
#include "modules/interface/module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/constructor_magic.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

10
webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
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@ -13,13 +13,13 @@
* This file includes unit tests for the VP8 packetizer. * This file includes unit tests for the VP8 packetizer.
*/ */
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
#include "compile_assert.h" #include "webrtc/system_wrappers/interface/compile_assert.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

4
webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
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@ -10,8 +10,8 @@
#include <cassert> #include <cassert>
#include "common_types.h" #include "webrtc/common_types.h"
#include "rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
namespace webrtc { namespace webrtc {

4
webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
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@ -13,8 +13,8 @@
#include <map> #include <map>
#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

8
webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
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@ -13,11 +13,11 @@
* This file includes unit tests for the RtpHeaderExtensionMap. * This file includes unit tests for the RtpHeaderExtensionMap.
*/ */
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
#include "rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

8
webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "rtp_packet_history.h" #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include <assert.h> #include <assert.h>
#include <cstring> // memset #include <cstring> // memset
#include "critical_section_wrapper.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "rtp_utility.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "trace.h" #include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc { namespace webrtc {

6
webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
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@ -15,9 +15,9 @@
#include <vector> #include <vector>
#include "module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"
#include "rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

10
webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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@ -10,12 +10,12 @@
* This file includes unit tests for the RTPPacketHistory. * This file includes unit tests for the RTPPacketHistory.
*/ */
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
#include "clock.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_packet_history.h" #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include "rtp_rtcp_defines.h" #include "webrtc/system_wrappers/interface/clock.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

4
webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
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@ -10,8 +10,8 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
#include "gtest/gtest.h" #include "testing/gmock/include/gmock/gmock.h"
#include "gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h" #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h"

2
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -10,8 +10,8 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include <string.h>
#include <cassert> #include <cassert>
#include <string.h>
#include "webrtc/common_types.h" #include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"

6
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "rtp_sender_audio.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include <string.h> //memcpy
#include <cassert> //assert #include <cassert> //assert
#include <string.h> //memcpy
#include "trace_event.h" #include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc { namespace webrtc {
RTPSenderAudio::RTPSenderAudio(const int32_t id, Clock* clock, RTPSenderAudio::RTPSenderAudio(const int32_t id, Clock* clock,

14
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
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@ -11,14 +11,12 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH) #include "webrtc/common_types.h"
#include "common_types.h" // Transport #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
#include "typedefs.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "dtmf_queue.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "rtp_utility.h" #include "webrtc/typedefs.h"
#include "rtp_sender.h"
namespace webrtc { namespace webrtc {
class RTPSenderAudio: public DTMFqueue class RTPSenderAudio: public DTMFqueue

2
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -12,7 +12,7 @@
* This file includes unit tests for the RTPSender. * This file includes unit tests for the RTPSender.
*/ */
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/pacing/include/mock/mock_paced_sender.h" #include "webrtc/modules/pacing/include/mock/mock_paced_sender.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"

24
webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
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@ -8,21 +8,19 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "rtp_sender_video.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
#include "critical_section_wrapper.h" #include <assert.h>
#include "trace.h" #include <stdlib.h>
#include "trace_event.h" #include <string.h>
#include "rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include <string.h> // memcpy #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include <cassert> // assert #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include <cstdlib> // srand #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "producer_fec.h" #include "webrtc/system_wrappers/interface/trace_event.h"
#include "rtp_format_vp8.h"
#include "rtp_format_video_generic.h"
namespace webrtc { namespace webrtc {
enum { REDForFECHeaderLength = 1 }; enum { REDForFECHeaderLength = 1 };

22
webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
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@ -13,18 +13,16 @@
#include <list> #include <list>
#include "typedefs.h" #include "webrtc/common_types.h"
#include "common_types.h" // Transport #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_rtcp_config.h" #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
#include "rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "video_codec_information.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
#include "bitrate.h" #include "webrtc/typedefs.h"
#include "rtp_sender.h"
#include "producer_fec.h"
namespace webrtc { namespace webrtc {
class CriticalSectionWrapper; class CriticalSectionWrapper;

9
webrtc/modules/rtp_rtcp/source/rtp_utility.cc
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@ -8,15 +8,18 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include <cassert> #include <cassert>
#include <cmath> // ceil #include <cmath> // ceil
#include <cstring> // memcpy #include <cstring> // memcpy
#if defined(_WIN32) #if defined(_WIN32)
// Order for these headers are important
#include <Windows.h> // FILETIME #include <Windows.h> // FILETIME
#include <WinSock.h> // timeval #include <WinSock.h> // timeval
#include <MMSystem.h> // timeGetTime #include <MMSystem.h> // timeGetTime
#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC)) #elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
#include <sys/time.h> // gettimeofday #include <sys/time.h> // gettimeofday
@ -26,8 +29,8 @@
#include <stdio.h> #include <stdio.h>
#endif #endif
#include "system_wrappers/interface/tick_util.h" #include "webrtc/system_wrappers/interface/tick_util.h"
#include "system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace.h"
#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400)) #if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
#define DEBUG_PRINT(...) \ #define DEBUG_PRINT(...) \

8
webrtc/modules/rtp_rtcp/source/rtp_utility.h
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@ -13,10 +13,10 @@
#include <cstddef> // size_t, ptrdiff_t #include <cstddef> // size_t, ptrdiff_t
#include "typedefs.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "rtp_rtcp_config.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "rtp_rtcp_defines.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
enum RtpVideoCodecTypes enum RtpVideoCodecTypes

8
webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc
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@ -13,10 +13,10 @@
* This file conatins unit tests for the ModuleRTPUtility. * This file conatins unit tests for the ModuleRTPUtility.
*/ */
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "typedefs.h" // NOLINT(build/include) #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

10
webrtc/modules/rtp_rtcp/source/ssrc_database.cc
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "ssrc_database.h" #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
#include "critical_section_wrapper.h"
#include "trace.h"
#include <assert.h>
#include <stdlib.h> #include <stdlib.h>
#include <cassert>
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#ifdef _WIN32 #ifdef _WIN32
#include <windows.h> #include <windows.h>

4
webrtc/modules/rtp_rtcp/source/ssrc_database.h
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@ -15,8 +15,8 @@
#include <map> #include <map>
#endif #endif
#include "system_wrappers/interface/static_instance.h" #include "webrtc/system_wrappers/interface/static_instance.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
class CriticalSectionWrapper; class CriticalSectionWrapper;

4
webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "tmmbr_help.h" #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
#include <assert.h> #include <assert.h>
#include <limits> #include <limits>
#include <string.h> #include <string.h>
#include "rtp_rtcp_config.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
namespace webrtc { namespace webrtc {
TMMBRSet::TMMBRSet() : TMMBRSet::TMMBRSet() :

8
webrtc/modules/rtp_rtcp/source/tmmbr_help.h
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@ -12,13 +12,9 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
#include <vector> #include <vector>
#include "typedefs.h"
#include "critical_section_wrapper.h" #include "webrtc/typedefs.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#ifndef NULL
#define NULL 0
#endif
namespace webrtc { namespace webrtc {
class TMMBRSet class TMMBRSet

4
webrtc/modules/rtp_rtcp/source/video_codec_information.h
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@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
#include "rtp_rtcp_config.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc { namespace webrtc {
class VideoCodecInformation class VideoCodecInformation

3
webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h" #include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
#include <assert.h> #include <assert.h>
#include <stdlib.h> // NULL #include <stdlib.h> // NULL
@ -265,4 +265,3 @@ int Vp8PartitionAggregator::CalcNumberOfFragments(int large_partition_size,
} }
} // namespace } // namespace

6
webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
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@ -13,9 +13,9 @@
#include <vector> #include <vector>
#include "modules/interface/module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"
#include "system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/constructor_magic.h"
#include "typedefs.h" // NOLINT(build/include) #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {

5
webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
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@ -10,8 +10,8 @@
#include <stdlib.h> // NULL #include <stdlib.h> // NULL
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h" #include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
namespace webrtc { namespace webrtc {
@ -212,4 +212,3 @@ TEST(Vp8PartitionAggregator, TestCalcNumberOfFragments) {
} }
} // namespace } // namespace

14
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "BWEConvergenceTest.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h"
#include <fstream> #include <fstream>
#include <string> #include <string>
#include "TestSenderReceiver.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
#include "TestLoadGenerator.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
#include "event_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "thread_wrapper.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "tick_util.h" #include "webrtc/system_wrappers/interface/tick_util.h"
BWEConvergenceTestUp::BWEConvergenceTestUp(std::string testName, int startRateKbps, int availBWkbps) BWEConvergenceTestUp::BWEConvergenceTestUp(std::string testName, int startRateKbps, int availBWkbps)
@ -62,5 +62,3 @@ bool BWEConvergenceTestUp::StoppingCriterionMaster()
{ {
return ((_sendrec->BitrateSent() / 1000.0) > (0.9 * _availBWkbps)); return ((_sendrec->BitrateSent() / 1000.0) > (0.9 * _availBWkbps));
} }

8
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h
View File

@ -13,11 +13,9 @@
#include <string> #include <string>
#include "BWETestBase.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
#include "TestSenderReceiver.h"
class BWEConvergenceTestUp : public BWEOneWayTest class BWEConvergenceTestUp : public BWEOneWayTest
{ {

12
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc
View File

@ -8,13 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <fstream> #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h"
#include <math.h> #include <math.h>
#include "BWEStabilityTest.h" #include <fstream>
#include "TestLoadGenerator.h"
#include "tick_util.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
#include "critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
BWEStabilityTest::BWEStabilityTest(std::string testName, int rateKbps, int testDurationSeconds) BWEStabilityTest::BWEStabilityTest(std::string testName, int rateKbps, int testDurationSeconds)

8
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h
View File

@ -13,11 +13,9 @@
#include <string> #include <string>
#include "BWETestBase.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
#include "TestSenderReceiver.h"
class BWEStabilityTest : public BWEOneWayTest class BWEStabilityTest : public BWEOneWayTest
{ {

17
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
View File

@ -11,18 +11,18 @@
// BWEStandAlone.cpp : Defines the entry point for the console application. // BWEStandAlone.cpp : Defines the entry point for the console application.
// //
#include <string>
#include <stdio.h> #include <stdio.h>
#include <string>
#include "event_wrapper.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "udp_transport.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "rtp_rtcp.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "trace.h" #include "webrtc/test/channel_transport/udp_transport.h"
#include "TestSenderReceiver.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
#include "TestLoadGenerator.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
#include "MatlabPlot.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
//#include "vld.h" //#include "vld.h"
@ -197,4 +197,3 @@ int main(int argc, char* argv[])
//return 0; //return 0;
} }

16
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc
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@ -8,20 +8,20 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "BWETestBase.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
#include <algorithm> // sort #include <algorithm> // sort
#include <fstream> #include <fstream>
#include <math.h>
#include <string> #include <string>
#include <vector> #include <vector>
#include <math.h>
#include "TestSenderReceiver.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
#include "TestLoadGenerator.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
#include "event_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "thread_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "tick_util.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/tick_util.h"
double StatVec::Mean() double StatVec::Mean()

4
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
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@ -14,9 +14,9 @@
#include <string> #include <string>
#include <vector> #include <vector>
#include "typedefs.h" #include "webrtc/typedefs.h"
#include "TestSenderReceiver.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
class StatVec : public std::vector<double> class StatVec : public std::vector<double>

23
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc
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@ -11,27 +11,23 @@
// BWETester.cpp : Defines the entry point for the console application. // BWETester.cpp : Defines the entry point for the console application.
// //
#include <time.h>
#include <fstream> #include <fstream>
#include <string>
#include <iostream> #include <iostream>
#include <ctime> #include <string>
#include "event_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "trace.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h"
#include "BWEStabilityTest.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h"
#include "BWEConvergenceTest.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h"
#include "BWETwoWayLimitFinding.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
#include "MatlabPlot.h"
//#include "vld.h"
#ifdef MATLAB #ifdef MATLAB
MatlabEngine eng; MatlabEngine eng;
#endif #endif
class testContainer class testContainer
{ {
public: public:
@ -67,7 +63,6 @@ public:
std::string _instr; std::string _instr;
}; };
// This is were the test cases are created. // This is were the test cases are created.
// Syntax: // Syntax:
// tests->push_back(new testContainer( // tests->push_back(new testContainer(

5
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "BWETwoWayLimitFinding.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h"
#include "TestLoadGenerator.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
BWETwoWayLimitFinding::BWETwoWayLimitFinding( BWETwoWayLimitFinding::BWETwoWayLimitFinding(
@ -72,4 +72,3 @@ bool BWETwoWayLimitFinding::StoppingCriterionMaster()
return (_forwLimitReached && _revLimitReached); return (_forwLimitReached && _revLimitReached);
} }

2
webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h
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@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_ #define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
#include "BWETestBase.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
class BWETwoWayLimitFinding : public BWETest class BWETwoWayLimitFinding : public BWETest
{ {

21
webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
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@ -8,19 +8,22 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "MatlabPlot.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
#include <math.h>
#include <stdio.h>
#include <algorithm>
#include <sstream>
#ifdef MATLAB #ifdef MATLAB
#include "engine.h" #include "engine.h"
#endif #endif
#include "event_wrapper.h"
#include "thread_wrapper.h"
#include "critical_section_wrapper.h"
#include "tick_util.h"
#include <sstream> #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include <algorithm> #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include <math.h> #include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include <stdio.h> #include "webrtc/system_wrappers/interface/tick_util.h"
using namespace webrtc; using namespace webrtc;

2
webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
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@ -15,7 +15,7 @@
#include <string> #include <string>
#include <vector> #include <vector>
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
class CriticalSectionWrapper; class CriticalSectionWrapper;

16
webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
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@ -8,15 +8,17 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <algorithm> // for max function #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
#include <stdio.h> #include <stdio.h>
#include "TestLoadGenerator.h" #include <algorithm>
#include "TestSenderReceiver.h"
#include "event_wrapper.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
#include "thread_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "tick_util.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
bool SenderThreadFunction(void *obj) bool SenderThreadFunction(void *obj)

4
webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
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@ -13,8 +13,8 @@
#include <stdlib.h> #include <stdlib.h>
#include "typedefs.h" #include "webrtc/modules/interface/module_common_types.h"
#include "module_common_types.h" #include "webrtc/typedefs.h"
class TestSenderReceiver; class TestSenderReceiver;
namespace webrtc { namespace webrtc {

19
webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
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@ -8,18 +8,19 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <stdio.h> #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
#include "rtp_rtcp.h" #include <stdio.h>
#include "udp_transport.h"
#include "event_wrapper.h"
#include "thread_wrapper.h"
#include "tick_util.h"
#include "critical_section_wrapper.h"
#include "TestSenderReceiver.h"
#include "TestLoadGenerator.h"
#include <stdlib.h> #include <stdlib.h>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/test/channel_transport/udp_transport.h"
#define NR_OF_SOCKET_BUFFERS 500 #define NR_OF_SOCKET_BUFFERS 500

8
webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
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@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_ #define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
#include "typedefs.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_rtcp_defines.h" #include "webrtc/test/channel_transport/udp_transport.h"
#include "udp_transport.h" #include "webrtc/typedefs.h"
class TestLoadGenerator; class TestLoadGenerator;
namespace webrtc { namespace webrtc {

10
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
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@ -10,13 +10,13 @@
#include <algorithm> #include <algorithm>
#include <vector> #include <vector>
#include <gtest/gtest.h> #include "testing/gtest/include/gtest/gtest.h"
#include "test_api.h" #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
#include "common_types.h" #include "webrtc/common_types.h"
#include "rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
using namespace webrtc; using namespace webrtc;

12
webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
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@ -11,12 +11,12 @@
#include <algorithm> #include <algorithm>
#include <vector> #include <vector>
#include "gmock/gmock.h" #include "testing/gmock/include/gmock/gmock.h"
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "test_api.h" #include "webrtc/common_types.h"
#include "common_types.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
using namespace webrtc; using namespace webrtc;

2
webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
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@ -13,7 +13,7 @@
#include <algorithm> #include <algorithm>
#include <vector> #include <vector>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h" #include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"

10
webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
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@ -20,12 +20,12 @@
#include <ctime> #include <ctime>
#include <list> #include <list>
#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h" #include "webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h"
#include "modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "modules/rtp_rtcp/source/forward_error_correction_internal.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
#include "rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "testsupport/fileutils.h" #include "webrtc/test/testsupport/fileutils.h"
//#define VERBOSE_OUTPUT //#define VERBOSE_OUTPUT

2
webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
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@ -45,7 +45,7 @@
#include <math.h> #include <math.h>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
#include "webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h" #include "webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h"

15
webrtc/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc
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@ -8,15 +8,17 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <cassert>
#include <windows.h> #include <windows.h>
#include <iostream>
#include <assert.h>
#include <tchar.h> #include <tchar.h>
#include "rtp_rtcp.h" #include <iostream>
#include "common_types.h"
#include "rtcp_utility.h" #include "webrtc/common_types.h"
#include "tmmbr_help.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
#define TEST_STR "Test TMMBR." #define TEST_STR "Test TMMBR."
#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl #define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
@ -1031,4 +1033,3 @@ int _tmain(int argc, _TCHAR* argv[])
return 0; return 0;
} }