├── webrtc-lib └── webrtc │ ├── modules │ ├── audio_coding │ │ ├── codecs │ │ │ ├── OWNERS │ │ │ ├── g711 │ │ │ │ └── OWNERS │ │ │ ├── isac │ │ │ │ ├── empty.cc │ │ │ │ ├── fix │ │ │ │ │ ├── test │ │ │ │ │ │ └── QA │ │ │ │ │ │ │ ├── ChannelFiles.txt │ │ │ │ │ │ │ ├── InputFilesFew.txt │ │ │ │ │ │ │ ├── ListOfTestCases.xls │ │ │ │ │ │ │ ├── diffiSACPLC.txt │ │ │ │ │ │ │ ├── InputFiles.txt │ │ │ │ │ │ │ └── runiSACPLC.txt │ │ │ │ │ ├── source │ │ │ │ │ │ ├── audio_decoder_isacfix.cc │ │ │ │ │ │ └── audio_encoder_isacfix.cc │ │ │ │ │ └── include │ │ │ │ │ │ ├── audio_decoder_isacfix.h │ │ │ │ │ │ └── audio_encoder_isacfix.h │ │ │ │ ├── locked_bandwidth_info.cc │ │ │ │ ├── main │ │ │ │ │ ├── source │ │ │ │ │ │ ├── audio_decoder_isac.cc │ │ │ │ │ │ └── audio_encoder_isac.cc │ │ │ │ │ └── include │ │ │ │ │ │ ├── audio_decoder_isac.h │ │ │ │ │ │ └── audio_encoder_isac.h │ │ │ │ └── bandwidth_info.h │ │ │ ├── ilbc │ │ │ │ ├── test │ │ │ │ │ └── empty.cc │ │ │ │ └── index_conv_dec.h │ │ │ ├── audio_encoder.h │ │ │ ├── audio_decoder.h │ │ │ ├── pcm16b │ │ │ │ ├── pcm16b_common.h │ │ │ │ └── pcm16b_common.cc │ │ │ ├── builtin_audio_encoder_factory.h │ │ │ ├── builtin_audio_decoder_factory.h │ │ │ └── audio_format_conversion.h │ │ ├── neteq │ │ │ ├── tools │ │ │ │ ├── DEPS │ │ │ │ ├── audio_sink.cc │ │ │ │ └── packet_source.cc │ │ │ ├── post_decode_vad_unittest.cc │ │ │ ├── random_vector_unittest.cc │ │ │ ├── background_noise_unittest.cc │ │ │ ├── test │ │ │ │ └── NETEQTEST_DummyRTPpacket.h │ │ │ └── tick_timer.cc │ │ ├── DEPS │ │ ├── OWNERS │ │ ├── CMakeLists.txt │ │ ├── audio_network_adaptor │ │ │ └── controller.cc │ │ └── test │ │ │ └── ACMTest.h │ ├── video_coding │ │ ├── codecs │ │ │ ├── stereo │ │ │ │ └── OWNERS │ │ │ ├── h264 │ │ │ │ ├── OWNERS │ │ │ │ └── DEPS │ │ │ ├── test │ │ │ │ └── android_test_initializer.h │ │ │ └── interface │ │ │ │ └── video_codec_interface.h │ │ ├── OWNERS │ │ ├── DEPS │ │ ├── qp_parser.h │ │ └── utility │ │ │ └── vp9_uncompressed_header_parser.h │ ├── include │ │ └── DEPS │ ├── media_file │ │ ├── DEPS │ │ └── OWNERS │ ├── audio_processing │ │ ├── test │ │ │ ├── py_quality_assessment │ │ │ │ ├── apm_configs │ │ │ │ │ └── default.json │ │ │ │ ├── quality_assessment │ │ │ │ │ ├── apm_configs │ │ │ │ │ │ └── default.json │ │ │ │ │ ├── __init__.py │ │ │ │ │ └── results.css │ │ │ │ ├── output │ │ │ │ │ └── README.md │ │ │ │ └── OWNERS │ │ │ ├── conversational_speech │ │ │ │ ├── OWNERS │ │ │ │ └── config.cc │ │ │ └── android │ │ │ │ └── apmtest │ │ │ │ ├── res │ │ │ │ └── values │ │ │ │ │ └── strings.xml │ │ │ │ └── default.properties │ │ ├── OWNERS │ │ ├── DEPS │ │ ├── aec_dump │ │ │ └── mock_aec_dump.cc │ │ ├── aec3 │ │ │ ├── echo_path_variability.cc │ │ │ ├── downsampled_render_buffer.cc │ │ │ ├── echo_path_variability.h │ │ │ └── aec3_common.cc │ │ ├── include │ │ │ └── config.cc │ │ ├── agc │ │ │ └── utility.h │ │ ├── transient │ │ │ ├── test │ │ │ │ ├── readPCM.m │ │ │ │ ├── plotDetection.m │ │ │ │ └── readDetection.m │ │ │ └── common.h │ │ ├── level_controller │ │ │ └── level_controller_constants.h │ │ └── vad │ │ │ └── common.h │ ├── pacing │ │ ├── DEPS │ │ ├── OWNERS │ │ └── CMakeLists.txt │ ├── desktop_capture │ │ ├── DEPS │ │ ├── win │ │ │ ├── cursor_test_data │ │ │ │ ├── 1_24bpp.cur │ │ │ │ ├── 1_32bpp.cur │ │ │ │ ├── 1_8bpp.cur │ │ │ │ ├── 2_1bpp.cur │ │ │ │ ├── 2_32bpp.cur │ │ │ │ ├── 3_32bpp.cur │ │ │ │ └── 3_4bpp.cur │ │ │ ├── cursor.h │ │ │ └── cursor_unittest_resources.h │ │ ├── OWNERS │ │ ├── screen_capturer_null.cc │ │ ├── window_finder.cc │ │ ├── shared_memory.cc │ │ ├── mock_desktop_capturer_callback.cc │ │ ├── resolution_tracker.cc │ │ ├── screen_drawer.cc │ │ ├── screen_drawer_mac.cc │ │ └── test_utils.h │ ├── video_capture │ │ ├── DEPS │ │ ├── OWNERS │ │ ├── test │ │ │ └── video_capture_main_mac.mm │ │ └── external │ │ │ └── video_capture_external.cc │ ├── video_processing │ │ ├── DEPS │ │ └── OWNERS │ ├── bitrate_controller │ │ ├── DEPS │ │ ├── OWNERS │ │ └── CMakeLists.txt │ ├── congestion_controller │ │ ├── DEPS │ │ ├── OWNERS │ │ ├── CMakeLists.txt │ │ └── congestion_controller_unittests_helper.h │ ├── rtp_rtcp │ │ ├── DEPS │ │ ├── OWNERS │ │ ├── mocks │ │ │ └── mock_rtcp_rtt_stats.h │ │ └── source │ │ │ ├── video_codec_information.h │ │ │ └── rtcp_nack_stats.cc │ ├── OWNERS │ ├── remote_bitrate_estimator │ │ ├── DEPS │ │ ├── OWNERS │ │ └── CMakeLists.txt │ ├── utility │ │ ├── OWNERS │ │ ├── DEPS │ │ ├── CMakeLists.txt │ │ └── include │ │ │ └── audio_frame_operations.h │ ├── CMakeLists.txt │ ├── audio_mixer │ │ ├── OWNERS │ │ ├── DEPS │ │ └── output_rate_calculator.h │ └── audio_device │ │ ├── OWNERS │ │ ├── ios │ │ └── objc │ │ │ ├── RTCAudioSession.h │ │ │ └── RTCAudioSessionConfiguration.h │ │ ├── android │ │ └── ensure_initialized.h │ │ ├── DEPS │ │ └── audio_device_config.h │ ├── system_wrappers │ ├── DEPS │ ├── OWNERS │ ├── source │ │ ├── cpu_features_android.c │ │ ├── rw_lock.cc │ │ └── sleep.cc │ ├── CMakeLists.txt │ └── include │ │ ├── cpu_info.h │ │ └── sleep.h │ ├── rtc_base │ ├── java │ │ └── src │ │ │ └── org │ │ │ └── webrtc │ │ │ └── OWNERS │ ├── DEPS │ ├── OWNERS │ ├── atomicops_unittest.cc │ ├── noop.cc │ ├── noop.mm │ ├── sigslot.cc │ ├── Dummy.java │ ├── win32socketinit.h │ ├── messagehandler.cc │ ├── asyncresolverinterface.cc │ ├── openssl.h │ ├── memory_usage_unittest.cc │ ├── sha1.h │ ├── logging_mac.mm │ ├── macutils.h │ ├── proxyinfo.cc │ ├── memory_usage.h │ ├── gunit_prod.h │ ├── asyncpacketsocket.cc │ ├── compile_assert_c.h │ ├── md5digest.cc │ ├── sha1digest.cc │ ├── ratelimiter.cc │ ├── cpu_time.h │ ├── task_queue_posix.h │ ├── onetimeevent_unittest.cc │ └── platform_thread_types.h │ ├── test │ ├── OWNERS │ ├── CMakeLists.txt │ ├── mock_audio_decoder.cc │ ├── run_test.cc │ ├── gmock.h │ ├── run_loop.h │ ├── run_loop.cc │ ├── run_test.h │ ├── null_platform_renderer.cc │ ├── constants.h │ ├── null_transport.cc │ ├── constants.cc │ ├── fake_videorenderer.h │ ├── gtest.h │ ├── mock_transport.h │ ├── null_transport.h │ └── statistics.h │ ├── api │ ├── audio_codecs │ │ ├── OWNERS │ │ └── g722 │ │ │ └── audio_encoder_g722_config.h │ ├── video │ │ ├── OWNERS │ │ └── video_rotation.h │ ├── stats │ │ └── OWNERS │ ├── CMakeLists.txt │ ├── OWNERS │ ├── ortc │ │ ├── mediadescription.cc │ │ ├── sessiondescription.cc │ │ ├── sessiondescription_unittest.cc │ │ └── mediadescription_unittest.cc │ ├── mediastreaminterface.cc │ ├── rtpsender.h │ ├── webrtcsdp.h │ ├── umametrics.cc │ ├── datachannel.h │ ├── mediastream.h │ ├── mediastreamtrack.h │ ├── streamcollection.h │ ├── videotracksource.h │ ├── optional.cc │ ├── mediatypes.h │ └── video_codecs │ │ └── BUILD.gn │ ├── logging │ ├── OWNERS │ └── rtc_event_log │ │ ├── DEPS │ │ ├── events │ │ ├── rtc_event_logging_started.cc │ │ ├── rtc_event_logging_stopped.cc │ │ ├── rtc_event_audio_playout.cc │ │ ├── rtc_event_probe_result_success.cc │ │ ├── rtc_event_probe_result_failure.cc │ │ ├── rtc_event_logging_started.h │ │ └── rtc_event_logging_stopped.h │ │ ├── encoder │ │ └── rtc_event_log_encoder.h │ │ └── rtc_event_log_factory.cc │ ├── audio │ ├── OWNERS │ ├── DEPS │ └── conversion.h │ ├── base │ └── libevent │ │ └── ReadMe.md │ └── testfield.cc ├── multipathvideo ├── model │ ├── forsometest.h │ ├── path.h │ ├── path.cc │ ├── recvpath.cc │ ├── recvpath.h │ ├── forsomtest.cc │ └── mpconstants.h └── wscript ├── webrtc-results ├── location.png ├── webrtc_1_bw.png ├── webrtc_4_bw.png ├── webrtc_1_delay.png ├── webrtc_4_delay.png ├── pro_loss.py └── bw_plot.sh ├── razor-ns3 └── model │ ├── sim_receiver.h │ ├── razor │ ├── cc │ │ ├── razor_api.h │ │ ├── razor_callback.h │ │ ├── receiver_congestion_controller.c │ │ ├── receiver_congestion_controller.h │ │ ├── sender_congestion_controller.c │ │ ├── sender_congestion_controller.h │ │ ├── razor_log.c │ │ └── razor_log.h │ ├── common │ │ ├── cf_skiplist.h │ │ ├── cf_unwrapper.h │ │ ├── cf_crc32.h │ │ ├── cf_hex.h │ │ └── cf_list.h │ ├── pacing │ │ ├── pace_sender.c │ │ ├── pace_sender.h │ │ ├── pacer_queue.c │ │ ├── pacer_queue.h │ │ ├── alr_detector.c │ │ ├── alr_detector.h │ │ ├── interval_budget.c │ │ └── interval_budget.h │ ├── estimator │ │ ├── rate_stat.c │ │ ├── rate_stat.h │ │ ├── trendline.c │ │ ├── trendline.h │ │ ├── cc_loss_stat.c │ │ ├── cc_loss_stat.h │ │ ├── inter_arrival.c │ │ ├── inter_arrival.h │ │ ├── kalman_filter.c │ │ ├── kalman_filter.h │ │ ├── delay_base_bwe.c │ │ ├── delay_base_bwe.h │ │ ├── sender_history.c │ │ ├── aimd_rate_control.c │ │ ├── bitrate_controller.c │ │ ├── bitrate_controller.h │ │ ├── estimator_common.c │ │ ├── estimator_common.h │ │ ├── feedback_adapter.c │ │ ├── feedback_adapter.h │ │ ├── overuse_detector.c │ │ ├── overuse_detector.h │ │ ├── ack_bitrate_estimator.c │ │ ├── ack_bitrate_estimator.h │ │ ├── remote_estimator_proxy.c │ │ ├── remote_estimator_proxy.h │ │ ├── remote_bitrate_estimator.c │ │ ├── remote_bitrate_estimator.h │ │ ├── sender_bandwidth_estimator.c │ │ ├── sender_bandwidth_estimator.h │ │ └── sender_history.h │ ├── sim_transport │ │ ├── ReadMe.txt │ │ ├── sim_proto.c │ │ ├── sim_proto.h │ │ ├── sim_limiter.c │ │ ├── sim_sender.c │ │ ├── sim_sender.h │ │ ├── sim_session.c │ │ ├── sim_session.h │ │ ├── sim_external.c │ │ ├── sim_external.h │ │ ├── sim_internal.h │ │ ├── sim_receiver.c │ │ └── sim_receiver.h │ └── .gitignore │ ├── c_call_cplus.cc │ ├── c_call_cplus.h │ ├── sim_constants.h │ ├── gcc_sender.h │ └── razortrace.h ├── razor-example ├── gcc_razor_0remb.png ├── gcc_razor_1remb.png └── wscript ├── webrtc-ns3 └── model │ ├── webrtcreceiver.cc │ ├── webrtcconstants.h │ ├── target.h │ ├── webrtclog.cc │ ├── webrtclog.h │ ├── webrtcrecord.h │ ├── simulationclock.h │ ├── gclock.h │ ├── simulationclock.cc │ └── webrtctrace.h ├── nada └── ns3-rmcat │ └── model │ └── syncodecs │ ├── .gitignore │ └── README.md ├── mytrace └── model │ └── mytrace.h ├── scratch └── wscript └── scream ├── wscript └── model └── videoenc.h /webrtc-lib/webrtc/modules/audio_coding/codecs/OWNERS: -------------------------------------------------------------------------------- 1 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/g711/OWNERS: -------------------------------------------------------------------------------- 1 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/empty.cc: -------------------------------------------------------------------------------- 1 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/ilbc/test/empty.cc: -------------------------------------------------------------------------------- 1 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/system_wrappers/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | ] 3 | 4 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/java/src/org/webrtc/OWNERS: -------------------------------------------------------------------------------- 1 | magjed@webrtc.org 2 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/OWNERS: -------------------------------------------------------------------------------- 1 | phoglund@webrtc.org 2 | stefan@webrtc.org 3 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_coding/codecs/stereo/OWNERS: -------------------------------------------------------------------------------- 1 | emircan@webrtc.org 2 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/audio_codecs/OWNERS: -------------------------------------------------------------------------------- 1 | kwiberg@webrtc.org 2 | ossu@webrtc.org 3 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/include/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+common_video", 3 | ] 4 | -------------------------------------------------------------------------------- /multipathvideo/model/forsometest.h: -------------------------------------------------------------------------------- 1 | #ifndef FOR_SOME_TEST_H 2 | #define FOR_SOME_TEST_H 3 | #endif 4 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/OWNERS: -------------------------------------------------------------------------------- 1 | skvlad@webrtc.org 2 | stefan@webrtc.org 3 | terelius@webrtc.org 4 | 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/webrtc-lib/webrtc/modules/audio_processing/test/android/apmtest/res/values/strings.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | apmtest 4 | 5 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ListOfTestCases.xls: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/SoonyangZhang/rmcat-ns3/HEAD/webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ListOfTestCases.xls -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/OWNERS: -------------------------------------------------------------------------------- 1 | # These are for the common case of adding or renaming files. If you're doing 2 | # structural changes, please get a review from a reviewer in this file. 3 | per-file *.gn=* 4 | per-file *.gni=* 5 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+call", 3 | "+common_audio", 4 | "+logging/rtc_event_log", 5 | "+audio_coding/neteq/neteq_unittest.pb.h", # Different path. 6 | "+system_wrappers", 7 | ] 8 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/test/py_quality_assessment/OWNERS: -------------------------------------------------------------------------------- 1 | aleloi@webrtc.org 2 | alessiob@webrtc.org 3 | henrik.lundin@webrtc.org 4 | ivoc@webrtc.org 5 | peah@webrtc.org 6 | 7 | per-file *.gn=* 8 | per-file *.gni=* 9 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+call", 3 | "+modules/audio_coding/audio_network_adaptor", 4 | "+modules/remote_bitrate_estimator/include", 5 | "+modules/rtp_rtcp", 6 | "+system_wrappers", 7 | ] 8 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/remote_bitrate_estimator/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+logging/rtc_event_log", 3 | "+system_wrappers", 4 | ] 5 | 6 | specific_include_rules = { 7 | "nada\.h": [ 8 | "+voice_engine", 9 | ], 10 | } 11 | -------------------------------------------------------------------------------- /razor-ns3/model/c_call_cplus.h: -------------------------------------------------------------------------------- 1 | #ifndef C_CALL_CPLUS_H 2 | #define C_CALL_CPLUS_H 3 | #include 4 | 5 | #ifdef __cplusplus 6 | extern "C" { 7 | #endif 8 | int64_t get_clock_time(); 9 | #ifdef __cplusplus 10 | } 11 | #endif 12 | #endif 13 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/utility/OWNERS: -------------------------------------------------------------------------------- 1 | perkj@webrtc.org 2 | 3 | # These are for the common case of adding or renaming files. If you're doing 4 | # structural changes, please get a review from a reviewer in this file. 5 | per-file *.gn=* 6 | per-file *.gni=* 7 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/utility/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | # TODO(aleloi): remove when clients update. See 3 | # bugs.webrtc.org/6548. 4 | "+audio/utility/audio_frame_operations.h", 5 | "+common_audio", 6 | "+common_video", 7 | "+system_wrappers", 8 | ] 9 | -------------------------------------------------------------------------------- /webrtc-ns3/model/target.h: -------------------------------------------------------------------------------- 1 | #ifndef TARGET_H 2 | #define TARGET_H 3 | #include 4 | namespace ns3 5 | { 6 | class Target 7 | { 8 | public: 9 | virtual ~Target(){} 10 | virtual void TargetReceive(uint32_t len){} 11 | }; 12 | } 13 | #endif 14 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/stats/OWNERS: -------------------------------------------------------------------------------- 1 | hbos@webrtc.org 2 | hta@webrtc.org 3 | 4 | # These are for the common case of adding or renaming files. If you're doing 5 | # structural changes, please get a review from a reviewer in this file. 6 | per-file *.gn=* 7 | per-file *.gni=* 8 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/audio/OWNERS: -------------------------------------------------------------------------------- 1 | solenberg@webrtc.org 2 | ossu@webrtc.org 3 | 4 | # These are for the common case of adding or renaming files. If you're doing 5 | # structural changes, please get a review from a reviewer in this file. 6 | per-file *.gn=* 7 | per-file *.gni=* 8 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/base/libevent/ReadMe.md: -------------------------------------------------------------------------------- 1 | libevent should install here. 2 | And change the CMakeLists.txt under rtc_base: 3 | include_directories(/the_path_to_libevent_header/libevent/include) 4 | FIND_LIBRARY(COMM_LIB event "/the_path_to_libevent_lib/libevent/lib" NO_DEFAULT_PATH) 5 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/CMakeLists.txt: -------------------------------------------------------------------------------- 1 | add_subdirectory(pacing) 2 | add_subdirectory(utility) 3 | add_subdirectory(congestion_controller) 4 | add_subdirectory(bitrate_controller) 5 | add_subdirectory(remote_bitrate_estimator) 6 | add_subdirectory(rtp_rtcp) 7 | add_subdirectory(audio_coding) 8 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_mixer/OWNERS: -------------------------------------------------------------------------------- 1 | aleloi@webrtc.org 2 | henrik.lundin@webrtc.org 3 | 4 | # These are for the common case of adding or renaming files. If you're doing 5 | # structural changes, please get a review from a reviewer in this file. 6 | per-file *.gn=* 7 | per-file *.gni=* 8 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_processing/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | marpan@webrtc.org 3 | 4 | # These are for the common case of adding or renaming files. If you're doing 5 | # structural changes, please get a review from a reviewer in this file. 6 | per-file *.gn=* 7 | per-file *.gni=* 8 | -------------------------------------------------------------------------------- /razor-ns3/model/sim_constants.h: -------------------------------------------------------------------------------- 1 | #ifndef SIM_CONSTANTS_H 2 | #define SIM_CONSTANTS_H 3 | 4 | #define SIM_MIN_BITRATE 80000 /*10KB*/ 5 | #define SIM_MAX_BITRATE 16000000 /*2MB*/ 6 | #define SIM_START_BITRATE 800000 /*100KB*/ 7 | 8 | 9 | 10 | 11 | #endif /* SIM_CONSTANTS_H */ 12 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/media_file/OWNERS: -------------------------------------------------------------------------------- 1 | mflodman@webrtc.org 2 | perkj@webrtc.org 3 | niklas.enbom@webrtc.org 4 | 5 | # These are for the common case of adding or renaming files. If you're doing 6 | # structural changes, please get a review from a reviewer in this file. 7 | per-file *.gn=* 8 | per-file *.gni=* 9 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/bitrate_controller/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | terelius@webrtc.org 3 | mflodman@webrtc.org 4 | 5 | # These are for the common case of adding or renaming files. If you're doing 6 | # structural changes, please get a review from a reviewer in this file. 7 | per-file *.gn=* 8 | per-file *.gni=* 9 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/utility/CMakeLists.txt: -------------------------------------------------------------------------------- 1 | # c++11 required 2 | set(CMAKE_CXX_STANDARD 11) 3 | set(CMAKE_CXX_STANDARD_REQUIRED ON) 4 | include_directories(../../../) 5 | include_directories(../../) 6 | set(utility_FILES 7 | source/process_thread_impl.cc 8 | ) 9 | add_library(utility STATIC ${utility_FILES}) 10 | -------------------------------------------------------------------------------- /razor-example/wscript: -------------------------------------------------------------------------------- 1 | ## -*- Mode: python; py-indent-offset: 4; indent-tabs-mode: nil; coding: utf-8; -*- 2 | 3 | def build(bld): 4 | obj = bld.create_ns3_program('gcc_tcp', 5 | ['core','razor-ns3','internet','network','point-to-point', 'applications','traffic-control']) 6 | obj.source = 'gcc_tcp.cc' 7 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/pacing/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | mflodman@webrtc.org 3 | asapersson@webrtc.org 4 | philipel@webrtc.org 5 | 6 | # These are for the common case of adding or renaming files. If you're doing 7 | # structural changes, please get a review from a reviewer in this file. 8 | per-file *.gn=* 9 | per-file *.gni=* 10 | -------------------------------------------------------------------------------- /webrtc-ns3/model/webrtclog.cc: -------------------------------------------------------------------------------- 1 | #include "webrtclog.h" 2 | #include 3 | namespace ns3 4 | { 5 | void LogSinkConsole::OnLogMessage(const std::string &message) 6 | { 7 | std::cout< 13 | #include 14 | 15 | uint32_t crc32(uint32_t crc, const void *buf, size_t size); 16 | 17 | #endif /* COMMON_CRC32_H_ */ 18 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_coding/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+vpx", 3 | "+call", 4 | "+common_video", 5 | "+system_wrappers", 6 | "+rtc_tools", 7 | ] 8 | 9 | specific_include_rules = { 10 | "android_test_initializer\.cc": [ 11 | "+base/android", 12 | "+sdk", 13 | ], 14 | "(.*test\.cc|.*test\.h|.*test\.mm)": [ 15 | "+media/base", 16 | "+media/engine", 17 | "+sdk", 18 | ], 19 | } 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+base/third_party/libevent", 3 | "+json", 4 | "+third_party/jsoncpp", 5 | "+system_wrappers", 6 | ] 7 | 8 | specific_include_rules = { 9 | "gunit_prod.h": [ 10 | "+gtest", 11 | "+testing/base/gunit_prod.h", 12 | ], 13 | "protobuf_utils.h": [ 14 | "+third_party/protobuf", 15 | ], 16 | "gunit\.h": [ 17 | "+testing/base/public/gunit.h" 18 | ], 19 | } 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/bitrate_controller/CMakeLists.txt: -------------------------------------------------------------------------------- 1 | # c++11 required 2 | set(CMAKE_CXX_STANDARD 11) 3 | set(CMAKE_CXX_STANDARD_REQUIRED ON) 4 | include_directories(../../../) 5 | include_directories(../../) 6 | set(bitrate_controller_FILES 7 | bitrate_controller_impl.cc 8 | send_side_bandwidth_estimation.cc 9 | ) 10 | add_library(bitrate_controller SHARED ${bitrate_controller_FILES}) 11 | target_link_libraries(bitrate_controller system_wrappers rtc_base pacing) 12 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/test/android/apmtest/default.properties: -------------------------------------------------------------------------------- 1 | # This file is automatically generated by Android Tools. 2 | # Do not modify this file -- YOUR CHANGES WILL BE ERASED! 3 | # 4 | # This file must be checked in Version Control Systems. 5 | # 6 | # To customize properties used by the Ant build system use, 7 | # "build.properties", and override values to adapt the script to your 8 | # project structure. 9 | 10 | # Project target. 11 | target=android-9 12 | -------------------------------------------------------------------------------- /razor-ns3/model/razor/common/cf_hex.h: -------------------------------------------------------------------------------- 1 | /*- 2 | * Copyright (c) 2017-2018 wenba, Inc. 3 | * All rights reserved. 4 | * 5 | * See the file LICENSE for redistribution information. 6 | */ 7 | 8 | #ifndef __CF_HEX_H_ 9 | #define __CF_HEX_H_ 10 | 11 | #include 12 | 13 | char* wb_bin2asc(const uint8_t *in, int32_t in_size, char* out, int32_t out_size); 14 | 15 | void wb_asc2bin(const char* in, int32_t in_size, uint8_t* out, int32_t out_size, int32_t* out_length); 16 | 17 | #endif 18 | 19 | 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 2 | # 3 | # Use of this source code is governed by a BSD-style license 4 | # that can be found in the LICENSE file in the root of the source 5 | # tree. An additional intellectual property rights grant can be found 6 | # in the file PATENTS. All contributing project authors may 7 | # be found in the AUTHORS file in the root of the source tree. 8 | -------------------------------------------------------------------------------- /mytrace/model/mytrace.h: -------------------------------------------------------------------------------- 1 | #ifndef MY_TRACE_H 2 | #define MY_TRACE_H 3 | #include 4 | #include 5 | #include 6 | namespace ns3 7 | { 8 | class MyTracer 9 | { 10 | public: 11 | MyTracer(); 12 | ~MyTracer(); 13 | void OpenTraceFile(std::string filename); 14 | void CloseTraceFile(); 15 | void RateTrace(float rate); 16 | void DelayTrace(uint64_t delay); 17 | private: 18 | std::fstream m_traceRateFile;// 19 | std::fstream m_traceDelayFile; 20 | }; 21 | } 22 | #endif -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/pacing/CMakeLists.txt: -------------------------------------------------------------------------------- 1 | # c++11 required 2 | set(CMAKE_CXX_STANDARD 11) 3 | set(CMAKE_CXX_STANDARD_REQUIRED ON) 4 | include_directories(../../../) 5 | include_directories(../../) 6 | set(pacing_FILES 7 | alr_detector.cc 8 | bitrate_prober.cc 9 | interval_budget.cc 10 | paced_sender.cc 11 | packet_queue.cc 12 | packet_queue2.cc 13 | #packet_queue_interface.cc 14 | packet_router.cc 15 | ) 16 | add_library(pacing SHARED ${pacing_FILES}) 17 | target_link_libraries(pacing logging rtc_base ) 18 | -------------------------------------------------------------------------------- /webrtc-ns3/model/webrtcrecord.h: -------------------------------------------------------------------------------- 1 | #pragma once 2 | #include 3 | namespace ns3{ 4 | class WebrtcSeqRecord{ 5 | public: 6 | WebrtcSeqRecord(); 7 | ~WebrtcSeqRecord(){} 8 | bool IsInitialized(); 9 | bool ShoudGenerateReport(uint32_t thresh); 10 | void Reset(); 11 | uint32_t TotalNumber(); 12 | uint32_t PacketLost(); 13 | uint8_t FractionLoss(); 14 | void OnNewSeq(uint32_t seq); 15 | uint32_t m_lastSeq; 16 | uint32_t m_highSeq; 17 | int64_t m_recvCount{0}; 18 | }; 19 | } 20 | -------------------------------------------------------------------------------- /razor-ns3/model/razor/.gitignore: -------------------------------------------------------------------------------- 1 | # Prerequisites 2 | *.d 3 | 4 | # Object files 5 | *.o 6 | *.ko 7 | *.obj 8 | *.elf 9 | 10 | # Linker output 11 | *.ilk 12 | *.map 13 | *.exp 14 | 15 | # Precompiled Headers 16 | *.gch 17 | *.pch 18 | 19 | # Executables 20 | *.exe 21 | *.out 22 | *.app 23 | *.i*86 24 | *.x86_64 25 | *.hex 26 | 27 | # Debug files 28 | *.dSYM/ 29 | *.su 30 | *.idb 31 | *.pdb 32 | 33 | # Kernel Module Compile Results 34 | *.mod* 35 | *.cmd 36 | .tmp_versions/ 37 | modules.order 38 | Module.symvers 39 | Mkfile.old 40 | dkms.conf 41 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/ortc/mediadescription.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "api/ortc/mediadescription.h" 12 | 13 | namespace webrtc {} 14 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/ortc/sessiondescription.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "api/ortc/sessiondescription.h" 12 | 13 | namespace webrtc {} 14 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h" 12 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/OWNERS: -------------------------------------------------------------------------------- 1 | henrika@webrtc.org 2 | henrikg@webrtc.org 3 | hta@webrtc.org 4 | juberti@webrtc.org 5 | mflodman@webrtc.org 6 | perkj@webrtc.org 7 | pthatcher@webrtc.org 8 | sergeyu@chromium.org 9 | tommi@webrtc.org 10 | deadbeef@webrtc.org 11 | kwiberg@webrtc.org 12 | 13 | # These are for the common case of adding or renaming files. If you're doing 14 | # structural changes, please get a review from a reviewer in this file. 15 | per-file *.gn=* 16 | per-file *.gni=* 17 | 18 | per-file rate_statistics*=sprang@webrtc.org 19 | per-file rate_statistics*=stefan@webrtc.org 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/atomicops_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2011 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // TODO(pbos): Move AtomicOps tests to here from 12 | // webrtc/rtc_base/criticalsection_unittest.cc. 13 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h" 12 | -------------------------------------------------------------------------------- /webrtc-ns3/model/simulationclock.h: -------------------------------------------------------------------------------- 1 | #ifndef SIMULATION_CLOCK_H 2 | #define SIMULATION_CLOCK_H 3 | #include "ns3/simulator.h" 4 | #include "ns3/nstime.h" 5 | #include "system_wrappers/include/clock.h" 6 | namespace ns3 7 | { 8 | 9 | class SimulationClock:public webrtc::Clock 10 | { 11 | public: 12 | SimulationClock(){} 13 | ~SimulationClock(){} 14 | int64_t TimeInMilliseconds() const override; 15 | int64_t TimeInMicroseconds() const override; 16 | webrtc::NtpTime CurrentNtpTime() const override; 17 | int64_t CurrentNtpInMilliseconds() const override; 18 | }; 19 | } 20 | #endif 21 | -------------------------------------------------------------------------------- /scratch/wscript: -------------------------------------------------------------------------------- 1 | ## -*- Mode: python; py-indent-offset: 4; indent-tabs-mode: nil; coding: utf-8; -*- 2 | 3 | def build(bld): 4 | #obj = bld.create_ns3_program('webrtc-tcp' ['application','core','internet','network','webrtc-ns3','mytrace','traffic-control']) 5 | #obj.source = 'webrtc-tcp.cc' 6 | #obj = bld.create_ns3_program('nada' ['application','core','internet','network','ns3-rmcat','mytrace','traffic-control']) 7 | #obj.source = 'nada.cc' 8 | obj = bld.create_ns3_program('screamex' ['application','core','internet','network','scream','mytrace','traffic-control']) 9 | obj.source = 'screamex.cc' 10 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/system_wrappers/source/cpu_features_android.c: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include 12 | 13 | uint64_t WebRtc_GetCPUFeaturesARM(void) { 14 | return android_getCpuFeatures(); 15 | } 16 | -------------------------------------------------------------------------------- /nada/ns3-rmcat/model/syncodecs/README.md: -------------------------------------------------------------------------------- 1 | # syncodecs 2 | Synthetic codecs for evaluation of RMCAT work 3 | 4 | Syncodecs are a set of C++ classes implementing adaptive synthetic video codecs. "Synthetic" means that they do not encode real video; they provide a sequence of packets/frames whose size/frequency adapt to the configured target sending rate. This configured target sending rate can be changed over time. The synthetic codecs can be used both in simulation environments (ns2, ns3), and in physical testbeds running real endpoints. 5 | 6 | For further information, please look at the extensive Doxygen documentation and examples in the ".h" files. 7 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/diffiSACPLC.txt: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | (set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character 3 | 4 | LOGFILE=logplc.txt 5 | echo "START PLC TEST" > $LOGFILE 6 | 7 | OUTDIR1=../dataqaplc_0 8 | OUTDIR2=../dataqaplc_1 9 | 10 | diff $OUTDIR1/outplc1.pcm $OUTDIR2/outplc1.pcm 11 | diff $OUTDIR1/outplc2.pcm $OUTDIR2/outplc2.pcm 12 | diff $OUTDIR1/outplc3.pcm $OUTDIR2/outplc3.pcm 13 | diff $OUTDIR1/outplc4.pcm $OUTDIR2/outplc4.pcm 14 | diff $OUTDIR1/outplc5.pcm $OUTDIR2/outplc5.pcm 15 | diff $OUTDIR1/outplc6.pcm $OUTDIR2/outplc6.pcm 16 | 17 | echo DONE! 18 | 19 | 20 | 21 | -------------------------------------------------------------------------------- /webrtc-ns3/model/gclock.h: -------------------------------------------------------------------------------- 1 | #ifndef G_CLOCK_H 2 | #define G_CLOCK_H 3 | #include"rtc_base/timeutils.h" 4 | #include "ns3/nstime.h" 5 | #include "ns3/simulator.h" 6 | namespace ns3 7 | { 8 | class GClock:public rtc::ClockInterface 9 | { 10 | public: 11 | GClock(){} 12 | ~GClock(){} 13 | int64_t TimeNanos() const override 14 | { 15 | return Simulator::Now().GetNanoSeconds(); 16 | } 17 | }; 18 | static GClock f_clock; 19 | void SetClockForWebrtc() 20 | { 21 | rtc::SetClockForTesting(&f_clock); 22 | } 23 | int64_t GetMilliSecondsFromWebrtc()//just for testing; 24 | { 25 | return rtc::TimeMillis(); 26 | } 27 | } 28 | #endif 29 | -------------------------------------------------------------------------------- /razor-ns3/model/razor/cc/razor_log.c: -------------------------------------------------------------------------------- 1 | /*- 2 | * Copyright (c) 2017-2018 wenba, Inc. 3 | * All rights reserved. 4 | * 5 | * See the file LICENSE for redistribution information. 6 | */ 7 | 8 | #include "razor_log.h" 9 | #include "razor_api.h" 10 | #include "cf_platform.h" 11 | 12 | static razor_log_func g_log_cb = NULL; 13 | 14 | void ex_razor_log(int level, const char* file, int line, const char *fmt, ...) 15 | { 16 | va_list vl; 17 | if (g_log_cb != NULL) { 18 | va_start(vl, fmt); 19 | g_log_cb(level, file, line, fmt, vl); 20 | va_end(vl); 21 | } 22 | } 23 | 24 | void razor_setup_log(razor_log_func log_cb) 25 | { 26 | g_log_cb = log_cb; 27 | } 28 | 29 | 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_device/android/ensure_initialized.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | namespace webrtc { 12 | namespace audiodevicemodule { 13 | 14 | void EnsureInitialized(); 15 | 16 | } // namespace audiodevicemodule 17 | } // namespace webrtc 18 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/noop.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // This file is only needed to make ninja happy on some platforms. 12 | // On some platforms it is not possible to link an rtc_static_library 13 | // without any source file listed in the GN target. 14 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/noop.mm: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // This file is only needed to make ninja happy on some platforms. 12 | // On some platforms it is not possible to link an rtc_static_library 13 | // without any source file listed in the GN target. 14 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/sigslot.cc: -------------------------------------------------------------------------------- 1 | // sigslot.h: Signal/Slot classes 2 | // 3 | // Written by Sarah Thompson (sarah@telergy.com) 2002. 4 | // 5 | // License: Public domain. You are free to use this code however you like, with 6 | // the proviso that the author takes on no responsibility or liability for any 7 | // use. 8 | 9 | #include "rtc_base/sigslot.h" 10 | 11 | namespace sigslot { 12 | 13 | #ifdef _SIGSLOT_HAS_POSIX_THREADS 14 | 15 | pthread_mutex_t* multi_threaded_global::get_mutex() { 16 | static pthread_mutex_t g_mutex = PTHREAD_MUTEX_INITIALIZER; 17 | return &g_mutex; 18 | } 19 | 20 | #endif // _SIGSLOT_HAS_POSIX_THREADS 21 | 22 | } // namespace sigslot 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/InputFiles.txt: -------------------------------------------------------------------------------- 1 | DTMF_16kHz_long.pcm 2 | DTMF_16kHz_short.pcm 3 | F00.INP 4 | F01.INP 5 | F02.INP 6 | F03.INP 7 | F04.INP 8 | F05.INP 9 | F06.INP 10 | longtest.pcm 11 | ltest_speech_clean.pcm 12 | ltest_music.pcm 13 | ltest_speech_noisy.pcm 14 | misc2.pcm 15 | purenb.pcm 16 | sawsweep_380_60.pcm 17 | sinesweep.pcm 18 | sinesweep_half.pcm 19 | speechmusic.pcm 20 | speechmusic_nb.pcm 21 | speechoffice0dB.pcm 22 | speech_and_misc_NB.pcm 23 | speech_and_misc_WB.pcm 24 | testM4.pcm 25 | testM4D_rev.pcm 26 | testM4D.pcm 27 | testfile.pcm 28 | tone_cisco.pcm 29 | tone_cisco_long.pcm 30 | wb_contspeech.pcm 31 | wb_speech_office25db.pcm -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/Dummy.java: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /** 12 | * This class only exists as glue in a transition. 13 | * TODO(kjellander): Remove. 14 | * See https://bugs.webrtc.org/7634 for more details. 15 | */ 16 | class Dummy { 17 | Dummy() { 18 | } 19 | } 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/remote_bitrate_estimator/CMakeLists.txt: -------------------------------------------------------------------------------- 1 | # c++11 required 2 | set(CMAKE_CXX_STANDARD 11) 3 | set(CMAKE_CXX_STANDARD_REQUIRED ON) 4 | include_directories(../../../) 5 | include_directories(../../) 6 | set(remote_bitrate_estimator_FILES 7 | aimd_rate_control.cc 8 | bwe_defines.cc 9 | inter_arrival.cc 10 | overuse_detector.cc 11 | overuse_estimator.cc 12 | remote_bitrate_estimator_abs_send_time.cc 13 | remote_bitrate_estimator_single_stream.cc 14 | remote_estimator_proxy.cc 15 | send_time_history.cc 16 | ) 17 | add_library(remote_bitrate_estimator SHARED ${remote_bitrate_estimator_FILES}) 18 | target_link_libraries(remote_bitrate_estimator rtp_rtcp_format system_wrappers rtc_base pacing) 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/win32socketinit.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2009 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_WIN32SOCKETINIT_H_ 12 | #define RTC_BASE_WIN32SOCKETINIT_H_ 13 | 14 | namespace rtc { 15 | 16 | void EnsureWinsockInit(); 17 | 18 | } // namespace rtc 19 | 20 | #endif // RTC_BASE_WIN32SOCKETINIT_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_device/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+common_audio", 3 | "+system_wrappers", 4 | ] 5 | 6 | specific_include_rules = { 7 | "ensure_initialized\.cc": [ 8 | "+base/android", 9 | ], 10 | "audio_device_ios\.h": [ 11 | "+sdk/objc", 12 | ], 13 | "audio_device_ios\.mm": [ 14 | "+sdk/objc", 15 | ], 16 | "audio_device_unittest_ios\.mm": [ 17 | "+sdk/objc", 18 | ], 19 | "RTCAudioSession\.h": [ 20 | "+sdk/objc", 21 | ], 22 | "RTCAudioSessionConfiguration\.h": [ 23 | "+sdk/objc", 24 | ], 25 | "RTCAudioSessionDelegateAdapter\.h": [ 26 | "+sdk/objc", 27 | ], 28 | "voice_processing_audio_unit\.mm": [ 29 | "+sdk/objc", 30 | ], 31 | } 32 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/messagehandler.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2004 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/messagehandler.h" 12 | #include "rtc_base/messagequeue.h" 13 | 14 | namespace rtc { 15 | 16 | MessageHandler::~MessageHandler() { 17 | MessageQueueManager::Clear(this); 18 | } 19 | 20 | } // namespace rtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/mock_audio_decoder.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "test/mock_audio_decoder.h" 12 | 13 | namespace webrtc { 14 | 15 | MockAudioDecoder::MockAudioDecoder() = default; 16 | MockAudioDecoder::~MockAudioDecoder() { 17 | Die(); 18 | } 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/run_test.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "test/run_test.h" 12 | 13 | #include 14 | 15 | namespace webrtc { 16 | namespace test { 17 | 18 | void RunTest(void(*test)()) { 19 | (*test)(); 20 | } 21 | 22 | } // namespace test 23 | } // namespace webrtc 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/mediastreaminterface.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "api/mediastreaminterface.h" 12 | 13 | namespace webrtc { 14 | 15 | const char MediaStreamTrackInterface::kVideoKind[] = "video"; 16 | const char MediaStreamTrackInterface::kAudioKind[] = "audio"; 17 | 18 | } // namespace webrtc 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/gmock.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef TEST_GMOCK_H_ 12 | #define TEST_GMOCK_H_ 13 | 14 | #include "rtc_base/ignore_wundef.h" 15 | 16 | RTC_PUSH_IGNORING_WUNDEF() 17 | #include "testing/gmock/include/gmock/gmock.h" 18 | RTC_POP_IGNORING_WUNDEF() 19 | 20 | #endif // TEST_GMOCK_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #include "modules/audio_processing/aec_dump/mock_aec_dump.h" 11 | 12 | namespace webrtc { 13 | 14 | namespace test { 15 | 16 | MockAecDump::MockAecDump() = default; 17 | MockAecDump::~MockAecDump() = default; 18 | } 19 | } 20 | -------------------------------------------------------------------------------- /scream/wscript: -------------------------------------------------------------------------------- 1 | # -*- Mode: python; py-indent-offset: 4; indent-tabs-mode: nil; coding: utf-8; -*- 2 | def build(bld): 3 | module = bld.create_ns3_module('scream', ['network']) 4 | module.source = [ 5 | 'model/scream-header.cc', 6 | 'model/screamsender.cc', 7 | 'model/screamreceiver.cc', 8 | 'model/rtpqueue.cc', 9 | 'model/screamRx.cc', 10 | 'model/screamTx.cc', 11 | 'model/videoenc.cc', 12 | ] 13 | headers = bld(features='ns3header') 14 | headers.module = 'scream' 15 | headers.source = [ 16 | 'model/scream-header.h', 17 | 'model/screamsender.h', 18 | 'model/screamreceiver.h', 19 | 'model/rtpqueue.h', 20 | 'model/screamRx.h', 21 | 'model/screamTx.h', 22 | 'model/videoenc.h', 23 | ] 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/asyncresolverinterface.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/asyncresolverinterface.h" 12 | 13 | namespace rtc { 14 | 15 | AsyncResolverInterface::AsyncResolverInterface() { 16 | } 17 | 18 | AsyncResolverInterface::~AsyncResolverInterface() = default; 19 | 20 | }; // namespace rtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_capture/test/video_capture_main_mac.mm: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "test/gtest.h" 12 | #include "test/testsupport/mac/run_threaded_main_mac.h" 13 | 14 | int ImplementThisToRunYourTest(int argc, char** argv) { 15 | testing::InitGoogleTest(&argc, argv); 16 | return RUN_ALL_TESTS(); 17 | } 18 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/openssl.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2013 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_OPENSSL_H_ 12 | #define RTC_BASE_OPENSSL_H_ 13 | 14 | #include 15 | 16 | #if (OPENSSL_VERSION_NUMBER < 0x10000000L) 17 | #error OpenSSL is older than 1.0.0, which is the minimum supported version. 18 | #endif 19 | 20 | #endif // RTC_BASE_OPENSSL_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/run_loop.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #ifndef TEST_RUN_LOOP_H_ 11 | #define TEST_RUN_LOOP_H_ 12 | 13 | namespace webrtc { 14 | namespace test { 15 | 16 | // Blocks until the user presses enter. 17 | void PressEnterToContinue(); 18 | 19 | } // namespace test 20 | } // namespace webrtc 21 | 22 | #endif // TEST_RUN_LOOP_H_ 23 | -------------------------------------------------------------------------------- /razor-ns3/model/razor/cc/razor_log.h: -------------------------------------------------------------------------------- 1 | /*- 2 | * Copyright (c) 2017-2018 wenba, Inc. 3 | * All rights reserved. 4 | * 5 | * See the file LICENSE for redistribution information. 6 | */ 7 | 8 | #ifndef __razor_log_h_ 9 | #define __razor_log_h_ 10 | 11 | #include 12 | #include 13 | #include "razor_api.h" 14 | 15 | void ex_razor_log(int level, const char* file, int line, const char *fmt, ...); 16 | 17 | #define razor_debug(...) ex_razor_log(0, __FILE__, __LINE__, __VA_ARGS__) 18 | #define razor_info(...) ex_razor_log(1, __FILE__, __LINE__, __VA_ARGS__) 19 | #define razor_warn(...) ex_razor_log(2, __FILE__, __LINE__, __VA_ARGS__) 20 | #define razor_error(...) ex_razor_log(3, __FILE__, __LINE__, __VA_ARGS__) 21 | 22 | 23 | #endif 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/rtpsender.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_RTPSENDER_H_ 12 | #define API_RTPSENDER_H_ 13 | 14 | // Including this file is deprecated. It is no longer part of the public API. 15 | // This only includes the file in its new location for backwards compatibility. 16 | #include "pc/rtpsender.h" 17 | 18 | #endif // API_RTPSENDER_H_ 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/webrtcsdp.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_WEBRTCSDP_H_ 12 | #define API_WEBRTCSDP_H_ 13 | 14 | // Including this file is deprecated. It is no longer part of the public API. 15 | // This only includes the file in its new location for backwards compatibility. 16 | #include "pc/webrtcsdp.h" 17 | 18 | #endif // API_WEBRTCSDP_H_ 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/umametrics.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "api/umametrics.h" 12 | 13 | namespace webrtc { 14 | 15 | void MetricsObserverInterface::IncrementSparseEnumCounter( 16 | PeerConnectionEnumCounterType type, 17 | int counter) { 18 | IncrementEnumCounter(type, counter, 0 /* Ignored */); 19 | } 20 | 21 | } // namespace webrtc 22 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/system_wrappers/CMakeLists.txt: -------------------------------------------------------------------------------- 1 | # c++11 required 2 | set(CMAKE_CXX_STANDARD 11) 3 | set(CMAKE_CXX_STANDARD_REQUIRED ON) 4 | include_directories(../../) 5 | include_directories(../) 6 | add_definitions(-D__STDC_FORMAT_MACROS) 7 | set(system_wrapper_FILES 8 | source/aligned_malloc.cc 9 | source/atomic32.cc 10 | source/clock.cc 11 | source/cpu_features.cc 12 | source/cpu_info.cc 13 | source/event.cc 14 | source/event_timer_posix.cc 15 | source/file_impl.cc 16 | source/rtp_to_ntp_estimator.cc 17 | source/rw_lock.cc 18 | source/rw_lock_posix.cc 19 | source/sleep.cc 20 | source/timestamp_extrapolator.cc 21 | source/metrics_default.cc 22 | ) 23 | add_library(system_wrappers SHARED ${system_wrapper_FILES}) 24 | target_link_libraries(system_wrappers rtc_base pthread rt) 25 | -------------------------------------------------------------------------------- /multipathvideo/wscript: -------------------------------------------------------------------------------- 1 | ## -*- Mode: python; py-indent-offset: 4; indent-tabs-mode: nil; coding: utf-8; -*- 2 | 3 | def build(bld): 4 | mp= bld.create_ns3_module('multipathvideo', ['core','network','internet']) 5 | mp.source = [ 6 | 'model/mpcommon.cc', 7 | #'model/mpreceiver.cc', 8 | #'model/recvpath.cc', 9 | #'model/razor_header.cc', 10 | #'model/path.cc', 11 | #'model/mpsender.cc', 12 | ] 13 | headers = bld(features='ns3header') 14 | headers.module = 'multipathvideo' 15 | headers.source = [ 16 | 'model/mpcommon.h', 17 | #'model/mpreceiver.h', 18 | #'model/recvpath.h', 19 | #'model/razor_header.h', 20 | #'model/path.h', 21 | #'model/mpsender.h', 22 | ] 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/datachannel.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_DATACHANNEL_H_ 12 | #define API_DATACHANNEL_H_ 13 | 14 | // Including this file is deprecated. It is no longer part of the public API. 15 | // This only includes the file in its new location for backwards compatibility. 16 | #include "pc/datachannel.h" 17 | 18 | #endif // API_DATACHANNEL_H_ 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/mediastream.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_MEDIASTREAM_H_ 12 | #define API_MEDIASTREAM_H_ 13 | 14 | // Including this file is deprecated. It is no longer part of the public API. 15 | // This only includes the file in its new location for backwards compatibility. 16 | #include "pc/mediastream.h" 17 | 18 | #endif // API_MEDIASTREAM_H_ 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/run_loop.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #include "test/run_loop.h" 11 | 12 | #include 13 | 14 | namespace webrtc { 15 | namespace test { 16 | 17 | void PressEnterToContinue() { 18 | puts(">> Press ENTER to continue..."); 19 | while (getc(stdin) != '\n' && !feof(stdin)); 20 | } 21 | } // namespace test 22 | } // namespace webrtc 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/audio_network_adaptor/controller.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/audio_network_adaptor/controller.h" 12 | 13 | namespace webrtc { 14 | 15 | Controller::NetworkMetrics::NetworkMetrics() = default; 16 | 17 | Controller::NetworkMetrics::~NetworkMetrics() = default; 18 | 19 | } // namespace webrtc 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/test/ACMTest.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_TEST_ACMTEST_H_ 12 | #define MODULES_AUDIO_CODING_TEST_ACMTEST_H_ 13 | 14 | class ACMTest { 15 | public: 16 | ACMTest() {} 17 | virtual ~ACMTest() {} 18 | virtual void Perform() = 0; 19 | }; 20 | 21 | #endif // MODULES_AUDIO_CODING_TEST_ACMTEST_H_ 22 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACPLC.txt: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | (set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character 3 | 4 | LOGFILE=logplc.txt 5 | echo "START PLC TEST" > $LOGFILE 6 | 7 | ISAC=../Release/kenny.exe 8 | 9 | INDIR=../data/orig 10 | OUTDIR=../dataqaplc_0 11 | mkdir -p $OUTDIR 12 | 13 | $ISAC 12000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc1.pcm 14 | $ISAC 20000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc2.pcm 15 | $ISAC 32000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc3.pcm 16 | $ISAC 12000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc4.pcm 17 | $ISAC 20000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc5.pcm 18 | $ISAC 32000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc6.pcm 19 | 20 | echo DONE! 21 | 22 | 23 | 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/memory_usage_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/memory_usage.h" 12 | #include 13 | #include "test/gtest.h" 14 | 15 | namespace rtc { 16 | 17 | TEST(GetMemoryUsage, SimpleTest) { 18 | int64_t used_bytes = GetProcessResidentSizeBytes(); 19 | EXPECT_GE(used_bytes, 0); 20 | } 21 | 22 | } // namespace rtc 23 | 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/congestion_controller/CMakeLists.txt: -------------------------------------------------------------------------------- 1 | # c++11 required 2 | set(CMAKE_CXX_STANDARD 11) 3 | set(CMAKE_CXX_STANDARD_REQUIRED ON) 4 | include_directories(../../../) 5 | include_directories(../../) 6 | set(congestion_controller_FILES 7 | acknowledged_bitrate_estimator.cc 8 | bitrate_estimator.cc 9 | delay_based_bwe.cc 10 | median_slope_estimator.cc 11 | probe_controller.cc 12 | probe_bitrate_estimator.cc 13 | receive_side_congestion_controller.cc 14 | send_side_congestion_controller.cc 15 | transport_feedback_adapter.cc 16 | trendline_estimator.cc 17 | ) 18 | add_library(congestion_controller SHARED ${congestion_controller_FILES}) 19 | target_link_libraries(congestion_controller remote_bitrate_estimator bitrate_controller rtp_rtcp_format system_wrappers pacing logging rtc_base) 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/run_test.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #ifndef TEST_RUN_TEST_H_ 11 | #define TEST_RUN_TEST_H_ 12 | 13 | namespace webrtc { 14 | namespace test { 15 | 16 | // Running a test function on a separate thread, if required by the OS. 17 | void RunTest(void(*test)()); 18 | 19 | } // namespace test 20 | } // namespace webrtc 21 | 22 | #endif // TEST_RUN_TEST_H_ 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/aec3/echo_path_variability.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_processing/aec3/echo_path_variability.h" 12 | 13 | namespace webrtc { 14 | 15 | EchoPathVariability::EchoPathVariability(bool gain_change, bool delay_change) 16 | : gain_change(gain_change), delay_change(delay_change) {} 17 | 18 | } // namespace webrtc 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/mediastreamtrack.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_MEDIASTREAMTRACK_H_ 12 | #define API_MEDIASTREAMTRACK_H_ 13 | 14 | // Including this file is deprecated. It is no longer part of the public API. 15 | // This only includes the file in its new location for backwards compatibility. 16 | #include "pc/mediastreamtrack.h" 17 | 18 | #endif // API_MEDIASTREAMTRACK_H_ 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/streamcollection.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_STREAMCOLLECTION_H_ 12 | #define API_STREAMCOLLECTION_H_ 13 | 14 | // Including this file is deprecated. It is no longer part of the public API. 15 | // This only includes the file in its new location for backwards compatibility. 16 | #include "pc/streamcollection.h" 17 | 18 | #endif // API_STREAMCOLLECTION_H_ 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/screen_capturer_null.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/desktop_capture/desktop_capturer.h" 12 | 13 | namespace webrtc { 14 | 15 | // static 16 | std::unique_ptr DesktopCapturer::CreateRawScreenCapturer( 17 | const DesktopCaptureOptions& options) { 18 | return nullptr; 19 | } 20 | 21 | } // namespace webrtc 22 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/videotracksource.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_VIDEOTRACKSOURCE_H_ 12 | #define API_VIDEOTRACKSOURCE_H_ 13 | 14 | // Including this file is deprecated. It is no longer part of the public API. 15 | // This only includes the file in its new location for backwards compatibility. 16 | #include "pc/videotracksource.h" 17 | 18 | #endif // API_VIDEOTRACKSOURCE_H_ 19 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/include/config.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_processing/include/config.h" 12 | 13 | namespace webrtc { 14 | 15 | Config::Config() {} 16 | 17 | Config::~Config() { 18 | for (OptionMap::iterator it = options_.begin(); it != options_.end(); ++it) { 19 | delete it->second; 20 | } 21 | } 22 | 23 | } // namespace webrtc 24 | -------------------------------------------------------------------------------- /scream/model/videoenc.h: -------------------------------------------------------------------------------- 1 | #ifndef VIDEO_ENC 2 | #define VIDEO_ENC 3 | #include "ns3/callback.h" 4 | namespace ns3 5 | { 6 | class RtpQueue; 7 | #define MAX_FRAMES 10000 8 | class VideoEnc { 9 | public: 10 | VideoEnc(RtpQueue* rtpQueue, float frameRate, char *fname, int ixOffset=0); 11 | 12 | int encode(float time); 13 | 14 | void setTargetBitrate(float targetBitrate); 15 | typedef Callback RateCallback; 16 | void SetRateCallback(RateCallback cb){m_rateCb=cb;} 17 | RtpQueue* rtpQueue; 18 | float frameSize[MAX_FRAMES]; 19 | int nFrames; 20 | float targetBitrate; 21 | float frameRate; 22 | float nominalBitrate; 23 | unsigned int seqNr; 24 | int ix; 25 | float m_fs{40}; 26 | RateCallback m_rateCb; 27 | }; 28 | } 29 | #endif 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/sha1.h: -------------------------------------------------------------------------------- 1 | /* 2 | * SHA-1 in C 3 | * By Steve Reid 4 | * 100% Public Domain 5 | * 6 | */ 7 | 8 | // Ported to C++, Google style, under namespace rtc. 9 | 10 | #ifndef RTC_BASE_SHA1_H_ 11 | #define RTC_BASE_SHA1_H_ 12 | 13 | #include 14 | #include 15 | 16 | namespace rtc { 17 | 18 | struct SHA1_CTX { 19 | uint32_t state[5]; 20 | // TODO: Change bit count to uint64_t. 21 | uint32_t count[2]; // Bit count of input. 22 | uint8_t buffer[64]; 23 | }; 24 | 25 | #define SHA1_DIGEST_SIZE 20 26 | 27 | void SHA1Init(SHA1_CTX* context); 28 | void SHA1Update(SHA1_CTX* context, const uint8_t* data, size_t len); 29 | void SHA1Final(SHA1_CTX* context, uint8_t digest[SHA1_DIGEST_SIZE]); 30 | 31 | } // namespace rtc 32 | 33 | #endif // RTC_BASE_SHA1_H_ 34 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/logging_mac.mm: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2016 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/logging.h" 12 | 13 | #import 14 | 15 | 16 | namespace rtc { 17 | std::string DescriptionFromOSStatus(OSStatus err) { 18 | NSError* error = 19 | [NSError errorWithDomain:NSOSStatusErrorDomain code:err userInfo:nil]; 20 | return error.description.UTF8String; 21 | } 22 | } // namespace rtc 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h" 12 | 13 | namespace webrtc { 14 | 15 | LockedIsacBandwidthInfo::LockedIsacBandwidthInfo() : ref_count_(0) { 16 | bwinfo_.in_use = 0; 17 | } 18 | 19 | LockedIsacBandwidthInfo::~LockedIsacBandwidthInfo() = default; 20 | 21 | } // namespace webrtc 22 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" 12 | 13 | #include "modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h" 14 | 15 | namespace webrtc { 16 | 17 | // Explicit instantiation: 18 | template class AudioDecoderIsacT; 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" 12 | 13 | #include "modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h" 14 | 15 | namespace webrtc { 16 | 17 | // Explicit instantiation: 18 | template class AudioEncoderIsacT; 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/macutils.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2007 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_MACUTILS_H_ 12 | #define RTC_BASE_MACUTILS_H_ 13 | 14 | #include 15 | #include 16 | 17 | namespace rtc { 18 | bool ToUtf8(const CFStringRef str16, std::string* str8); 19 | bool ToUtf16(const std::string& str8, CFStringRef* str16); 20 | } // namespace rtc 21 | 22 | #endif // RTC_BASE_MACUTILS_H_ 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" 12 | 13 | #include "modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h" 14 | 15 | namespace webrtc { 16 | 17 | // Explicit instantiation: 18 | template class AudioDecoderIsacT; 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" 12 | 13 | #include "modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h" 14 | 15 | namespace webrtc { 16 | 17 | // Explicit instantiation: 18 | template class AudioEncoderIsacT; 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // Unit tests for PostDecodeVad class. 12 | 13 | #include "modules/audio_coding/neteq/post_decode_vad.h" 14 | 15 | #include "test/gtest.h" 16 | 17 | namespace webrtc { 18 | 19 | TEST(PostDecodeVad, CreateAndDestroy) { 20 | PostDecodeVad vad; 21 | } 22 | 23 | // TODO(hlundin): Write more tests. 24 | 25 | } // namespace webrtc 26 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_coding/codecs/test/android_test_initializer.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_TEST_INITIALIZER_H_ 12 | #define MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_TEST_INITIALIZER_H_ 13 | 14 | namespace webrtc { 15 | 16 | void InitializeAndroidObjects(); 17 | 18 | } // namespace webrtc 19 | 20 | #endif // MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_TEST_INITIALIZER_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/neteq/random_vector_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // Unit tests for RandomVector class. 12 | 13 | #include "modules/audio_coding/neteq/random_vector.h" 14 | 15 | #include "test/gtest.h" 16 | 17 | namespace webrtc { 18 | 19 | TEST(RandomVector, CreateAndDestroy) { 20 | RandomVector random_vector; 21 | } 22 | 23 | // TODO(hlundin): Write more tests. 24 | 25 | } // namespace webrtc 26 | -------------------------------------------------------------------------------- /razor-ns3/model/gcc_sender.h: -------------------------------------------------------------------------------- 1 | #ifndef GCC_SENDER_H 2 | #define GCC_SENDER_H 3 | #include "pace_sender.h" 4 | #include "ns3/timer.h" 5 | #include "ns3/data-rate.h" 6 | namespace ns3 7 | { 8 | class GccSender 9 | { 10 | public: 11 | GccSender(); 12 | ~GccSender(); 13 | void ChangePacingRate(uint64_t bitrate); 14 | void NotifyPacingPerformed (void); 15 | void Start(void); 16 | void InsertPacket(int num,uint32_t length); 17 | static void DataSend(void* handler, uint32_t packet_id, int retrans, size_t size); 18 | private: 19 | Timer m_pacingTimer{Timer::REMOVE_ON_DESTROY}; 20 | uint64_t m_pacingTime;// in unit MilliSeconds(m_pacingTime) 21 | DataRate m_minPacingRate; 22 | DataRate m_curentPacingRate; 23 | pace_sender_t *m_paceSender; 24 | uint64_t m_paceBitRate; 25 | uint32_t m_sequence; 26 | 27 | }; 28 | } 29 | #endif 30 | -------------------------------------------------------------------------------- /razor-ns3/model/razortrace.h: -------------------------------------------------------------------------------- 1 | #ifndef TRACE_RAZORTRACE_H_ 2 | #define TRACE_RAZORTRACE_H_ 3 | #include 4 | #include 5 | #include 6 | namespace ns3{ 7 | class RazorTraceReceiver{ 8 | public: 9 | RazorTraceReceiver(){} 10 | ~RazorTraceReceiver(){ 11 | Close(); 12 | } 13 | void OpenTraceLossFile(std::string name); 14 | void CloseTraceLossFile(); 15 | void OnLossSeq(uint32_t seq); 16 | private: 17 | void Close(); 18 | std::fstream m_lossSeq; 19 | }; 20 | class RazorTraceSender{ 21 | public: 22 | RazorTraceSender(){} 23 | ~RazorTraceSender(){ 24 | Close(); 25 | } 26 | void OpenTraceSentFile(std::string name); 27 | void CloseTraceSentFile(); 28 | void OnSentSeq(uint32_t seq); 29 | private: 30 | void Close(); 31 | std::fstream m_sendSeq; 32 | }; 33 | } 34 | #endif /* TRACE_RAZORTRACE_H_ */ 35 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/events/rtc_event_logging_started.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "logging/rtc_event_log/events/rtc_event_logging_started.h" 12 | 13 | namespace webrtc { 14 | 15 | RtcEvent::Type RtcEventLoggingStarted::GetType() const { 16 | return RtcEvent::Type::LoggingStarted; 17 | } 18 | 19 | bool RtcEventLoggingStarted::IsConfigEvent() const { 20 | return false; 21 | } 22 | 23 | } // namespace webrtc 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/events/rtc_event_logging_stopped.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "logging/rtc_event_log/events/rtc_event_logging_stopped.h" 12 | 13 | namespace webrtc { 14 | 15 | RtcEvent::Type RtcEventLoggingStopped::GetType() const { 16 | return RtcEvent::Type::LoggingStopped; 17 | } 18 | 19 | bool RtcEventLoggingStopped::IsConfigEvent() const { 20 | return false; 21 | } 22 | 23 | } // namespace webrtc 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/ortc/sessiondescription_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "api/ortc/sessiondescription.h" 12 | #include "test/gtest.h" 13 | 14 | namespace webrtc { 15 | 16 | class SessionDescriptionTest : public testing::Test {}; 17 | 18 | TEST_F(SessionDescriptionTest, CreateSessionDescription) { 19 | SessionDescription s(-1, "0"); 20 | EXPECT_EQ(-1, s.session_id()); 21 | EXPECT_EQ("0", s.session_version()); 22 | } 23 | } 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_processing/aec3/downsampled_render_buffer.h" 12 | 13 | namespace webrtc { 14 | 15 | DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size) 16 | : buffer(downsampled_buffer_size, 0.f) {} 17 | 18 | DownsampledRenderBuffer::~DownsampledRenderBuffer() = default; 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/proxyinfo.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2004 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/proxyinfo.h" 12 | 13 | namespace rtc { 14 | 15 | const char * ProxyToString(ProxyType proxy) { 16 | const char * const PROXY_NAMES[] = { "none", "https", "socks5", "unknown" }; 17 | return PROXY_NAMES[proxy]; 18 | } 19 | 20 | ProxyInfo::ProxyInfo() : type(PROXY_NONE), autodetect(false) { 21 | } 22 | ProxyInfo::~ProxyInfo() = default; 23 | 24 | } // namespace rtc 25 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/null_platform_renderer.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "test/video_renderer.h" 12 | 13 | namespace webrtc { 14 | namespace test { 15 | 16 | VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title, 17 | size_t width, 18 | size_t height) { 19 | return NULL; 20 | } 21 | } // test 22 | } // webrtc 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/video/video_rotation.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_VIDEO_VIDEO_ROTATION_H_ 12 | #define API_VIDEO_VIDEO_ROTATION_H_ 13 | 14 | namespace webrtc { 15 | 16 | // enum for clockwise rotation. 17 | enum VideoRotation { 18 | kVideoRotation_0 = 0, 19 | kVideoRotation_90 = 90, 20 | kVideoRotation_180 = 180, 21 | kVideoRotation_270 = 270 22 | }; 23 | 24 | } // namespace webrtc 25 | 26 | #endif // API_VIDEO_VIDEO_ROTATION_H_ 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/window_finder.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/desktop_capture/window_finder.h" 12 | 13 | namespace webrtc { 14 | 15 | WindowFinder::Options::Options() = default; 16 | WindowFinder::Options::~Options() = default; 17 | WindowFinder::Options::Options(const WindowFinder::Options& other) = default; 18 | WindowFinder::Options::Options(WindowFinder::Options&& other) = default; 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/audio_encoder.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // This file is for backwards compatibility only! Use 12 | // webrtc/api/audio_codecs/audio_encoder.h instead! 13 | // TODO(ossu): Remove it. 14 | 15 | #ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ 16 | #define MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ 17 | 18 | #include "api/audio_codecs/audio_encoder.h" 19 | 20 | #endif // MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/audio_decoder.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // This file is for backwards compatibility only! Use 12 | // webrtc/api/audio_codecs/audio_decoder.h instead! 13 | // TODO(kwiberg): Remove it. 14 | 15 | #ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ 16 | #define MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ 17 | 18 | #include "api/audio_codecs/audio_decoder.h" 19 | 20 | #endif // MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/agc/utility.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_ 12 | #define MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_ 13 | 14 | // TODO(turajs): Add description of function. 15 | double Loudness2Db(double loudness); 16 | 17 | double Linear2Loudness(double rms); 18 | 19 | double Db2Loudness(double db); 20 | 21 | double Dbfs2Loudness(double dbfs); 22 | 23 | #endif // MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_ 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/neteq/background_noise_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // Unit tests for BackgroundNoise class. 12 | 13 | #include "modules/audio_coding/neteq/background_noise.h" 14 | 15 | #include "test/gtest.h" 16 | 17 | namespace webrtc { 18 | 19 | TEST(BackgroundNoise, CreateAndDestroy) { 20 | size_t channels = 1; 21 | BackgroundNoise bgn(channels); 22 | } 23 | 24 | // TODO(hlundin): Write more tests. 25 | 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/audio/DEPS: -------------------------------------------------------------------------------- 1 | include_rules = [ 2 | "+call", 3 | "+common_audio", 4 | "+logging/rtc_event_log", 5 | "+modules/audio_coding", 6 | "+modules/audio_device", 7 | "+modules/audio_mixer", 8 | "+modules/audio_processing/include", 9 | "+modules/bitrate_controller", 10 | "+modules/congestion_controller", 11 | "+modules/pacing", 12 | "+modules/remote_bitrate_estimator", 13 | "+modules/rtp_rtcp", 14 | "+system_wrappers", 15 | "+voice_engine", 16 | ] 17 | 18 | specific_include_rules = { 19 | "audio_send_stream.cc": [ 20 | "+modules/audio_coding/codecs/cng/audio_encoder_cng.h", 21 | ], 22 | # TODO(ossu): Remove this exception when builtin_audio_encoder_factory.h 23 | # has moved to api/, or when the proper mocks have been made. 24 | "audio_send_stream_unittest.cc": [ 25 | "+modules/audio_coding/codecs/builtin_audio_encoder_factory.h", 26 | ], 27 | } 28 | 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef NETEQTEST_DUMMYRTPPACKET_H 12 | #define NETEQTEST_DUMMYRTPPACKET_H 13 | 14 | #include "NETEQTEST_RTPpacket.h" 15 | 16 | class NETEQTEST_DummyRTPpacket : public NETEQTEST_RTPpacket { 17 | public: 18 | int readFromFile(FILE* fp) override; 19 | int writeToFile(FILE* fp) override; 20 | void parseHeader() override; 21 | }; 22 | 23 | #endif // NETEQTEST_DUMMYRTPPACKET_H 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/memory_usage.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #ifndef RTC_BASE_MEMORY_USAGE_H_ 11 | #define RTC_BASE_MEMORY_USAGE_H_ 12 | 13 | #include 14 | 15 | namespace rtc { 16 | 17 | // Returns current memory used by the process in bytes (working set size on 18 | // Windows and resident set size on other platforms). 19 | // Returns -1 on failure. 20 | int64_t GetProcessResidentSizeBytes(); 21 | 22 | } // namespace rtc 23 | 24 | #endif // RTC_BASE_MEMORY_USAGE_H_ 25 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/system_wrappers/include/cpu_info.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_ 12 | #define SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_ 13 | 14 | #include "typedefs.h" // NOLINT(build/include) 15 | 16 | namespace webrtc { 17 | 18 | class CpuInfo { 19 | public: 20 | static uint32_t DetectNumberOfCores(); 21 | 22 | private: 23 | CpuInfo() {} 24 | }; 25 | 26 | } // namespace webrtc 27 | 28 | #endif // SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_ 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/gunit_prod.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2012 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_GUNIT_PROD_H_ 12 | #define RTC_BASE_GUNIT_PROD_H_ 13 | 14 | #if defined(WEBRTC_ANDROID) 15 | // Android doesn't use gtest at all, so anything that relies on gtest should 16 | // check this define first. 17 | #define NO_GTEST 18 | #elif defined (GTEST_RELATIVE_PATH) 19 | #include "gtest/gtest_prod.h" 20 | #else 21 | #include "testing/base/gunit_prod.h" 22 | #endif 23 | 24 | #endif // RTC_BASE_GUNIT_PROD_H_ 25 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_ 12 | #define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_ 13 | 14 | #include 15 | 16 | #include "api/audio_codecs/audio_decoder_factory.h" 17 | 18 | namespace webrtc { 19 | void Pcm16BAppendSupportedCodecSpecs(std::vector* specs); 20 | } 21 | 22 | #endif // MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_ 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/win/cursor.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_DESKTOP_CAPTURE_WIN_CURSOR_H_ 12 | #define MODULES_DESKTOP_CAPTURE_WIN_CURSOR_H_ 13 | 14 | #include 15 | 16 | namespace webrtc { 17 | 18 | class MouseCursor; 19 | 20 | // Converts an HCURSOR into a |MouseCursor| instance. 21 | MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor); 22 | 23 | } // namespace webrtc 24 | 25 | #endif // MODULES_DESKTOP_CAPTURE_WIN_CURSOR_H_ 26 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/constants.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | namespace webrtc { 12 | namespace test { 13 | 14 | extern const int kAudioLevelExtensionId; 15 | extern const int kTOffsetExtensionId; 16 | extern const int kAbsSendTimeExtensionId; 17 | extern const int kTransportSequenceNumberExtensionId; 18 | extern const int kVideoRotationExtensionId; 19 | extern const int kVideoContentTypeExtensionId; 20 | extern const int kVideoTimingExtensionId; 21 | } // namespace test 22 | } // namespace webrtc 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/null_transport.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #include "test/null_transport.h" 11 | 12 | namespace webrtc { 13 | namespace test { 14 | 15 | bool NullTransport::SendRtp(const uint8_t* packet, 16 | size_t length, 17 | const PacketOptions& options) { 18 | return true; 19 | } 20 | 21 | bool NullTransport::SendRtcp(const uint8_t* packet, size_t length) { 22 | return true; 23 | } 24 | 25 | } // namespace test 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "logging/rtc_event_log/events/rtc_event_audio_playout.h" 12 | 13 | namespace webrtc { 14 | 15 | RtcEventAudioPlayout::RtcEventAudioPlayout(uint32_t ssrc) : ssrc_(ssrc) {} 16 | 17 | RtcEvent::Type RtcEventAudioPlayout::GetType() const { 18 | return RtcEvent::Type::AudioPlayout; 19 | } 20 | 21 | bool RtcEventAudioPlayout::IsConfigEvent() const { 22 | return false; 23 | } 24 | 25 | } // namespace webrtc 26 | -------------------------------------------------------------------------------- /webrtc-results/pro_loss.py: -------------------------------------------------------------------------------- 1 | ''' 2 | https://blog.csdn.net/qq_29422251/article/details/77713741 3 | ''' 4 | def ReadLossInfo(fileName): 5 | count=0 6 | #cache the last line 7 | line="" 8 | for index, line in enumerate(open(fileName,'r')): 9 | count += 1 10 | lineArr = line.strip().split() 11 | return count,int(lineArr[1]) 12 | instance=1 13 | flows=3; 14 | fileout="webrtc_loss.txt" 15 | name="%s_webrtc_%s_feedback.txt" 16 | fout=open(fileout,'w') 17 | for case in range(instance): 18 | total_recv=0 19 | total=0 20 | average_loss=0.0 21 | for i in range(flows): 22 | filename=name%(str(case+1),str(i+1)) 23 | recv,max_recv=ReadLossInfo(filename) 24 | total_recv+=recv 25 | total+=max_recv 26 | average_loss=float(total-total_recv)/total 27 | fout.write(str(case+1)+"\t") 28 | fout.write(str(average_loss)+"\n") 29 | fout.close() 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/optional.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2016 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "api/optional.h" 12 | 13 | namespace rtc { 14 | namespace optional_internal { 15 | 16 | #if RTC_HAS_ASAN 17 | 18 | void* FunctionThatDoesNothingImpl(void* x) { 19 | return x; 20 | } 21 | 22 | #endif 23 | 24 | struct NulloptArg { 25 | constexpr NulloptArg() {} 26 | }; 27 | 28 | static NulloptArg nullopt_arg; 29 | 30 | } // namespace optional_internal 31 | 32 | const nullopt_t nullopt(rtc::optional_internal::nullopt_arg); 33 | 34 | } // namespace rtc 35 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css: -------------------------------------------------------------------------------- 1 | /* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 2 | * 3 | * Use of this source code is governed by a BSD-style license 4 | * that can be found in the LICENSE file in the root of the source 5 | * tree. An additional intellectual property rights grant can be found 6 | * in the file PATENTS. All contributing project authors may 7 | * be found in the AUTHORS file in the root of the source tree. 8 | */ 9 | 10 | td.selected-score { 11 | background-color: #DDD; 12 | } 13 | 14 | td.single-score-cell{ 15 | text-align: center; 16 | } 17 | 18 | .audio-inspector { 19 | text-align: center; 20 | } 21 | 22 | .audio-inspector div{ 23 | margin-bottom: 0; 24 | padding-bottom: 0; 25 | padding-top: 0; 26 | } 27 | 28 | .audio-inspector div div{ 29 | margin-bottom: 0; 30 | padding-bottom: 0; 31 | padding-top: 0; 32 | } 33 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_ 12 | #define MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_ 13 | 14 | #pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; " 15 | "use video_coding/include") 16 | 17 | #include "modules/video_coding/include/video_codec_interface.h" 18 | 19 | #endif // MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_ 20 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/constants.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "test/constants.h" 12 | 13 | namespace webrtc { 14 | namespace test { 15 | 16 | const int kAudioLevelExtensionId = 5; 17 | const int kTOffsetExtensionId = 6; 18 | const int kAbsSendTimeExtensionId = 7; 19 | const int kTransportSequenceNumberExtensionId = 8; 20 | const int kVideoRotationExtensionId = 9; 21 | const int kVideoContentTypeExtensionId = 10; 22 | const int kVideoTimingExtensionId = 11; 23 | 24 | } // namespace test 25 | } // namespace webrtc 26 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/asyncpacketsocket.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/asyncpacketsocket.h" 12 | 13 | namespace rtc { 14 | 15 | PacketTimeUpdateParams::PacketTimeUpdateParams() 16 | : rtp_sendtime_extension_id(-1), 17 | srtp_auth_tag_len(-1), 18 | srtp_packet_index(-1) { 19 | } 20 | 21 | PacketTimeUpdateParams::~PacketTimeUpdateParams() = default; 22 | 23 | AsyncPacketSocket::AsyncPacketSocket() { 24 | } 25 | 26 | AsyncPacketSocket::~AsyncPacketSocket() { 27 | } 28 | 29 | }; // namespace rtc 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_ 12 | #define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_ 13 | 14 | #include "typedefs.h" // NOLINT(build/include) 15 | 16 | typedef struct { 17 | int in_use; 18 | int32_t send_bw_avg; 19 | int32_t send_max_delay_avg; 20 | int16_t bottleneck_idx; 21 | int16_t jitter_info; 22 | } IsacBandwidthInfo; 23 | 24 | #endif // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_ 25 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/compile_assert_c.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_COMPILE_ASSERT_C_H_ 12 | #define RTC_BASE_COMPILE_ASSERT_C_H_ 13 | 14 | // Use this macro to verify at compile time that certain restrictions are met. 15 | // The argument is the boolean expression to evaluate. 16 | // Example: 17 | // RTC_COMPILE_ASSERT(sizeof(foo) < 128); 18 | // Note: In C++, use static_assert instead! 19 | #define RTC_COMPILE_ASSERT(expression) switch (0) {case 0: case expression:;} 20 | 21 | #endif // RTC_BASE_COMPILE_ASSERT_C_H_ 22 | -------------------------------------------------------------------------------- /razor-ns3/model/razor/estimator/sender_history.h: -------------------------------------------------------------------------------- 1 | /*- 2 | * Copyright (c) 2017-2018 wenba, Inc. 3 | * All rights reserved. 4 | * 5 | * See the file LICENSE for redistribution information. 6 | */ 7 | 8 | #ifndef __sender_history_h_ 9 | #define __sender_history_h_ 10 | 11 | #include "estimator_common.h" 12 | #include "cf_platform.h" 13 | #include "cf_skiplist.h" 14 | #include "cf_unwrapper.h" 15 | 16 | typedef struct 17 | { 18 | uint32_t limited_ms; 19 | 20 | cf_unwrapper_t wrapper; 21 | int64_t last_ack_seq_num; 22 | skiplist_t* l; 23 | }sender_history_t; 24 | 25 | sender_history_t* sender_history_create(uint32_t limited_ms); 26 | void sender_history_destroy(sender_history_t* hist); 27 | 28 | void sender_history_add(sender_history_t* hist, packet_feedback_t* packet); 29 | int sender_history_get(sender_history_t* hist, uint16_t seq, packet_feedback_t* packet); 30 | 31 | size_t sender_history_outstanding_bytes(sender_history_t* hist); 32 | #endif 33 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/system_wrappers/include/sleep.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | // An OS-independent sleep function. 11 | 12 | #ifndef SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_ 13 | #define SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_ 14 | 15 | namespace webrtc { 16 | 17 | // This function sleeps for the specified number of milliseconds. 18 | // It may return early if the thread is woken by some other event, 19 | // such as the delivery of a signal on Unix. 20 | void SleepMs(int msecs); 21 | 22 | } // namespace webrtc 23 | 24 | #endif // SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_ 25 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // This file is for backwards compatibility only! Use 12 | // webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead! 13 | // TODO(ossu): Remove it. 14 | 15 | #ifndef MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_ 16 | #define MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_ 17 | 18 | #include "api/audio_codecs/builtin_audio_encoder_factory.h" 19 | 20 | #endif // MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/utility/include/audio_frame_operations.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ 12 | #define MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ 13 | // The contents of this file have moved to 14 | // //audio/utility. This file is deprecated. 15 | 16 | // TODO(aleloi): Remove this file when clients have updated their 17 | // includes. See bugs.webrtc.org/6548. 18 | #include "audio/utility/audio_frame_operations.h" 19 | 20 | #endif // #ifndef MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/system_wrappers/source/rw_lock.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "system_wrappers/include/rw_lock_wrapper.h" 12 | 13 | #include 14 | 15 | #if defined(_WIN32) 16 | #include "system_wrappers/source/rw_lock_win.h" 17 | #else 18 | #include "system_wrappers/source/rw_lock_posix.h" 19 | #endif 20 | 21 | namespace webrtc { 22 | 23 | RWLockWrapper* RWLockWrapper::CreateRWLock() { 24 | #ifdef _WIN32 25 | return RWLockWin::Create(); 26 | #else 27 | return RWLockPosix::Create(); 28 | #endif 29 | } 30 | 31 | } // namespace webrtc 32 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // This file is for backwards compatibility only! Use 12 | // webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead! 13 | // TODO(kwiberg): Remove it. 14 | 15 | #ifndef MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_ 16 | #define MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_ 17 | 18 | #include "api/audio_codecs/builtin_audio_decoder_factory.h" 19 | 20 | #endif // MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_ 21 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_capture/external/video_capture_external.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/video_capture/video_capture_impl.h" 12 | #include "rtc_base/refcount.h" 13 | #include "rtc_base/refcountedobject.h" 14 | 15 | namespace webrtc { 16 | 17 | namespace videocapturemodule { 18 | 19 | rtc::scoped_refptr VideoCaptureImpl::Create( 20 | const char* deviceUniqueIdUTF8) { 21 | return new rtc::RefCountedObject(); 22 | } 23 | 24 | } // namespace videocapturemodule 25 | 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/transient/test/readPCM.m: -------------------------------------------------------------------------------- 1 | % 2 | % Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | % 4 | % Use of this source code is governed by a BSD-style license 5 | % that can be found in the LICENSE file in the root of the source 6 | % tree. An additional intellectual property rights grant can be found 7 | % in the file PATENTS. All contributing project authors may 8 | % be found in the AUTHORS file in the root of the source tree. 9 | % 10 | 11 | function [x, t] = readPCM(file, fs) 12 | %[x, t] = readPCM(file, fs) 13 | % 14 | %Reads a signal from a PCM file. 15 | % 16 | %x: The read signal after normalization. 17 | %t: The respective time vector. 18 | % 19 | %file: The PCM file where the signal is stored in int16 format. 20 | %fs: The signal sample rate in Hertz. 21 | fid = fopen(file); 22 | x = fread(fid, inf, 'int16'); 23 | fclose(fid); 24 | x = x - mean(x); 25 | x = x / max(abs(x)); 26 | t = 0:(1 / fs):((length(x) - 1) / fs); 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/shared_memory.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/desktop_capture/shared_memory.h" 12 | 13 | namespace webrtc { 14 | 15 | #if defined(WEBRTC_WIN) 16 | const SharedMemory::Handle SharedMemory::kInvalidHandle = NULL; 17 | #else 18 | const SharedMemory::Handle SharedMemory::kInvalidHandle = -1; 19 | #endif 20 | 21 | SharedMemory::SharedMemory(void* data, size_t size, Handle handle, int id) 22 | : data_(data), 23 | size_(size), 24 | handle_(handle), 25 | id_(id) { 26 | } 27 | 28 | } // namespace webrtc 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/fake_videorenderer.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef TEST_FAKE_VIDEORENDERER_H_ 12 | #define TEST_FAKE_VIDEORENDERER_H_ 13 | 14 | #include "api/video/video_frame.h" 15 | #include "media/base/videosinkinterface.h" 16 | 17 | namespace webrtc { 18 | namespace test { 19 | 20 | class FakeVideoRenderer : public rtc::VideoSinkInterface { 21 | public: 22 | void OnFrame(const webrtc::VideoFrame& frame) override {} 23 | }; 24 | 25 | } // namespace test 26 | } // namespace webrtc 27 | 28 | #endif // TEST_FAKE_VIDEORENDERER_H_ 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/neteq/tools/audio_sink.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/neteq/tools/audio_sink.h" 12 | 13 | namespace webrtc { 14 | namespace test { 15 | 16 | bool AudioSinkFork::WriteArray(const int16_t* audio, size_t num_samples) { 17 | return left_sink_->WriteArray(audio, num_samples) && 18 | right_sink_->WriteArray(audio, num_samples); 19 | } 20 | 21 | bool VoidAudioSink::WriteArray(const int16_t* audio, size_t num_samples) { 22 | return true; 23 | } 24 | 25 | } // namespace test 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/mock_desktop_capturer_callback.cc: -------------------------------------------------------------------------------- 1 | /* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 | * 3 | * Use of this source code is governed by a BSD-style license 4 | * that can be found in the LICENSE file in the root of the source 5 | * tree. An additional intellectual property rights grant can be found 6 | * in the file PATENTS. All contributing project authors may 7 | * be found in the AUTHORS file in the root of the source tree. 8 | */ 9 | 10 | #include "modules/desktop_capture/mock_desktop_capturer_callback.h" 11 | 12 | namespace webrtc { 13 | 14 | MockDesktopCapturerCallback::MockDesktopCapturerCallback() = default; 15 | MockDesktopCapturerCallback::~MockDesktopCapturerCallback() = default; 16 | 17 | void MockDesktopCapturerCallback::OnCaptureResult( 18 | DesktopCapturer::Result result, 19 | std::unique_ptr frame) { 20 | OnCaptureResultPtr(result, &frame); 21 | } 22 | 23 | } // namespace webrtc 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/testfield.cc: -------------------------------------------------------------------------------- 1 | //#define NDEBUG 2 | #include "rtc_base/logging.h" 3 | #include "test/field_trial.h" 4 | #include "system_wrappers/include/field_trial.h" 5 | #include 6 | #include 7 | #include 8 | 9 | class LogSinkConsole:public rtc::LogSink 10 | { 11 | public: 12 | void OnLogMessage(const std::string &message) override 13 | { 14 | std::cout<<"log concole message"<<' '<(v) / (1 << 8); 19 | } 20 | 21 | // Convert fixed point number with 14 bit fractional part, to floating point. 22 | inline float Q14ToFloat(uint32_t v) { 23 | return static_cast(v) / (1 << 14); 24 | } 25 | } // namespace webrtc 26 | 27 | #endif // AUDIO_CONVERSION_H_ 28 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/md5digest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/md5digest.h" 12 | 13 | namespace rtc { 14 | 15 | size_t Md5Digest::Size() const { 16 | return kSize; 17 | } 18 | 19 | void Md5Digest::Update(const void* buf, size_t len) { 20 | MD5Update(&ctx_, static_cast(buf), len); 21 | } 22 | 23 | size_t Md5Digest::Finish(void* buf, size_t len) { 24 | if (len < kSize) { 25 | return 0; 26 | } 27 | MD5Final(&ctx_, static_cast(buf)); 28 | MD5Init(&ctx_); // Reset for next use. 29 | return kSize; 30 | } 31 | 32 | }; // namespace rtc 33 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/gtest.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef TEST_GTEST_H_ 12 | #define TEST_GTEST_H_ 13 | 14 | #include "rtc_base/ignore_wundef.h" 15 | 16 | RTC_PUSH_IGNORING_WUNDEF() 17 | #include "testing/gtest/include/gtest/gtest.h" 18 | RTC_POP_IGNORING_WUNDEF() 19 | 20 | // GTEST_HAS_DEATH_TEST is set to 1 when death tests are supported, but appears 21 | // to be left unset if they're not supported. Rather than depend on this, we 22 | // set it to 0 ourselves here. 23 | #ifndef GTEST_HAS_DEATH_TEST 24 | #define GTEST_HAS_DEATH_TEST 0 25 | #endif 26 | 27 | #endif // TEST_GTEST_H_ 28 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "logging/rtc_event_log/events/rtc_event_probe_result_success.h" 12 | 13 | namespace webrtc { 14 | 15 | RtcEventProbeResultSuccess::RtcEventProbeResultSuccess(int id, int bitrate_bps) 16 | : id_(id), bitrate_bps_(bitrate_bps) {} 17 | 18 | RtcEvent::Type RtcEventProbeResultSuccess::GetType() const { 19 | return RtcEvent::Type::ProbeResultSuccess; 20 | } 21 | 22 | bool RtcEventProbeResultSuccess::IsConfigEvent() const { 23 | return false; 24 | } 25 | 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h" 12 | 13 | namespace webrtc { 14 | 15 | void Pcm16BAppendSupportedCodecSpecs(std::vector* specs) { 16 | for (uint8_t num_channels : {1, 2}) { 17 | for (int sample_rate_hz : {8000, 16000, 32000}) { 18 | specs->push_back( 19 | {{"L16", sample_rate_hz, num_channels}, 20 | {sample_rate_hz, num_channels, sample_rate_hz * num_channels * 16}}); 21 | } 22 | } 23 | } 24 | 25 | } // namespace webrtc 26 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_ 12 | #define MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_ 13 | 14 | #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 15 | #include "test/gmock.h" 16 | 17 | namespace webrtc { 18 | 19 | class MockRtcpRttStats : public RtcpRttStats { 20 | public: 21 | MOCK_METHOD1(OnRttUpdate, void(int64_t rtt)); 22 | MOCK_CONST_METHOD0(LastProcessedRtt, int64_t()); 23 | }; 24 | } // namespace webrtc 25 | #endif // MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_ 26 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/sha1digest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2015 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/sha1digest.h" 12 | 13 | namespace rtc { 14 | 15 | size_t Sha1Digest::Size() const { 16 | return kSize; 17 | } 18 | 19 | void Sha1Digest::Update(const void* buf, size_t len) { 20 | SHA1Update(&ctx_, static_cast(buf), len); 21 | } 22 | 23 | size_t Sha1Digest::Finish(void* buf, size_t len) { 24 | if (len < kSize) { 25 | return 0; 26 | } 27 | SHA1Final(&ctx_, static_cast(buf)); 28 | SHA1Init(&ctx_); // Reset for next use. 29 | return kSize; 30 | } 31 | 32 | } // namespace rtc 33 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/transient/common.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_ 12 | #define MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_ 13 | namespace webrtc { 14 | namespace ts { 15 | 16 | static const float kPi = 3.14159265358979323846f; 17 | static const int kChunkSizeMs = 10; 18 | enum { 19 | kSampleRate8kHz = 8000, 20 | kSampleRate16kHz = 16000, 21 | kSampleRate32kHz = 32000, 22 | kSampleRate48kHz = 48000 23 | }; 24 | 25 | } // namespace ts 26 | } // namespace webrtc 27 | #endif // MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_ 28 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/resolution_tracker.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/desktop_capture/resolution_tracker.h" 12 | 13 | namespace webrtc { 14 | 15 | bool ResolutionTracker::SetResolution(DesktopSize size) { 16 | if (!initialized_) { 17 | initialized_ = true; 18 | last_size_ = size; 19 | return false; 20 | } 21 | 22 | if (last_size_.equals(size)) { 23 | return false; 24 | } 25 | 26 | last_size_ = size; 27 | return true; 28 | } 29 | 30 | void ResolutionTracker::Reset() { 31 | initialized_ = false; 32 | } 33 | 34 | } // namespace webrtc 35 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/screen_drawer.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/desktop_capture/screen_drawer.h" 12 | 13 | namespace webrtc { 14 | 15 | namespace { 16 | std::unique_ptr g_screen_drawer_lock; 17 | } // namespace 18 | 19 | ScreenDrawerLock::ScreenDrawerLock() = default; 20 | ScreenDrawerLock::~ScreenDrawerLock() = default; 21 | 22 | ScreenDrawer::ScreenDrawer() { 23 | g_screen_drawer_lock = ScreenDrawerLock::Create(); 24 | } 25 | 26 | ScreenDrawer::~ScreenDrawer() { 27 | g_screen_drawer_lock.reset(); 28 | } 29 | 30 | } // namespace webrtc 31 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_coding/qp_parser.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_VIDEO_CODING_QP_PARSER_H_ 12 | #define MODULES_VIDEO_CODING_QP_PARSER_H_ 13 | 14 | #include "modules/video_coding/encoded_frame.h" 15 | 16 | namespace webrtc { 17 | 18 | class QpParser { 19 | public: 20 | QpParser() {} 21 | ~QpParser() {} 22 | 23 | // Parses an encoded |frame| and extracts the |qp|. 24 | // Returns true on success, false otherwise. 25 | bool GetQp(const VCMEncodedFrame& frame, int* qp); 26 | }; 27 | 28 | } // namespace webrtc 29 | 30 | #endif // MODULES_VIDEO_CODING_QP_PARSER_H_ 31 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/ratelimiter.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2012 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/ratelimiter.h" 12 | 13 | namespace rtc { 14 | 15 | bool RateLimiter::CanUse(size_t desired, double time) { 16 | return ((time > period_end_ && desired <= max_per_period_) || 17 | (used_in_period_ + desired) <= max_per_period_); 18 | } 19 | 20 | void RateLimiter::Use(size_t used, double time) { 21 | if (time > period_end_) { 22 | period_start_ = time; 23 | period_end_ = time + period_length_; 24 | used_in_period_ = 0; 25 | } 26 | used_in_period_ += used; 27 | } 28 | 29 | } // namespace rtc 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/mock_transport.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef TEST_MOCK_TRANSPORT_H_ 12 | #define TEST_MOCK_TRANSPORT_H_ 13 | 14 | #include "api/call/transport.h" 15 | #include "test/gmock.h" 16 | 17 | namespace webrtc { 18 | 19 | class MockTransport : public Transport { 20 | public: 21 | MOCK_METHOD3(SendRtp, 22 | bool(const uint8_t* data, 23 | size_t len, 24 | const PacketOptions& options)); 25 | MOCK_METHOD2(SendRtcp, bool(const uint8_t* data, size_t len)); 26 | }; 27 | } // namespace webrtc 28 | #endif // TEST_MOCK_TRANSPORT_H_ 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/aec3/echo_path_variability.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_ 12 | #define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_ 13 | 14 | namespace webrtc { 15 | 16 | struct EchoPathVariability { 17 | EchoPathVariability(bool gain_change, bool delay_change); 18 | 19 | bool AudioPathChanged() const { return gain_change || delay_change; } 20 | bool gain_change; 21 | bool delay_change; 22 | }; 23 | 24 | } // namespace webrtc 25 | 26 | #endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_ 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/screen_drawer_mac.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // TODO(zijiehe): Implement ScreenDrawerMac 12 | 13 | #include "modules/desktop_capture/screen_drawer.h" 14 | #include "modules/desktop_capture/screen_drawer_lock_posix.h" 15 | #include "rtc_base/ptr_util.h" 16 | 17 | namespace webrtc { 18 | 19 | // static 20 | std::unique_ptr ScreenDrawerLock::Create() { 21 | return rtc::MakeUnique(); 22 | } 23 | 24 | // static 25 | std::unique_ptr ScreenDrawer::Create() { 26 | return nullptr; 27 | } 28 | 29 | } // namespace webrtc 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/cpu_time.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_CPU_TIME_H_ 12 | #define RTC_BASE_CPU_TIME_H_ 13 | 14 | #include 15 | 16 | namespace rtc { 17 | 18 | // Returns total CPU time of a current process in nanoseconds. 19 | // Time base is unknown, therefore use only to calculate deltas. 20 | int64_t GetProcessCpuTimeNanos(); 21 | 22 | // Returns total CPU time of a current thread in nanoseconds. 23 | // Time base is unknown, therefore use only to calculate deltas. 24 | int64_t GetThreadCpuTimeNanos(); 25 | 26 | } // namespace rtc 27 | 28 | #endif // RTC_BASE_CPU_TIME_H_ 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/task_queue_posix.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2016 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_TASK_QUEUE_POSIX_H_ 12 | #define RTC_BASE_TASK_QUEUE_POSIX_H_ 13 | 14 | #include 15 | 16 | namespace rtc { 17 | 18 | class TaskQueue; 19 | 20 | namespace internal { 21 | 22 | class AutoSetCurrentQueuePtr { 23 | public: 24 | explicit AutoSetCurrentQueuePtr(TaskQueue* q); 25 | ~AutoSetCurrentQueuePtr(); 26 | 27 | private: 28 | TaskQueue* const prev_; 29 | }; 30 | 31 | pthread_key_t GetQueuePtrTls(); 32 | 33 | } // namespace internal 34 | } // namespace rtc 35 | 36 | #endif // RTC_BASE_TASK_QUEUE_POSIX_H_ 37 | -------------------------------------------------------------------------------- /webrtc-ns3/model/simulationclock.cc: -------------------------------------------------------------------------------- 1 | #include "simulationclock.h" 2 | #include "ns3/log.h" 3 | namespace ns3 4 | { 5 | NS_LOG_COMPONENT_DEFINE("WebrtcSimulationClock"); 6 | int64_t SimulationClock::TimeInMilliseconds() const 7 | { 8 | return Simulator::Now().GetMilliSeconds(); 9 | } 10 | int64_t SimulationClock::TimeInMicroseconds() const 11 | { 12 | return Simulator::Now().GetMicroSeconds(); 13 | } 14 | webrtc::NtpTime SimulationClock::CurrentNtpTime() const 15 | { 16 | NS_LOG_FUNCTION("this should not happen"); 17 | int64_t now_ms =Simulator::Now().GetMilliSeconds(); 18 | uint32_t seconds = (now_ms / 1000) + webrtc::kNtpJan1970; 19 | uint32_t fractions = 20 | static_cast((now_ms % 1000) * webrtc::kMagicNtpFractionalUnit / 1000); 21 | return webrtc::NtpTime(seconds, fractions); 22 | } 23 | int64_t SimulationClock::CurrentNtpInMilliseconds() const 24 | { 25 | NS_LOG_FUNCTION("this should not happen"); 26 | return Simulator::Now().GetMilliSeconds() + 1000 * static_cast(webrtc::kNtpJan1970); 27 | } 28 | } 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/neteq/tick_timer.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/neteq/tick_timer.h" 12 | 13 | namespace webrtc { 14 | 15 | TickTimer::Stopwatch::Stopwatch(const TickTimer& ticktimer) 16 | : ticktimer_(ticktimer), starttick_(ticktimer.ticks()) {} 17 | 18 | TickTimer::Countdown::Countdown(const TickTimer& ticktimer, 19 | uint64_t ticks_to_count) 20 | : stopwatch_(ticktimer.GetNewStopwatch()), 21 | ticks_to_count_(ticks_to_count) {} 22 | 23 | TickTimer::Countdown::~Countdown() = default; 24 | 25 | } // namespace webrtc 26 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/level_controller/level_controller_constants.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_ 12 | #define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_ 13 | 14 | namespace webrtc { 15 | 16 | const float kMaxLcGain = 10; 17 | const float kMaxLcNoisePower = 100.f * 100.f; 18 | const float kTargetLcPeakLevel = 16384.f; 19 | const float kTargetLcPeakLeveldBFS = -6.0206f; 20 | 21 | } // namespace webrtc 22 | 23 | #endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_ 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/mediatypes.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_MEDIATYPES_H_ 12 | #define API_MEDIATYPES_H_ 13 | 14 | #include 15 | 16 | namespace cricket { 17 | 18 | enum MediaType { 19 | MEDIA_TYPE_AUDIO, 20 | MEDIA_TYPE_VIDEO, 21 | MEDIA_TYPE_DATA 22 | }; 23 | 24 | std::string MediaTypeToString(MediaType type); 25 | // Aborts on invalid string. Only expected to be used on strings that are 26 | // guaranteed to be valid, such as MediaStreamTrackInterface::kind(). 27 | MediaType MediaTypeFromString(const std::string& type_str); 28 | 29 | } // namespace cricket 30 | 31 | #endif // API_MEDIATYPES_H_ 32 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_H_ 12 | #define LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_H_ 13 | 14 | #include 15 | 16 | #include "logging/rtc_event_log/events/rtc_event.h" 17 | 18 | namespace webrtc { 19 | class RtcEventLogEncoder { 20 | public: 21 | virtual ~RtcEventLogEncoder() = default; 22 | 23 | virtual std::string Encode(const RtcEvent& event) = 0; 24 | }; 25 | 26 | } // namespace webrtc 27 | 28 | #endif // LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_H_ 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/rtc_event_log_factory.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "logging/rtc_event_log/rtc_event_log_factory.h" 12 | 13 | #include "logging/rtc_event_log/rtc_event_log.h" 14 | 15 | namespace webrtc { 16 | 17 | std::unique_ptr RtcEventLogFactory::CreateRtcEventLog( 18 | RtcEventLog::EncodingType encoding_type) { 19 | return RtcEventLog::Create(encoding_type); 20 | } 21 | 22 | std::unique_ptr CreateRtcEventLogFactory() { 23 | return std::unique_ptr(new RtcEventLogFactory()); 24 | } 25 | 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/onetimeevent_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2016 The WebRTC Project Authors. All rights reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "rtc_base/onetimeevent.h" 12 | #include "rtc_base/gunit.h" 13 | 14 | namespace webrtc { 15 | 16 | TEST(OneTimeEventTest, ThreadSafe) { 17 | OneTimeEvent ot; 18 | 19 | // The one time event is expected to evaluate to true only the first time. 20 | EXPECT_TRUE(ot()); 21 | EXPECT_FALSE(ot()); 22 | EXPECT_FALSE(ot()); 23 | } 24 | 25 | TEST(OneTimeEventTest, ThreadUnsafe) { 26 | ThreadUnsafeOneTimeEvent ot; 27 | 28 | EXPECT_TRUE(ot()); 29 | EXPECT_FALSE(ot()); 30 | EXPECT_FALSE(ot()); 31 | } 32 | 33 | } // namespace webrtc 34 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/null_transport.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #ifndef TEST_NULL_TRANSPORT_H_ 11 | #define TEST_NULL_TRANSPORT_H_ 12 | 13 | #include "api/call/transport.h" 14 | 15 | namespace webrtc { 16 | 17 | class PacketReceiver; 18 | 19 | namespace test { 20 | class NullTransport : public Transport { 21 | public: 22 | bool SendRtp(const uint8_t* packet, 23 | size_t length, 24 | const PacketOptions& options) override; 25 | bool SendRtcp(const uint8_t* packet, size_t length) override; 26 | }; 27 | } // namespace test 28 | } // namespace webrtc 29 | 30 | #endif // TEST_NULL_TRANSPORT_H_ 31 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/audio_format_conversion.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_ 12 | #define MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_ 13 | 14 | #include "api/audio_codecs/audio_format.h" 15 | #include "common_types.h" // NOLINT(build/include) 16 | 17 | namespace webrtc { 18 | 19 | SdpAudioFormat CodecInstToSdp(const CodecInst& codec_inst); 20 | CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format); 21 | 22 | } // namespace webrtc 23 | 24 | #endif // MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_ 25 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/neteq/tools/packet_source.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_coding/neteq/tools/packet_source.h" 12 | 13 | namespace webrtc { 14 | namespace test { 15 | 16 | PacketSource::PacketSource() : use_ssrc_filter_(false), ssrc_(0) {} 17 | 18 | PacketSource::~PacketSource() = default; 19 | 20 | void PacketSource::FilterOutPayloadType(uint8_t payload_type) { 21 | filter_.set(payload_type, true); 22 | } 23 | 24 | void PacketSource::SelectSsrc(uint32_t ssrc) { 25 | use_ssrc_filter_ = true; 26 | ssrc_ = ssrc; 27 | } 28 | 29 | } // namespace test 30 | } // namespace webrtc 31 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/test/statistics.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | #ifndef TEST_STATISTICS_H_ 11 | #define TEST_STATISTICS_H_ 12 | 13 | #include "typedefs.h" // NOLINT(build/include) 14 | 15 | namespace webrtc { 16 | namespace test { 17 | 18 | class Statistics { 19 | public: 20 | Statistics(); 21 | 22 | void AddSample(double sample); 23 | 24 | double Mean() const; 25 | double Variance() const; 26 | double StandardDeviation() const; 27 | 28 | private: 29 | double sum_; 30 | double sum_squared_; 31 | uint64_t count_; 32 | }; 33 | } // namespace test 34 | } // namespace webrtc 35 | 36 | #endif // TEST_STATISTICS_H_ 37 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_ 12 | #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_ 13 | 14 | #include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h" 15 | #include "modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h" 16 | 17 | namespace webrtc { 18 | 19 | using AudioDecoderIsacFixImpl = AudioDecoderIsacT; 20 | 21 | } // namespace webrtc 22 | #endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_ 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_ 12 | #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_ 13 | 14 | #include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h" 15 | #include "modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h" 16 | 17 | namespace webrtc { 18 | 19 | using AudioEncoderIsacFixImpl = AudioEncoderIsacT; 20 | 21 | } // namespace webrtc 22 | #endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_ 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_ 12 | #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_ 13 | 14 | #include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h" 15 | #include "modules/audio_coding/codecs/isac/main/source/isac_float_type.h" 16 | 17 | namespace webrtc { 18 | 19 | using AudioDecoderIsacFloatImpl = AudioDecoderIsacT; 20 | 21 | } // namespace webrtc 22 | #endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ 12 | #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ 13 | 14 | #include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h" 15 | #include "modules/audio_coding/codecs/isac/main/source/isac_float_type.h" 16 | 17 | namespace webrtc { 18 | 19 | using AudioEncoderIsacFloatImpl = AudioEncoderIsacT; 20 | 21 | } // namespace webrtc 22 | #endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ 23 | -------------------------------------------------------------------------------- /razor-ns3/model/razor/common/cf_list.h: -------------------------------------------------------------------------------- 1 | /*- 2 | * Copyright (c) 2017-2018 wenba, Inc. 3 | * All rights reserved. 4 | * 5 | * See the file LICENSE for redistribution information. 6 | */ 7 | 8 | #ifndef __WB_LIST_H_ 9 | #define __WB_LIST_H_ 10 | 11 | #include 12 | 13 | #ifdef __cplusplus 14 | extern "C" { 15 | #endif 16 | 17 | typedef struct base_list_unit_t{ 18 | struct base_list_unit_t* next; 19 | void* pdata; 20 | }base_list_unit_t; 21 | 22 | typedef struct { 23 | base_list_unit_t* head; 24 | base_list_unit_t* tailer; 25 | size_t size; 26 | } base_list_t; 27 | 28 | 29 | base_list_t* create_list(); 30 | void destroy_list(base_list_t* l); 31 | 32 | void list_push(base_list_t* l, void* data); 33 | void* list_pop(base_list_t* l); 34 | void* list_front(base_list_t* l); 35 | 36 | size_t list_size(base_list_t* l); 37 | 38 | #define LIST_FOREACH(l, pos) \ 39 | for (pos = (l)->head; pos != NULL; pos = pos->next) 40 | 41 | 42 | #ifdef __cplusplus 43 | } 44 | #endif 45 | 46 | #endif 47 | 48 | 49 | 50 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h" 12 | 13 | namespace webrtc { 14 | 15 | RtcEventProbeResultFailure::RtcEventProbeResultFailure( 16 | int id, 17 | ProbeFailureReason failure_reason) 18 | : id_(id), failure_reason_(failure_reason) {} 19 | 20 | RtcEvent::Type RtcEventProbeResultFailure::GetType() const { 21 | return RtcEvent::Type::ProbeResultFailure; 22 | } 23 | 24 | bool RtcEventProbeResultFailure::IsConfigEvent() const { 25 | return false; 26 | } 27 | 28 | } // namespace webrtc 29 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/rtp_rtcp/source/video_codec_information.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ 12 | #define MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ 13 | 14 | #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" 15 | #include "modules/rtp_rtcp/source/rtp_utility.h" 16 | 17 | namespace webrtc { 18 | class VideoCodecInformation { 19 | public: 20 | virtual void Reset() = 0; 21 | 22 | virtual RtpVideoCodecTypes Type() = 0; 23 | virtual ~VideoCodecInformation() {} 24 | }; 25 | } // namespace webrtc 26 | 27 | #endif // MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ 28 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_device/audio_device_config.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_ 12 | #define AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_ 13 | 14 | // Enumerators 15 | // 16 | enum { GET_MIC_VOLUME_INTERVAL_MS = 1000 }; 17 | 18 | // Platform specifics 19 | // 20 | #if defined(_WIN32) 21 | #if (_MSC_VER >= 1400) 22 | #if !defined(WEBRTC_DUMMY_FILE_DEVICES) 23 | // Windows Core Audio is the default audio layer in Windows. 24 | // Only supported for VS 2005 and higher. 25 | #define WEBRTC_WINDOWS_CORE_AUDIO_BUILD 26 | #endif 27 | #endif 28 | #endif 29 | 30 | #endif // AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_ 31 | 32 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/video_codecs/BUILD.gn: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 2 | # 3 | # Use of this source code is governed by a BSD-style license 4 | # that can be found in the LICENSE file in the root of the source 5 | # tree. An additional intellectual property rights grant can be found 6 | # in the file PATENTS. All contributing project authors may 7 | # be found in the AUTHORS file in the root of the source tree. 8 | 9 | import("../../webrtc.gni") 10 | if (is_android) { 11 | import("//build/config/android/config.gni") 12 | import("//build/config/android/rules.gni") 13 | } 14 | 15 | rtc_source_set("video_codecs_api") { 16 | sources = [ 17 | "sdp_video_format.h", 18 | "video_decoder.h", 19 | "video_decoder_factory.h", 20 | "video_encoder.cc", 21 | "video_encoder.h", 22 | "video_encoder_factory.h", 23 | ] 24 | 25 | deps = [ 26 | "..:optional", 27 | "..:video_frame_api", 28 | "../..:webrtc_common", 29 | "../../common_video", 30 | "../../rtc_base:rtc_base_approved", 31 | ] 32 | } 33 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/transient/test/plotDetection.m: -------------------------------------------------------------------------------- 1 | % 2 | % Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | % 4 | % Use of this source code is governed by a BSD-style license 5 | % that can be found in the LICENSE file in the root of the source 6 | % tree. An additional intellectual property rights grant can be found 7 | % in the file PATENTS. All contributing project authors may 8 | % be found in the AUTHORS file in the root of the source tree. 9 | % 10 | 11 | function [] = plotDetection(PCMfile, DATfile, fs, chunkSize) 12 | %[] = plotDetection(PCMfile, DATfile, fs, chunkSize) 13 | % 14 | %Plots the signal alongside the detection values. 15 | % 16 | %PCMfile: The file of the input signal in PCM format. 17 | %DATfile: The file containing the detection values in binary float format. 18 | %fs: The sample rate of the signal in Hertz. 19 | %chunkSize: The chunk size used to compute the detection values in seconds. 20 | [x, tx] = readPCM(PCMfile, fs); 21 | [d, td] = readDetection(DATfile, fs, chunkSize); 22 | plot(tx, x, td, d); 23 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ 12 | #define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ 13 | 14 | namespace webrtc { 15 | 16 | // NOTE: This struct is still under development and may change without notice. 17 | struct AudioEncoderG722Config { 18 | bool IsOk() const { 19 | return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1; 20 | } 21 | int frame_size_ms = 20; 22 | int num_channels = 1; 23 | }; 24 | 25 | } // namespace webrtc 26 | 27 | #endif // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ 28 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/aec3/aec3_common.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_processing/aec3/aec3_common.h" 12 | 13 | #include "typedefs.h" // NOLINT(build/include) 14 | #include "system_wrappers/include/cpu_features_wrapper.h" 15 | 16 | namespace webrtc { 17 | 18 | Aec3Optimization DetectOptimization() { 19 | #if defined(WEBRTC_ARCH_X86_FAMILY) 20 | if (WebRtc_GetCPUInfo(kSSE2) != 0) { 21 | return Aec3Optimization::kSse2; 22 | } 23 | #endif 24 | 25 | #if defined(WEBRTC_HAS_NEON) 26 | return Aec3Optimization::kNeon; 27 | #endif 28 | 29 | return Aec3Optimization::kNone; 30 | } 31 | 32 | } // namespace webrtc 33 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/test/conversational_speech/config.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/audio_processing/test/conversational_speech/config.h" 12 | 13 | namespace webrtc { 14 | namespace test { 15 | namespace conversational_speech { 16 | 17 | const std::string& Config::audiotracks_path() const { 18 | return audiotracks_path_; 19 | } 20 | 21 | const std::string& Config::timing_filepath() const { 22 | return timing_filepath_; 23 | } 24 | 25 | const std::string& Config::output_path() const { 26 | return output_path_; 27 | } 28 | 29 | } // namespace conversational_speech 30 | } // namespace test 31 | } // namespace webrtc 32 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_VIDEO_CODING_UTILITY_VP9_UNCOMPRESSED_HEADER_PARSER_H_ 12 | #define MODULES_VIDEO_CODING_UTILITY_VP9_UNCOMPRESSED_HEADER_PARSER_H_ 13 | 14 | #include 15 | #include 16 | 17 | namespace webrtc { 18 | 19 | namespace vp9 { 20 | 21 | // Gets the QP, QP range: [0, 255]. 22 | // Returns true on success, false otherwise. 23 | bool GetQp(const uint8_t* buf, size_t length, int* qp); 24 | 25 | } // namespace vp9 26 | 27 | } // namespace webrtc 28 | 29 | #endif // MODULES_VIDEO_CODING_UTILITY_VP9_UNCOMPRESSED_HEADER_PARSER_H_ 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/api/ortc/mediadescription_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "api/ortc/mediadescription.h" 12 | #include "test/gtest.h" 13 | 14 | namespace webrtc { 15 | 16 | class MediaDescriptionTest : public testing::Test {}; 17 | 18 | TEST_F(MediaDescriptionTest, CreateMediaDescription) { 19 | MediaDescription m("a"); 20 | EXPECT_EQ("a", m.mid()); 21 | } 22 | 23 | TEST_F(MediaDescriptionTest, AddSdesParam) { 24 | MediaDescription m("a"); 25 | m.sdes_params().push_back(cricket::CryptoParams()); 26 | const std::vector& params = m.sdes_params(); 27 | EXPECT_EQ(1u, params.size()); 28 | } 29 | 30 | } // namespace webrtc 31 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/events/rtc_event_logging_started.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_LOGGING_STARTED_H_ 12 | #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_LOGGING_STARTED_H_ 13 | 14 | #include "logging/rtc_event_log/events/rtc_event.h" 15 | 16 | namespace webrtc { 17 | 18 | class RtcEventLoggingStarted final : public RtcEvent { 19 | public: 20 | ~RtcEventLoggingStarted() override = default; 21 | 22 | Type GetType() const override; 23 | 24 | bool IsConfigEvent() const override; 25 | }; 26 | 27 | } // namespace webrtc 28 | 29 | #endif // LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_LOGGING_STARTED_H_ 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/logging/rtc_event_log/events/rtc_event_logging_stopped.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_LOGGING_STOPPED_H_ 12 | #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_LOGGING_STOPPED_H_ 13 | 14 | #include "logging/rtc_event_log/events/rtc_event.h" 15 | 16 | namespace webrtc { 17 | 18 | class RtcEventLoggingStopped final : public RtcEvent { 19 | public: 20 | ~RtcEventLoggingStopped() override = default; 21 | 22 | Type GetType() const override; 23 | 24 | bool IsConfigEvent() const override; 25 | }; 26 | 27 | } // namespace webrtc 28 | 29 | #endif // LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_LOGGING_STOPPED_H_ 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /****************************************************************** 12 | 13 | iLBC Speech Coder ANSI-C Source Code 14 | 15 | WebRtcIlbcfix_IndexConvDec.h 16 | 17 | ******************************************************************/ 18 | 19 | #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_ 20 | #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_ 21 | 22 | #include "defines.h" 23 | 24 | void WebRtcIlbcfix_IndexConvDec( 25 | int16_t *index /* (i/o) Codebook indexes */ 26 | ); 27 | 28 | #endif 29 | -------------------------------------------------------------------------------- /webrtc-ns3/model/webrtctrace.h: -------------------------------------------------------------------------------- 1 | #ifndef MY_TRACE_H 2 | #define MY_TRACE_H 3 | #include 4 | #include 5 | #include 6 | namespace ns3 7 | { 8 | enum WebrtcTraceEnable:uint8_t{ 9 | E_WEBRTC_RATE=0x01, 10 | E_WEBRTC_DELAY=0x02, 11 | E_WEBRTC_FEEDBACK=0x04, 12 | E_WEBRTC_ALL=E_WEBRTC_RATE|E_WEBRTC_DELAY|E_WEBRTC_FEEDBACK, 13 | }; 14 | class WebrtcTrace 15 | { 16 | public: 17 | WebrtcTrace(){} 18 | ~WebrtcTrace(); 19 | void Log(std::string name,uint8_t enable); 20 | void RateTrace(float rate); 21 | void DelayTrace(uint64_t delay); 22 | void OnFeedback(uint32_t id,uint32_t seq); 23 | private: 24 | void Close(); 25 | void OpenTraceRateFile(std::string name); 26 | void CloseTraceRateFile(); 27 | void OpenTraceDelayFile(std::string name); 28 | void CloseTraceDelayFile(); 29 | void OpenTraceFeedbackFile(std::string name); 30 | void CloseTraceFeedbackFile(); 31 | std::fstream m_traceRateFile;// 32 | std::fstream m_traceDelayFile; 33 | std::fstream m_traceFeedback; 34 | }; 35 | } 36 | #endif 37 | -------------------------------------------------------------------------------- /webrtc-results/bw_plot.sh: -------------------------------------------------------------------------------- 1 | #! /bin/sh 2 | instance=4 3 | algo=webrtc 4 | file1=${instance}_${algo}_1_rate.txt 5 | file2=${instance}_${algo}_2_rate.txt 6 | file3=${instance}_${algo}_3_rate.txt 7 | output=${algo}_${instance} 8 | gnuplot< 15 | 16 | namespace webrtc { 17 | 18 | // Decides the sample rate of a mixing iteration given the preferred 19 | // sample rates of the sources. 20 | class OutputRateCalculator { 21 | public: 22 | virtual int CalculateOutputRate( 23 | const std::vector& preferred_sample_rates) = 0; 24 | virtual ~OutputRateCalculator() {} 25 | }; 26 | 27 | } // namespace webrtc 28 | 29 | #endif // MODULES_AUDIO_MIXER_OUTPUT_RATE_CALCULATOR_H_ 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/vad/common.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_AUDIO_PROCESSING_VAD_COMMON_H_ 12 | #define MODULES_AUDIO_PROCESSING_VAD_COMMON_H_ 13 | 14 | #include 15 | 16 | static const int kSampleRateHz = 16000; 17 | static const size_t kLength10Ms = kSampleRateHz / 100; 18 | static const size_t kMaxNumFrames = 4; 19 | 20 | struct AudioFeatures { 21 | double log_pitch_gain[kMaxNumFrames]; 22 | double pitch_lag_hz[kMaxNumFrames]; 23 | double spectral_peak[kMaxNumFrames]; 24 | double rms[kMaxNumFrames]; 25 | size_t num_frames; 26 | bool silence; 27 | }; 28 | 29 | #endif // MODULES_AUDIO_PROCESSING_VAD_COMMON_H_ 30 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/win/cursor_unittest_resources.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_ 12 | #define MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_ 13 | 14 | #define IDD_CURSOR1_24BPP 101 15 | #define IDD_CURSOR1_32BPP 102 16 | #define IDD_CURSOR1_8BPP 103 17 | 18 | #define IDD_CURSOR2_1BPP 104 19 | #define IDD_CURSOR2_32BPP 105 20 | 21 | #define IDD_CURSOR3_4BPP 106 22 | #define IDD_CURSOR3_32BPP 107 23 | 24 | #endif // MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_ 25 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/system_wrappers/source/sleep.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | // An OS-independent sleep function. 11 | 12 | #include "system_wrappers/include/sleep.h" 13 | 14 | #ifdef _WIN32 15 | // For Sleep() 16 | #include 17 | #else 18 | // For nanosleep() 19 | #include 20 | #endif 21 | 22 | namespace webrtc { 23 | 24 | void SleepMs(int msecs) { 25 | #ifdef _WIN32 26 | Sleep(msecs); 27 | #else 28 | struct timespec short_wait; 29 | struct timespec remainder; 30 | short_wait.tv_sec = msecs / 1000; 31 | short_wait.tv_nsec = (msecs % 1000) * 1000 * 1000; 32 | nanosleep(&short_wait, &remainder); 33 | #endif 34 | } 35 | 36 | } // namespace webrtc 37 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/audio_processing/transient/test/readDetection.m: -------------------------------------------------------------------------------- 1 | % 2 | % Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 | % 4 | % Use of this source code is governed by a BSD-style license 5 | % that can be found in the LICENSE file in the root of the source 6 | % tree. An additional intellectual property rights grant can be found 7 | % in the file PATENTS. All contributing project authors may 8 | % be found in the AUTHORS file in the root of the source tree. 9 | % 10 | 11 | function [d, t] = readDetection(file, fs, chunkSize) 12 | %[d, t] = readDetection(file, fs, chunkSize) 13 | % 14 | %Reads a detection signal from a DAT file. 15 | % 16 | %d: The detection signal. 17 | %t: The respective time vector. 18 | % 19 | %file: The DAT file where the detection signal is stored in float format. 20 | %fs: The signal sample rate in Hertz. 21 | %chunkSize: The chunk size used for the detection in seconds. 22 | fid = fopen(file); 23 | d = fread(fid, inf, 'float'); 24 | fclose(fid); 25 | t = 0:(1 / fs):(length(d) * chunkSize - 1 / fs); 26 | d = d(floor(t / chunkSize) + 1); 27 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/congestion_controller/congestion_controller_unittests_helper.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_CONGESTION_CONTROLLER_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_ 12 | #define MODULES_CONGESTION_CONTROLLER_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_ 13 | 14 | #include 15 | 16 | #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 17 | 18 | namespace webrtc { 19 | void ComparePacketFeedbackVectors(const std::vector& truth, 20 | const std::vector& input); 21 | } // namespace webrtc 22 | 23 | #endif // MODULES_CONGESTION_CONTROLLER_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_ 24 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/desktop_capture/test_utils.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_DESKTOP_CAPTURE_TEST_UTILS_H_ 12 | #define MODULES_DESKTOP_CAPTURE_TEST_UTILS_H_ 13 | 14 | #include "modules/desktop_capture/desktop_frame.h" 15 | 16 | namespace webrtc { 17 | 18 | // Clears a DesktopFrame |frame| by setting its data() into 0. 19 | void ClearDesktopFrame(DesktopFrame* frame); 20 | 21 | // Compares size() and data() of two DesktopFrames |left| and |right|. 22 | bool DesktopFrameDataEquals(const DesktopFrame& left, 23 | const DesktopFrame& right); 24 | 25 | } // namespace webrtc 26 | 27 | #endif // MODULES_DESKTOP_CAPTURE_TEST_UTILS_H_ 28 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/rtc_base/platform_thread_types.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef RTC_BASE_PLATFORM_THREAD_TYPES_H_ 12 | #define RTC_BASE_PLATFORM_THREAD_TYPES_H_ 13 | 14 | #if defined(WEBRTC_WIN) 15 | #include 16 | #include 17 | #elif defined(WEBRTC_POSIX) 18 | #include 19 | #include 20 | #endif 21 | 22 | namespace rtc { 23 | #if defined(WEBRTC_WIN) 24 | typedef DWORD PlatformThreadId; 25 | typedef DWORD PlatformThreadRef; 26 | #elif defined(WEBRTC_POSIX) 27 | typedef pid_t PlatformThreadId; 28 | typedef pthread_t PlatformThreadRef; 29 | #endif 30 | } // namespace rtc 31 | 32 | #endif // RTC_BASE_PLATFORM_THREAD_TYPES_H_ 33 | -------------------------------------------------------------------------------- /webrtc-lib/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "modules/rtp_rtcp/source/rtcp_nack_stats.h" 12 | 13 | #include "modules/include/module_common_types.h" 14 | 15 | namespace webrtc { 16 | 17 | RtcpNackStats::RtcpNackStats() 18 | : max_sequence_number_(0), 19 | requests_(0), 20 | unique_requests_(0) {} 21 | 22 | void RtcpNackStats::ReportRequest(uint16_t sequence_number) { 23 | if (requests_ == 0 || 24 | IsNewerSequenceNumber(sequence_number, max_sequence_number_)) { 25 | max_sequence_number_ = sequence_number; 26 | ++unique_requests_; 27 | } 28 | ++requests_; 29 | } 30 | 31 | } // namespace webrtc 32 | --------------------------------------------------------------------------------