├── tools ├── lsan │ └── OWNERS ├── perf │ ├── __init__.py │ └── perf ├── cpu │ ├── OWNERS │ └── README ├── valgrind-webrtc │ ├── drmemory │ │ └── OWNERS │ ├── memcheck │ │ ├── OWNERS │ │ ├── suppressions_mac.txt │ │ └── suppressions_win32.txt │ └── gtest_exclude │ │ ├── OWNERS │ │ ├── modules_unittests.gtest-memcheck.txt │ │ ├── libjingle_media_unittest.gtest-memcheck_mac.txt │ │ ├── system_wrappers_unittests.gtest-drmemory_win32.txt │ │ ├── video_engine_tests.gtest-memcheck.txt │ │ ├── common_video_unittests.gtest-drmemory_win32.txt │ │ ├── libjingle_unittest.gtest-memcheck.txt │ │ ├── libjingle_p2p_unittest.gtest-drmemory_win32.txt │ │ ├── libjingle_media_unittest.gtest-memcheck.txt │ │ ├── rtc_unittests.gtest-drmemory.txt │ │ ├── libjingle_peerconnection_unittest.gtest-drmemory_win32.txt │ │ ├── modules_tests.gtest-memcheck.txt │ │ ├── libjingle_unittest.gtest-memcheck_mac.txt │ │ ├── libjingle_p2p_unittest.gtest-memcheck.txt │ │ ├── modules_tests.gtest-drmemory.txt │ │ ├── rtc_unittests.gtest-memcheck.txt │ │ ├── modules_unittests.gtest-drmemory_win32.txt │ │ └── video_engine_tests.gtest-drmemory_win32.txt ├── python_charts │ ├── OWNERS │ ├── gviz_api.py │ ├── app.yaml │ └── webrtc │ │ └── __init__.py ├── msan │ ├── OWNERS │ └── blacklist.txt ├── OWNERS ├── whitespace.txt ├── .gitignore ├── codereview.settings └── DEPS ├── resources ├── OWNERS ├── att-uplink.rx.sha1 ├── ref03.aecdump.sha1 ├── att-downlink.rx.sha1 ├── audioproc.aecdump.sha1 ├── e2e_audio_in.pcm.sha1 ├── far16_stereo.pcm.sha1 ├── far32_stereo.pcm.sha1 ├── far44_stereo.pcm.sha1 ├── far48_stereo.pcm.sha1 ├── far8_stereo.pcm.sha1 ├── foreman_cif.yuv.sha1 ├── near16_stereo.pcm.sha1 ├── near32_stereo.pcm.sha1 ├── near44_stereo.pcm.sha1 ├── near48_stereo.pcm.sha1 ├── near8_stereo.pcm.sha1 ├── paris_qcif.yuv.sha1 ├── sprint-downlink.rx.sha1 ├── sprint-uplink.rx.sha1 ├── synthetic-trace.rx.sha1 ├── tmobile-uplink.rx.sha1 ├── audio_coding │ ├── F00.BIT20.sha1 │ ├── F00.BIT30.sha1 │ ├── F00.INP.sha1 │ ├── F00.OUT20.sha1 │ ├── F00.OUT30.sha1 │ ├── F01.BIT20.sha1 │ ├── F01.BIT30.sha1 │ ├── F01.INP.sha1 │ ├── F01.OUT20.sha1 │ ├── F01.OUT30.sha1 │ ├── F02.BIT20.sha1 │ ├── F02.BIT30.sha1 │ ├── F02.INP.sha1 │ ├── F02.OUT20.sha1 │ ├── F02.OUT30.sha1 │ ├── F03.BIT20.sha1 │ ├── F03.BIT30.sha1 │ ├── F03.INP.sha1 │ ├── F03.OUT20.sha1 │ ├── F03.OUT30.sha1 │ ├── F04.BIT20.sha1 │ ├── F04.BIT30.sha1 │ ├── F04.INP.sha1 │ ├── F04.OUT20.sha1 │ ├── F04.OUT30.sha1 │ ├── F05.BIT20.sha1 │ ├── F05.BIT30.sha1 │ ├── F05.INP.sha1 │ ├── F05.OUT20.sha1 │ ├── F05.OUT30.sha1 │ ├── F06.BIT20.sha1 │ ├── F06.BIT30.sha1 │ ├── F06.INP.sha1 │ ├── F06.OUT20.sha1 │ ├── F06.OUT30.sha1 │ ├── clean.chn.sha1 │ ├── tlm10.chn.sha1 │ ├── F00_tlm10.OUT20.sha1 │ ├── F00_tlm10.OUT30.sha1 │ ├── F01_tlm10.OUT20.sha1 │ ├── F01_tlm10.OUT30.sha1 │ ├── F02_tlm10.OUT20.sha1 │ ├── F02_tlm10.OUT30.sha1 │ ├── neteq_rtcp_stats.dat.sha1 │ ├── testfile32kHz.pcm.sha1 │ ├── teststereo32kHz.pcm.sha1 │ ├── music_stereo_48kHz.pcm.sha1 │ ├── neteq4_network_stats.dat.sha1 │ ├── neteq4_rtcp_stats.dat.sha1 │ ├── neteq4_universal_ref.pcm.sha1 │ ├── neteq_network_stats.dat.sha1 │ ├── neteq_universal_new.rtp.sha1 │ ├── neteq_universal_ref.pcm.sha1 │ ├── speech_mono_16kHz.pcm.sha1 │ ├── speech_mono_32_48kHz.pcm.sha1 │ ├── neteq_network_stats_win_32.dat.sha1 │ ├── neteq_universal_ref_win_32.pcm.sha1 │ ├── neteq4_network_stats_win_32.dat.sha1 │ ├── neteq4_universal_ref_win_32.pcm.sha1 │ ├── neteq4_universal_ref_win_64.pcm.sha1 │ └── READ.ME ├── foreman_cif_short.yuv.sha1 ├── images │ ├── webrtc_logo.jpg.sha1 │ ├── renderStartImage.jpg.sha1 │ ├── captureDeviceImage.jpg.sha1 │ └── renderTimeoutImage.jpg.sha1 ├── short_mixed_mono_48.dat.sha1 ├── short_mixed_mono_48.pcm.sha1 ├── speech_and_misc_wb.pcm.sha1 ├── tmobile-downlink.rx.sha1 ├── verizon3g-downlink.rx.sha1 ├── verizon3g-uplink.rx.sha1 ├── verizon4g-downlink.rx.sha1 ├── verizon4g-uplink.rx.sha1 ├── rtp_rtcp │ ├── H263_CIF_IFRAME.bin.sha1 │ ├── H263_CIF_PFRAME.bin.sha1 │ ├── H263_QCIF_IFRAME.bin.sha1 │ ├── RTCPPacketTMMBR0.bin.sha1 │ ├── RTCPPacketTMMBR1.bin.sha1 │ ├── RTCPPacketTMMBR2.bin.sha1 │ ├── RTCPPacketTMMBR3.bin.sha1 │ ├── RTCPPacketTMMBR4.bin.sha1 │ ├── RTCPPacketTMMBR4_1.bin.sha1 │ ├── RTCPPacketTMMBR4_2.bin.sha1 │ ├── RTCPPacketTMMBR5.bin.sha1 │ ├── H263Foreman_CIF_Iframe.bin.sha1 │ └── H263Foreman_CIF_Pframe.bin.sha1 ├── short_mixed_stereo_48.dat.sha1 ├── short_mixed_stereo_48.pcm.sha1 ├── video_coding │ ├── pltype103.rtp.sha1 │ ├── ssrcs-2.pcap.sha1 │ ├── ssrcs-3.pcap.sha1 │ ├── frame-loopback.pcap.sha1 │ ├── frame-ethernet-ii.pcap.sha1 │ └── pltype103_header_only.rtp.sha1 ├── voice_engine │ ├── audio_long8.pcm.sha1 │ ├── audio_tiny8.wav.sha1 │ ├── audio_long16.pcm.sha1 │ ├── audio_long16.wav.sha1 │ ├── audio_long8mulaw.wav.sha1 │ ├── audio_short16.pcm.sha1 │ ├── audio_tiny11.wav.sha1 │ ├── audio_tiny16.wav.sha1 │ ├── audio_tiny22.wav.sha1 │ ├── audio_tiny32.wav.sha1 │ ├── audio_tiny44.wav.sha1 │ ├── audio_tiny48.wav.sha1 │ ├── audio_long16noise.pcm.sha1 │ └── audio_long16big_endian.pcm.sha1 ├── audio_device │ ├── audio_short16.pcm.sha1 │ ├── audio_short44.pcm.sha1 │ ├── audio_short48.pcm.sha1 │ └── audio_short8.pcm.sha1 ├── audio_processing │ ├── agc │ │ ├── agc_vad.dat.sha1 │ │ ├── agc_audio.pcm.sha1 │ │ ├── agc_pitch_lag.dat.sha1 │ │ ├── agc_pitch_gain.dat.sha1 │ │ ├── agc_spectral_peak.dat.sha1 │ │ ├── agc_voicing_prob.dat.sha1 │ │ ├── agc_no_circular_buffer.dat.sha1 │ │ └── agc_with_circular_buffer.dat.sha1 │ └── transient │ │ ├── wpd0.dat.sha1 │ │ ├── wpd1.dat.sha1 │ │ ├── wpd2.dat.sha1 │ │ ├── wpd3.dat.sha1 │ │ ├── wpd4.dat.sha1 │ │ ├── wpd5.dat.sha1 │ │ ├── wpd6.dat.sha1 │ │ ├── wpd7.dat.sha1 │ │ ├── audio16kHz.pcm.sha1 │ │ ├── audio32kHz.pcm.sha1 │ │ ├── audio48kHz.pcm.sha1 │ │ ├── audio8kHz.pcm.sha1 │ │ ├── detect16kHz.dat.sha1 │ │ ├── detect32kHz.dat.sha1 │ │ ├── detect48kHz.dat.sha1 │ │ ├── detect8kHz.dat.sha1 │ │ ├── double-utils.dat.sha1 │ │ ├── float-utils.dat.sha1 │ │ ├── suppressed16kHz.pcm.sha1 │ │ ├── suppressed32kHz.pcm.sha1 │ │ ├── suppressed8kHz.pcm.sha1 │ │ └── ajm-macbook-1-spke16m.pcm.sha1 ├── deflicker_before_cif_short.yuv.sha1 ├── video_engine │ ├── renderStartImage.jpg.sha1 │ └── renderTimeoutImage.jpg.sha1 ├── foremanColorEnhanced_cif_short.yuv.sha1 ├── utility │ ├── encapsulated_pcm16b_8khz.wav.sha1 │ └── encapsulated_pcmu_8khz.wav.sha1 └── remote_bitrate_estimator │ ├── VideoSendersTest_BweTest_Multi1_1_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_Multi1_1_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyChoke_0_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyChoke_0_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyChoke_1_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyChoke_1_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyDelay_0_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyDelay_0_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyLoss_0_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_SteadyLoss_0_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke1_0_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke1_0_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke1_1_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke1_1_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke2_0_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke2_0_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke2_1_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingChoke2_1_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingDelay1_0_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingDelay1_0_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingLoss1_0_AST.bin.sha1 │ ├── VideoSendersTest_BweTest_IncreasingLoss1_0_TOF.bin.sha1 │ ├── VideoSendersTest_BweTest_UnlimitedSpeed_0_AST.bin.sha1 │ └── VideoSendersTest_BweTest_UnlimitedSpeed_0_TOF.bin.sha1 ├── COPYING ├── PATENTS ├── LICENSE ├── webrtc ├── build │ ├── download_vs_toolchain.py │ ├── OWNERS │ └── no_op.cc ├── examples │ ├── OWNERS │ └── android │ │ ├── OWNERS │ │ ├── media_demo │ │ ├── res │ │ │ ├── drawable │ │ │ │ └── logo.png │ │ │ ├── menu │ │ │ │ └── main_activity_actions.xml │ │ │ ├── values │ │ │ │ ├── integers.xml │ │ │ │ └── bools.xml │ │ │ └── layout │ │ │ │ └── dropdownitems.xml │ │ ├── src │ │ │ └── org │ │ │ │ └── webrtc │ │ │ │ └── webrtcdemo │ │ │ │ ├── MediaEngineObserver.java │ │ │ │ ├── MenuStateProvider.java │ │ │ │ └── NativeWebRtcContextRegistry.java │ │ └── project.properties │ │ └── opensl_loopback │ │ ├── res │ │ ├── drawable │ │ │ └── logo.png │ │ └── values │ │ │ └── strings.xml │ │ └── project.properties ├── tools │ ├── e2e_quality │ │ └── audio │ │ │ ├── perf │ │ │ ├── daemon.conf │ │ │ └── default.pa │ ├── loopback_test │ │ ├── OWNERS │ │ ├── README │ │ └── run-server.sh │ ├── rtcbot │ │ ├── OWNERS │ │ ├── bot │ │ │ └── browser │ │ │ │ └── index.html │ │ ├── rtcBotReportVisualizer │ │ │ └── index.html │ │ └── test │ │ │ └── ping_pong.js │ ├── OWNERS │ └── barcode_tools │ │ └── DEPS ├── modules │ ├── audio_device │ │ ├── android │ │ │ └── OWNERS │ │ ├── test │ │ │ └── android │ │ │ │ └── audio_device_android_test │ │ │ │ ├── src │ │ │ │ └── org │ │ │ │ │ └── webrtc │ │ │ │ │ └── voiceengine │ │ │ │ │ └── AudioDeviceAndroid.java │ │ │ │ ├── res │ │ │ │ ├── drawable │ │ │ │ │ └── icon.png │ │ │ │ ├── values │ │ │ │ │ └── strings.xml │ │ │ │ └── layout │ │ │ │ │ └── main.xml │ │ │ │ ├── .classpath │ │ │ │ └── default.properties │ │ ├── main │ │ │ ├── source │ │ │ │ ├── OWNERS │ │ │ │ └── audio_device.gypi │ │ │ └── interface │ │ │ │ └── audio_device.h │ │ ├── OWNERS │ │ └── dummy │ │ │ └── audio_device_utility_dummy.cc │ ├── audio_coding │ │ ├── OWNERS │ │ ├── codecs │ │ │ ├── isac │ │ │ │ ├── fix │ │ │ │ │ ├── test │ │ │ │ │ │ └── QA │ │ │ │ │ │ │ ├── ChannelFiles.txt │ │ │ │ │ │ │ ├── InputFilesFew.txt │ │ │ │ │ │ │ ├── ListOfTestCases.xls │ │ │ │ │ │ │ ├── diffiSACPLC.txt │ │ │ │ │ │ │ ├── InputFiles.txt │ │ │ │ │ │ │ └── runiSACPLC.txt │ │ │ │ │ └── source │ │ │ │ │ │ └── OWNERS │ │ │ │ ├── OWNERS │ │ │ │ └── main │ │ │ │ │ └── source │ │ │ │ │ └── OWNERS │ │ │ ├── OWNERS │ │ │ ├── cng │ │ │ │ └── OWNERS │ │ │ ├── g711 │ │ │ │ └── OWNERS │ │ │ ├── g722 │ │ │ │ └── OWNERS │ │ │ ├── ilbc │ │ │ │ └── OWNERS │ │ │ ├── opus │ │ │ │ └── OWNERS │ │ │ ├── pcm16b │ │ │ │ └── OWNERS │ │ │ ├── tools │ │ │ │ └── OWNERS │ │ │ └── audio_encoder.cc │ │ ├── main │ │ │ ├── OWNERS │ │ │ ├── acm2 │ │ │ │ ├── OWNERS │ │ │ │ └── acm_neteq_unittest.cc │ │ │ └── test │ │ │ │ └── ACMTest.h │ │ └── neteq │ │ │ ├── OWNERS │ │ │ ├── post_decode_vad_unittest.cc │ │ │ └── random_vector_unittest.cc │ ├── audio_processing │ │ ├── test │ │ │ └── android │ │ │ │ └── apmtest │ │ │ │ ├── jni │ │ │ │ └── Application.mk │ │ │ │ ├── res │ │ │ │ └── values │ │ │ │ │ └── strings.xml │ │ │ │ └── default.properties │ │ ├── OWNERS │ │ ├── transient │ │ │ └── test │ │ │ │ ├── readPCM.m │ │ │ │ ├── plotDetection.m │ │ │ │ └── readDetection.m │ │ ├── utility │ │ │ └── fft4g.h │ │ └── beamformer │ │ │ └── mock_beamformer.cc │ ├── video_coding │ │ ├── codecs │ │ │ ├── OWNERS │ │ │ ├── test │ │ │ │ └── OWNERS │ │ │ ├── tools │ │ │ │ └── OWNERS │ │ │ ├── vp8 │ │ │ │ └── OWNERS │ │ │ └── i420 │ │ │ │ └── main │ │ │ │ └── source │ │ │ │ ├── OWNERS │ │ │ │ └── i420.gypi │ │ ├── OWNERS │ │ ├── utility │ │ │ └── OWNERS │ │ └── main │ │ │ ├── source │ │ │ └── OWNERS │ │ │ └── test │ │ │ └── release_test.h │ ├── audio_conference_mixer │ │ ├── OWNERS │ │ └── source │ │ │ └── OWNERS │ ├── utility │ │ ├── OWNERS │ │ └── source │ │ │ └── OWNERS │ ├── media_file │ │ ├── OWNERS │ │ └── source │ │ │ └── OWNERS │ ├── video_processing │ │ ├── OWNERS │ │ └── main │ │ │ ├── source │ │ │ └── OWNERS │ │ │ └── test │ │ │ └── unit_test │ │ │ └── writeYUV420file.m │ ├── desktop_capture │ │ ├── win │ │ │ └── cursor_test_data │ │ │ │ ├── 1_8bpp.cur │ │ │ │ ├── 2_1bpp.cur │ │ │ │ ├── 3_4bpp.cur │ │ │ │ ├── 1_24bpp.cur │ │ │ │ ├── 1_32bpp.cur │ │ │ │ ├── 2_32bpp.cur │ │ │ │ └── 3_32bpp.cur │ │ ├── OWNERS │ │ ├── screen_capturer_null.cc │ │ ├── window_capturer.cc │ │ └── mouse_cursor_shape.h │ ├── video_render │ │ ├── test │ │ │ └── testAPI │ │ │ │ ├── renderStartImage.bmp │ │ │ │ ├── testAPI_android.cc │ │ │ │ └── testAPI.h │ │ └── OWNERS │ ├── rtp_rtcp │ │ ├── OWNERS │ │ ├── source │ │ │ └── OWNERS │ │ └── test │ │ │ ├── OWNERS │ │ │ └── testFec │ │ │ └── OWNERS │ ├── OWNERS │ ├── pacing │ │ └── OWNERS │ ├── bitrate_controller │ │ └── OWNERS │ ├── remote_bitrate_estimator │ │ └── OWNERS │ └── video_capture │ │ ├── OWNERS │ │ ├── ensure_initialized.h │ │ └── test │ │ └── video_capture_main_mac.mm ├── test │ ├── channel_transport │ │ └── OWNERS │ ├── OWNERS │ ├── run_all_unittests.cc │ ├── run_test.cc │ ├── run_loop.cc │ ├── run_test.h │ ├── testsupport │ │ └── always_passing_unittest.cc │ ├── null_transport.cc │ ├── run_loop.h │ └── null_platform_renderer.cc ├── .gitignore ├── voice_engine │ ├── test │ │ ├── win_test │ │ │ ├── WinTest.aps │ │ │ ├── res │ │ │ │ ├── WinTest.ico │ │ │ │ └── WinTest.rc2 │ │ │ └── stdafx.cc │ │ ├── android │ │ │ └── android_test │ │ │ │ ├── res │ │ │ │ ├── drawable │ │ │ │ │ └── icon.png │ │ │ │ └── values │ │ │ │ │ └── strings.xml │ │ │ │ ├── .classpath │ │ │ │ ├── default.properties │ │ │ │ └── jni │ │ │ │ └── Application.mk │ │ └── auto_test │ │ │ ├── fixtures │ │ │ └── after_streaming_fixture.cc │ │ │ └── automated_mode.h │ ├── OWNERS │ └── channel_unittest.cc ├── video_engine │ ├── test │ │ ├── auto_test │ │ │ ├── android │ │ │ │ ├── res │ │ │ │ │ └── drawable │ │ │ │ │ │ └── logo.png │ │ │ │ ├── default.properties │ │ │ │ └── .classpath │ │ │ ├── OWNERS │ │ │ ├── source │ │ │ │ ├── vie_window_manager_factory_win.cc │ │ │ │ └── vie_window_manager_factory_linux.cc │ │ │ └── primitives │ │ │ │ └── fake_stdin.h │ │ └── libvietest │ │ │ └── OWNERS │ ├── OWNERS │ └── video_engine.gyp ├── system_wrappers │ ├── OWNERS │ └── source │ │ ├── OWNERS │ │ └── cpu_features_android.c ├── OWNERS ├── p2p │ ├── OWNERS │ └── base │ │ ├── udpport.h │ │ ├── sessionid.h │ │ └── common.h ├── base │ ├── OWNERS │ ├── asyncfile.cc │ ├── mathutils.h │ ├── win32socketinit.h │ ├── messagehandler.cc │ ├── stringdigest.h │ ├── basicdefs.h │ ├── proxyinfo.cc │ ├── openssl.h │ ├── sha1.h │ └── scoped_autorelease_pool.mm ├── libjingle │ ├── OWNERS │ └── examples │ │ └── call │ │ └── Info.plist ├── overrides │ └── OWNERS ├── sound │ ├── OWNERS │ ├── sound_tests.gypi │ ├── alsasymboltable.cc │ ├── platformsoundsystem.h │ └── linuxsoundsystem.cc ├── video │ └── OWNERS ├── README.chromium ├── common_video │ ├── OWNERS │ └── interface │ │ ├── video_image.h │ │ └── i420_video_frame.h ├── codereview.settings ├── common_audio │ └── OWNERS ├── common.gyp ├── supplement.gypi └── rtc_unittests.isolate ├── chromium ├── OWNERS ├── README └── .gclient ├── talk ├── examples │ ├── OWNERS │ ├── objc │ │ ├── Icon.png │ │ ├── AppRTCDemo │ │ │ ├── ios │ │ │ │ └── resources │ │ │ │ │ ├── Default-568h.png │ │ │ │ │ ├── Roboto-Regular.ttf │ │ │ │ │ ├── ic_clear_black_24dp.png │ │ │ │ │ ├── ic_call_end_black_24dp.png │ │ │ │ │ ├── ic_clear_black_24dp@2x.png │ │ │ │ │ └── ic_call_end_black_24dp@2x.png │ │ │ └── third_party │ │ │ │ └── SocketRocket │ │ │ │ └── LICENSE │ │ └── README │ ├── android │ │ ├── jni │ │ │ └── Android.mk │ │ ├── res │ │ │ ├── drawable-hdpi │ │ │ │ ├── disconnect.png │ │ │ │ ├── ic_launcher.png │ │ │ │ ├── ic_loopback_call.png │ │ │ │ ├── ic_action_full_screen.png │ │ │ │ └── ic_action_return_from_full_screen.png │ │ │ ├── drawable-ldpi │ │ │ │ ├── disconnect.png │ │ │ │ ├── ic_launcher.png │ │ │ │ ├── ic_loopback_call.png │ │ │ │ ├── ic_action_full_screen.png │ │ │ │ └── ic_action_return_from_full_screen.png │ │ │ ├── drawable-mdpi │ │ │ │ ├── disconnect.png │ │ │ │ ├── ic_launcher.png │ │ │ │ ├── ic_loopback_call.png │ │ │ │ ├── ic_action_full_screen.png │ │ │ │ └── ic_action_return_from_full_screen.png │ │ │ ├── drawable-xhdpi │ │ │ │ ├── disconnect.png │ │ │ │ ├── ic_launcher.png │ │ │ │ ├── ic_loopback_call.png │ │ │ │ ├── ic_action_full_screen.png │ │ │ │ └── ic_action_return_from_full_screen.png │ │ │ ├── values │ │ │ │ ├── styles.xml │ │ │ │ └── arrays.xml │ │ │ ├── values-v21 │ │ │ │ └── styles.xml │ │ │ └── menu │ │ │ │ └── connect_menu.xml │ │ ├── third_party │ │ │ └── autobanh │ │ │ │ ├── autobanh.jar │ │ │ │ └── NOTICE │ │ ├── project.properties │ │ └── ant.properties │ └── androidtests │ │ ├── README │ │ ├── AndroidManifest.xml │ │ ├── project.properties │ │ └── ant.properties ├── media │ ├── webrtc │ │ └── OWNERS │ ├── testdata │ │ ├── faces_I400.jpg │ │ ├── faces_I411.jpg │ │ ├── faces_I420.jpg │ │ ├── faces_I422.jpg │ │ ├── faces_I444.jpg │ │ ├── video.rtpdump │ │ ├── voice.rtpdump │ │ ├── 1.frame_plus_1.byte │ │ ├── faces.1280x720_P420.yuv │ │ ├── h264-svc-99-640x360.rtpdump │ │ └── captured-320x240-2s-48.frames │ └── base │ │ └── mutedvideocapturer.cc ├── app │ └── webrtc │ │ ├── objctests │ │ └── README │ │ ├── androidtests │ │ ├── jni │ │ │ └── Android.mk │ │ ├── res │ │ │ ├── drawable-hdpi │ │ │ │ └── ic_launcher.png │ │ │ ├── drawable-ldpi │ │ │ │ └── ic_launcher.png │ │ │ ├── drawable-mdpi │ │ │ │ └── ic_launcher.png │ │ │ ├── drawable-xhdpi │ │ │ │ └── ic_launcher.png │ │ │ └── values │ │ │ │ └── strings.xml │ │ ├── project.properties │ │ └── ant.properties │ │ └── OWNERS ├── build │ ├── OWNERS │ └── objc_app.plist ├── codereview.settings └── OWNERS ├── LICENSE_THIRD_PARTY ├── data ├── rtp_rtcp │ ├── H263_CIF_IFRAME.bin │ ├── H263_CIF_PFRAME.bin │ ├── H263_QCIF_IFRAME.bin │ ├── RTCPPacketTMMBR0.bin │ ├── RTCPPacketTMMBR1.bin │ ├── RTCPPacketTMMBR2.bin │ ├── RTCPPacketTMMBR3.bin │ ├── RTCPPacketTMMBR4.bin │ ├── RTCPPacketTMMBR5.bin │ ├── RTCPPacketTMMBR4_1.bin │ ├── RTCPPacketTMMBR4_2.bin │ ├── H263Foreman_CIF_Iframe.bin │ └── H263Foreman_CIF_Pframe.bin ├── voice_engine │ ├── audio_long16.pcm │ ├── audio_long16.wav │ ├── audio_long8.pcm │ ├── audio_short16.pcm │ ├── audio_tiny11.wav │ ├── audio_tiny16.wav │ ├── audio_tiny22.wav │ ├── audio_tiny32.wav │ ├── audio_tiny44.wav │ ├── audio_tiny48.wav │ ├── audio_tiny8.wav │ ├── audio_long16noise.pcm │ ├── audio_long8mulaw.wav │ ├── audio_long16big_endian.pcm │ └── stereo_rtp_files │ │ ├── rtpplay.exe │ │ ├── stereo_g729.rtp │ │ ├── stereo_pcmu.rtp │ │ ├── HRTF_pcm16wb.rtp │ │ ├── hrtf_g722_1C_48.rtp │ │ ├── stereo_pcm16wb.rtp │ │ ├── stereo_pcmu_vad.rtp │ │ ├── stereo_g729_jitter.rtp │ │ ├── stereo_pcmu_jitter.rtp │ │ ├── HRTF_pcm16wb_jitter.rtp │ │ ├── stereo_pcm16wb_jitter.rtp │ │ ├── stereo_pcmu_vad_jitter.rtp │ │ ├── hrtf_g722_1C_48_jitterT2.rtp │ │ ├── toggling_stereo_g729_pt18_pt125.rtp │ │ └── README.txt ├── common_video │ └── jpeg │ │ └── webrtc_logo.jpg └── audio_processing │ ├── output_data_fixed.pb │ ├── output_data_float.pb │ └── android │ ├── output_data_fixed.pb │ └── output_data_float.pb ├── .clang-format ├── third_party ├── gflags │ └── OWNERS ├── winsdk_samples │ ├── OWNERS │ └── README.webrtc └── gtest-parallel │ └── README.webrtc ├── drover.properties ├── OWNERS ├── codereview.settings ├── license_template.txt └── BUILD.gn /tools/lsan/OWNERS: -------------------------------------------------------------------------------- 1 | * 2 | -------------------------------------------------------------------------------- /tools/perf/__init__.py: -------------------------------------------------------------------------------- 1 | -------------------------------------------------------------------------------- /resources/OWNERS: -------------------------------------------------------------------------------- 1 | * 2 | 3 | -------------------------------------------------------------------------------- /tools/perf/perf: -------------------------------------------------------------------------------- 1 | ../../perf -------------------------------------------------------------------------------- /COPYING: -------------------------------------------------------------------------------- 1 | Refer to talk/COPYING. -------------------------------------------------------------------------------- /PATENTS: -------------------------------------------------------------------------------- 1 | Refer to src/PATENTS. 2 | -------------------------------------------------------------------------------- /LICENSE: -------------------------------------------------------------------------------- 1 | Refer to webrtc/LICENSE. 2 | -------------------------------------------------------------------------------- /tools/cpu/OWNERS: -------------------------------------------------------------------------------- 1 | tommi@webrtc.org 2 | -------------------------------------------------------------------------------- /webrtc/build/download_vs_toolchain.py: -------------------------------------------------------------------------------- 1 | -------------------------------------------------------------------------------- /chromium/OWNERS: -------------------------------------------------------------------------------- 1 | kjellander@webrtc.org 2 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/drmemory/OWNERS: -------------------------------------------------------------------------------- 1 | * 2 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/memcheck/OWNERS: -------------------------------------------------------------------------------- 1 | * 2 | -------------------------------------------------------------------------------- /webrtc/examples/OWNERS: -------------------------------------------------------------------------------- 1 | henrike@webrtc.org 2 | -------------------------------------------------------------------------------- /tools/python_charts/OWNERS: -------------------------------------------------------------------------------- 1 | kjellander@webrtc.org 2 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/OWNERS: -------------------------------------------------------------------------------- 1 | * 2 | 3 | -------------------------------------------------------------------------------- /webrtc/examples/android/OWNERS: -------------------------------------------------------------------------------- 1 | henrike@webrtc.org 2 | -------------------------------------------------------------------------------- /webrtc/tools/e2e_quality/audio/perf: -------------------------------------------------------------------------------- 1 | ../../../../tools/perf -------------------------------------------------------------------------------- /webrtc/tools/loopback_test/OWNERS: -------------------------------------------------------------------------------- 1 | andresp@webrtc.org 2 | -------------------------------------------------------------------------------- /tools/msan/OWNERS: -------------------------------------------------------------------------------- 1 | pbos@webrtc.org 2 | kjellander@webrtc.org 3 | -------------------------------------------------------------------------------- /resources/att-uplink.rx.sha1: -------------------------------------------------------------------------------- 1 | 8ea680a570df53fba206b984f38d4762481322eb -------------------------------------------------------------------------------- /resources/ref03.aecdump.sha1: -------------------------------------------------------------------------------- 1 | 04f9f47938efa99d0389672ff2d83c10f04a1752 -------------------------------------------------------------------------------- /talk/examples/OWNERS: -------------------------------------------------------------------------------- 1 | glaznev@webrtc.org 2 | tkchin@webrtc.org 3 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/android/OWNERS: -------------------------------------------------------------------------------- 1 | henrike@webrtc.org 2 | -------------------------------------------------------------------------------- /webrtc/test/channel_transport/OWNERS: -------------------------------------------------------------------------------- 1 | henrikg@webrtc.org 2 | 3 | -------------------------------------------------------------------------------- /resources/att-downlink.rx.sha1: -------------------------------------------------------------------------------- 1 | 2a4617e3497d8faa2fdccfa8964da26499eb13d6 -------------------------------------------------------------------------------- /resources/audioproc.aecdump.sha1: -------------------------------------------------------------------------------- 1 | 8b53334fec495185a82a759104240b2c01e449b6 -------------------------------------------------------------------------------- /resources/e2e_audio_in.pcm.sha1: -------------------------------------------------------------------------------- 1 | 892d292c1dcd3b5cee5c71d8fa5d21118fbfe3f7 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-------------------------------------------------------------------------------- /resources/far8_stereo.pcm.sha1: -------------------------------------------------------------------------------- 1 | 6f543f892cd3740a3a1366a486a892766d1f0a79 -------------------------------------------------------------------------------- /resources/foreman_cif.yuv.sha1: -------------------------------------------------------------------------------- 1 | db8cc13114cfe550fefa264ea70427e1fa4f9bba -------------------------------------------------------------------------------- /resources/near16_stereo.pcm.sha1: -------------------------------------------------------------------------------- 1 | 42ef52b80a02f28fde9ccc6299543bbde882bea9 -------------------------------------------------------------------------------- /resources/near32_stereo.pcm.sha1: -------------------------------------------------------------------------------- 1 | ee9615ce39f2746b1b6ded5b2e31cdbc4b86c401 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-------------------------------------------------------------------------------- /resources/verizon3g-uplink.rx.sha1: -------------------------------------------------------------------------------- 1 | 87d02bc510eac4b8b4a2453687c6b49f39aa481c -------------------------------------------------------------------------------- /resources/verizon4g-downlink.rx.sha1: -------------------------------------------------------------------------------- 1 | 07716dffe2905abde0ae7bec475106e476bc9b25 -------------------------------------------------------------------------------- /resources/verizon4g-uplink.rx.sha1: -------------------------------------------------------------------------------- 1 | 2d32cd78d75549a5d0795bb9fbe35a00663f949a -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/OWNERS: -------------------------------------------------------------------------------- 1 | per-file BUILD.gn=kjellander@webrtc.org 2 | -------------------------------------------------------------------------------- /webrtc/tools/e2e_quality/audio/daemon.conf: -------------------------------------------------------------------------------- 1 | resample-method = speex-float-9 2 | -------------------------------------------------------------------------------- /webrtc/tools/rtcbot/OWNERS: -------------------------------------------------------------------------------- 1 | andresp@webrtc.org 2 | houssainy@google.com 3 | -------------------------------------------------------------------------------- /resources/audio_coding/F00_tlm10.OUT20.sha1: -------------------------------------------------------------------------------- 1 | 96fb5327ff7a1fe87bd4512773ce7347b4d72888 -------------------------------------------------------------------------------- /resources/audio_coding/F00_tlm10.OUT30.sha1: -------------------------------------------------------------------------------- 1 | ea527e8e61241ea73265abba5765793511c42291 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| e3c189b500d92fd0f10cb4c770c298cee7008749 -------------------------------------------------------------------------------- /resources/audio_coding/neteq_universal_new.rtp.sha1: -------------------------------------------------------------------------------- 1 | 8dd7585937f1f3b17666035a7aec5c0431d4b30b -------------------------------------------------------------------------------- /resources/audio_coding/neteq_universal_ref.pcm.sha1: -------------------------------------------------------------------------------- 1 | 236da353f05d329f6c83e441b80dbfcc18706cfb -------------------------------------------------------------------------------- /resources/audio_coding/speech_mono_16kHz.pcm.sha1: -------------------------------------------------------------------------------- 1 | 89f191b706f8028e52ffd64525de1921eacd772a -------------------------------------------------------------------------------- /resources/audio_coding/speech_mono_32_48kHz.pcm.sha1: 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/resources/voice_engine/audio_long16noise.pcm.sha1: -------------------------------------------------------------------------------- 1 | 573e82889a25b1db477e64eb4389907397ba19c7 -------------------------------------------------------------------------------- /talk/media/webrtc/OWNERS: -------------------------------------------------------------------------------- 1 | mflodman@webrtc.org 2 | pbos@webrtc.org 3 | pthatcher@webrtc.org 4 | -------------------------------------------------------------------------------- /resources/audio_coding/neteq_network_stats_win_32.dat.sha1: -------------------------------------------------------------------------------- 1 | 343061419a64ca99323fc0d25a43149a5d40cf07 -------------------------------------------------------------------------------- /resources/audio_coding/neteq_universal_ref_win_32.pcm.sha1: -------------------------------------------------------------------------------- 1 | 1d2d353be4345d30506866ca32fa72825c6d65b8 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-------------------------------------------------------------------------------- 1 | See ../objc/README for information on what this is and how to use it. 2 | -------------------------------------------------------------------------------- /talk/examples/objc/Icon.png: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/talk/examples/objc/Icon.png -------------------------------------------------------------------------------- /webrtc/.gitignore: -------------------------------------------------------------------------------- 1 | # This file is for projects that checkout webrtc/ directly (e.g. Chromium). 2 | *.mk 3 | -------------------------------------------------------------------------------- /tools/OWNERS: -------------------------------------------------------------------------------- 1 | kjellander@webrtc.org 2 | phoglund@webrtc.org 3 | niklas.enbom@webrtc.org 4 | andrew@webrtc.org 5 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | marpan@webrtc.org 3 | henrik.lundin@webrtc.org 4 | -------------------------------------------------------------------------------- /data/rtp_rtcp/H263_CIF_IFRAME.bin: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/rtp_rtcp/H263_CIF_IFRAME.bin -------------------------------------------------------------------------------- /data/rtp_rtcp/H263_CIF_PFRAME.bin: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/rtp_rtcp/H263_CIF_PFRAME.bin -------------------------------------------------------------------------------- /data/rtp_rtcp/H263_QCIF_IFRAME.bin: 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-------------------------------------------------------------------------------- /data/common_video/jpeg/webrtc_logo.jpg: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/common_video/jpeg/webrtc_logo.jpg -------------------------------------------------------------------------------- /data/rtp_rtcp/H263Foreman_CIF_Iframe.bin: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/rtp_rtcp/H263Foreman_CIF_Iframe.bin -------------------------------------------------------------------------------- /data/rtp_rtcp/H263Foreman_CIF_Pframe.bin: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/rtp_rtcp/H263Foreman_CIF_Pframe.bin -------------------------------------------------------------------------------- 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/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_TOF.bin.sha1: -------------------------------------------------------------------------------- 1 | 95a356db2eb39fd1bbf1ece1cfbf673b1f893487 -------------------------------------------------------------------------------- /resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_AST.bin.sha1: -------------------------------------------------------------------------------- 1 | 26e5e14a114a869a0777458520e6c27b5791863d -------------------------------------------------------------------------------- /resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_TOF.bin.sha1: -------------------------------------------------------------------------------- 1 | 26e5e14a114a869a0777458520e6c27b5791863d -------------------------------------------------------------------------------- /talk/media/base/mutedvideocapturer.cc: -------------------------------------------------------------------------------- 1 | // TODO(pthatcher): Delete this file. Pulse won't work without it for 2 | // some reason. 3 | -------------------------------------------------------------------------------- /talk/media/testdata/1.frame_plus_1.byte: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/talk/media/testdata/1.frame_plus_1.byte -------------------------------------------------------------------------------- /data/audio_processing/output_data_fixed.pb: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/audio_processing/output_data_fixed.pb -------------------------------------------------------------------------------- /data/audio_processing/output_data_float.pb: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/audio_processing/output_data_float.pb -------------------------------------------------------------------------------- /webrtc/modules/audio_conference_mixer/OWNERS: -------------------------------------------------------------------------------- 1 | henrike@webrtc.org 2 | andrew@webrtc.org 3 | 4 | per-file BUILD.gn=kjellander@webrtc.org 5 | -------------------------------------------------------------------------------- /data/voice_engine/audio_long16big_endian.pcm: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/voice_engine/audio_long16big_endian.pcm -------------------------------------------------------------------------------- /talk/media/testdata/faces.1280x720_P420.yuv: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/talk/media/testdata/faces.1280x720_P420.yuv -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/modules_unittests.gtest-memcheck.txt: -------------------------------------------------------------------------------- 1 | # Tests that are too slow. 2 | CommonFormats/AudioProcessingTest* 3 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/win_test/WinTest.aps: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/webrtc/voice_engine/test/win_test/WinTest.aps -------------------------------------------------------------------------------- /data/voice_engine/stereo_rtp_files/rtpplay.exe: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/data/voice_engine/stereo_rtp_files/rtpplay.exe 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Licensed under Apache 2.0 3 | This product includes software developed at Tavendo GmbH http://www.tavendo.de 4 | -------------------------------------------------------------------------------- /talk/build/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/OWNERS: -------------------------------------------------------------------------------- 1 | henrike@webrtc.org 2 | perkj@webrtc.org 3 | henrika@webrtc.org 4 | henrikg@webrtc.org 5 | mflodman@webrtc.org 6 | niklas.enbom@webrtc.org 7 | 8 | per-file BUILD.gn=kjellander@webrtc.org 9 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/res/drawable/icon.png: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/svn2github/webrtc/HEAD/webrtc/modules/audio_device/test/android/audio_device_android_test/res/drawable/icon.png -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/common_video_unittests.gtest-drmemory_win32.txt: -------------------------------------------------------------------------------- 1 | # Too slow on Dr Memory Full. 2 | # https://code.google.com/p/webrtc/issues/detail?id=3247 3 | TestScaler.BiLinearScaleTest 4 | TestScaler.BoxScaleTest 5 | -------------------------------------------------------------------------------- /webrtc/modules/rtp_rtcp/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/rtp_rtcp/test/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/utility/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/libjingle_unittest.gtest-memcheck.txt: -------------------------------------------------------------------------------- 1 | # Fails when run under memcheck 2 | # https://code.google.com/p/webrtc/issues/detail?id=1976 3 | PhysicalSocketTest.TestUdpReadyToSendIPv4 4 | ThreadTest.Release 5 | 6 | -------------------------------------------------------------------------------- /tools/whitespace.txt: -------------------------------------------------------------------------------- 1 | You can modify this file to create no-op changelists. 2 | 3 | Try to write something funny. And please don't add trailing whitespace. 4 | 5 | 6 | Whitespaces are cool. 7 | 8 | 9 | 10 | 11 | Really cool. 12 | -------------------------------------------------------------------------------- /webrtc/modules/media_file/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/rtp_rtcp/test/testFec/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/utility/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/libvietest/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/cng/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/g711/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/g722/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/ilbc/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/isac/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/opus/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/pcm16b/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/main/acm2/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/main/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/test/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/tools/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/vp8/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/main/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_conference_mixer/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_processing/main/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /chromium/README: -------------------------------------------------------------------------------- 1 | This .gclient file is used to do download a copy of Chromium. 2 | WebRTC uses the Chromium build toolchain and a number of shared 3 | dependencies by creating symlinks to folders in this checkout, 4 | using the ../setup_links.py script. 5 | 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/isac/fix/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/isac/main/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/i420/main/source/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/res/menu/main_activity_actions.xml: -------------------------------------------------------------------------------- 1 | 2 | 5 | -------------------------------------------------------------------------------- /webrtc/modules/OWNERS: -------------------------------------------------------------------------------- 1 | per-file *.isolate=kjellander@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 *.gyp=* 6 | per-file *.gypi=* 7 | -------------------------------------------------------------------------------- /third_party/gflags/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | 7 | per-file BUILD.gn=kjellander@webrtc.org 8 | -------------------------------------------------------------------------------- /tools/.gitignore: -------------------------------------------------------------------------------- 1 | *.pyc 2 | *~ 3 | .*.sw? 4 | .DS_Store 5 | .code_review_password 6 | .cproject 7 | .metadata 8 | .project 9 | .pydevproject 10 | .settings 11 | .status_password 12 | /third_party/gaeunit 13 | /third_party/google-visualization-python 14 | /third_party/oauth2 15 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/OWNERS: -------------------------------------------------------------------------------- 1 | per-file *.isolate=kjellander@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 *.gyp=* 6 | per-file *.gypi=* 7 | -------------------------------------------------------------------------------- /third_party/winsdk_samples/OWNERS: -------------------------------------------------------------------------------- 1 | 2 | # These are for the common case of adding or renaming files. If you're doing 3 | # structural changes, please get a review from a reviewer in this file. 4 | per-file *.gyp=* 5 | per-file *.gypi=* 6 | 7 | per-file BUILD.gn=kjellander@webrtc.org 8 | 9 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/libjingle_p2p_unittest.gtest-drmemory_win32.txt: -------------------------------------------------------------------------------- 1 | # Fails on Dr Memory Full. 2 | # https://code.google.com/p/webrtc/issues/detail?id=3158 3 | P2PTransportChannel*.* 4 | PortAllocatorTest.* 5 | PortTest.* 6 | PseudoTcpTest.TestSendBothUseLargeWindowScale 7 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/OWNERS: -------------------------------------------------------------------------------- 1 | per-file *.isolate=kjellander@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 *.gyp=* 6 | per-file *.gypi=* 7 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/tools/OWNERS: -------------------------------------------------------------------------------- 1 | per-file *.isolate=kjellander@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 *.gyp=* 6 | per-file *.gypi=* 7 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/res/values/strings.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | WebRTC Audio Device Android Test 5 | Run Test 6 | 7 | -------------------------------------------------------------------------------- /drover.properties: -------------------------------------------------------------------------------- 1 | BASE_URL = "https://webrtc.googlecode.com/svn" 2 | TRUNK_URL = BASE_URL + "/trunk" 3 | BRANCH_URL = BASE_URL + "/branches/$branch" 4 | SKIP_CHECK_WORKING = True 5 | FILE_PATTERN = file_pattern_ = r"[ ]+([MADUC])[ ]+/((?:trunk|branches/.*?)(.*)/(.*))" 6 | PROMPT_FOR_AUTHOR = False 7 | -------------------------------------------------------------------------------- /third_party/gtest-parallel/README.webrtc: -------------------------------------------------------------------------------- 1 | URL: https://github.com/google/gtest-parallel 2 | Version: e61a8975cc124c9a07cb903b76b46b3e669cd179 3 | License: Apache 2.0 4 | License File: LICENSE 5 | 6 | Description: 7 | Parallelization script for gtest binaries. 8 | 9 | Local Modifications: None 10 | -------------------------------------------------------------------------------- /webrtc/build/OWNERS: -------------------------------------------------------------------------------- 1 | kjellander@webrtc.org 2 | per-file tsan_suppressions_webrtc.cc=pbos@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 *.gyp=* 7 | per-file *.gypi=* 8 | -------------------------------------------------------------------------------- /tools/codereview.settings: -------------------------------------------------------------------------------- 1 | # This file is used by gcl to get repository specific information. 2 | CODE_REVIEW_SERVER: webrtc-codereview.appspot.com 3 | CC_LIST: webrtc-reviews@webrtc.org 4 | VIEW_VC: http://code.google.com/p/webrtc/source/detail?r= 5 | TRY_ON_UPLOAD: False 6 | TRYSERVER_ROOT: src/tools 7 | 8 | -------------------------------------------------------------------------------- /webrtc/OWNERS: -------------------------------------------------------------------------------- 1 | per-file *.isolate=kjellander@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 *.gyp=* 6 | per-file *.gypi=* 7 | 8 | per-file BUILD.gn=kjellander@webrtc.org 9 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/libjingle_media_unittest.gtest-memcheck.txt: -------------------------------------------------------------------------------- 1 | #TODO(wu): https://code.google.com/p/webrtc/issues/detail?id=2380 2 | WebRtcVideoMediaChannelTest.TwoStreamsSendAndUnsignalledRecv 3 | 4 | #TODO(jiayl): https://code.google.com/p/webrtc/issues/detail?id=3492 5 | SctpDataMediaChannelTest.* 6 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/rtc_unittests.gtest-drmemory.txt: -------------------------------------------------------------------------------- 1 | # Fails on Dr Memory Full. 2 | # https://code.google.com/p/webrtc/issues/detail?id=3158 3 | P2PTransportChannel*.* 4 | PortAllocatorTest.* 5 | PortTest.* 6 | PseudoTcpTest.TestSendBothUseLargeWindowScale 7 | SharedExclusiveLockTest.TestSharedShared 8 | -------------------------------------------------------------------------------- /webrtc/p2p/OWNERS: -------------------------------------------------------------------------------- 1 | henrika@webrtc.org 2 | henrike@webrtc.org 3 | henrikg@webrtc.org 4 | hta@webrtc.org 5 | jiayl@webrtc.org 6 | juberti@webrtc.org 7 | mflodman@webrtc.org 8 | perkj@webrtc.org 9 | pthatcher@webrtc.org 10 | sergeyu@chromium.org 11 | tommi@webrtc.org 12 | 13 | per-file BUILD.gn=kjellander@webrtc.org 14 | -------------------------------------------------------------------------------- /webrtc/base/OWNERS: -------------------------------------------------------------------------------- 1 | henrika@webrtc.org 2 | henrike@webrtc.org 3 | henrikg@webrtc.org 4 | hta@webrtc.org 5 | jiayl@webrtc.org 6 | juberti@webrtc.org 7 | mflodman@webrtc.org 8 | perkj@webrtc.org 9 | pthatcher@webrtc.org 10 | sergeyu@chromium.org 11 | tommi@webrtc.org 12 | 13 | per-file BUILD.gn=kjellander@webrtc.org 14 | -------------------------------------------------------------------------------- /data/voice_engine/stereo_rtp_files/README.txt: -------------------------------------------------------------------------------- 1 | Use RTP Play tool with command 'rtpplay.exe -v -T -f \ 127.0.0.1/1236' 2 | Example: rtpplay.exe -v -T -f hrtf_g722_1C_48.rtp 127.0.0.1/1236. 3 | This sends the stereo rtp file to port 1236. 4 | You can hear the voice getting panned from left, right and center. 5 | -------------------------------------------------------------------------------- /webrtc/examples/android/opensl_loopback/res/values/strings.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | WebRTCOpenSLLoopback 4 | StartCall 5 | StopCall 6 | Exit 7 | 8 | -------------------------------------------------------------------------------- /webrtc/libjingle/OWNERS: -------------------------------------------------------------------------------- 1 | henrika@webrtc.org 2 | henrike@webrtc.org 3 | henrikg@webrtc.org 4 | hta@webrtc.org 5 | jiayl@webrtc.org 6 | juberti@webrtc.org 7 | mflodman@webrtc.org 8 | perkj@webrtc.org 9 | pthatcher@webrtc.org 10 | sergeyu@chromium.org 11 | tommi@webrtc.org 12 | 13 | per-file BUILD.gn=kjellander@webrtc.org 14 | -------------------------------------------------------------------------------- /webrtc/overrides/OWNERS: -------------------------------------------------------------------------------- 1 | henrika@webrtc.org 2 | henrike@webrtc.org 3 | henrikg@webrtc.org 4 | hta@webrtc.org 5 | jiayl@webrtc.org 6 | juberti@webrtc.org 7 | mflodman@webrtc.org 8 | perkj@webrtc.org 9 | pthatcher@webrtc.org 10 | sergeyu@chromium.org 11 | tommi@webrtc.org 12 | 13 | per-file BUILD.gn=kjellander@webrtc.org 14 | -------------------------------------------------------------------------------- /webrtc/sound/OWNERS: -------------------------------------------------------------------------------- 1 | henrika@webrtc.org 2 | henrike@webrtc.org 3 | henrikg@webrtc.org 4 | hta@webrtc.org 5 | jiayl@webrtc.org 6 | juberti@webrtc.org 7 | mflodman@webrtc.org 8 | perkj@webrtc.org 9 | pthatcher@webrtc.org 10 | sergeyu@chromium.org 11 | tommi@webrtc.org 12 | 13 | 14 | per-file BUILD.gn=kjellander@webrtc.org 15 | -------------------------------------------------------------------------------- /webrtc/tools/OWNERS: -------------------------------------------------------------------------------- 1 | phoglund@webrtc.org 2 | kjellander@webrtc.org 3 | 4 | per-file *.isolate=kjellander@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 *.gyp=* 9 | per-file *.gypi=* 10 | -------------------------------------------------------------------------------- /webrtc/video/OWNERS: -------------------------------------------------------------------------------- 1 | mflodman@webrtc.org 2 | stefan@webrtc.org 3 | pbos@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 *.gyp=* 8 | per-file *.gypi=* 9 | 10 | per-file BUILD.gn=kjellander@webrtc.org 11 | -------------------------------------------------------------------------------- /webrtc/test/OWNERS: -------------------------------------------------------------------------------- 1 | kjellander@webrtc.org 2 | pbos@webrtc.org 3 | phoglund@webrtc.org 4 | 5 | per-file *.isolate=kjellander@webrtc.org 6 | 7 | # These are for the common case of adding or renaming files. If you're doing 8 | # structural changes, please get a review from a reviewer in this file. 9 | per-file *.gyp=* 10 | per-file *.gypi=* 11 | -------------------------------------------------------------------------------- /webrtc/modules/pacing/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | mflodman@webrtc.org 3 | asapersson@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 *.gyp=* 8 | per-file *.gypi=* 9 | 10 | per-file BUILD.gn=kjellander@webrtc.org 11 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/.classpath: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/OWNERS: -------------------------------------------------------------------------------- 1 | aluebs@webrtc.org 2 | andrew@webrtc.org 3 | bjornv@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 *.gyp=* 8 | per-file *.gypi=* 9 | 10 | per-file BUILD.gn=kjellander@webrtc.org 11 | -------------------------------------------------------------------------------- /talk/examples/android/res/values/styles.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | 8 | 9 | -------------------------------------------------------------------------------- /talk/examples/android/res/values-v21/styles.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | 8 | 9 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/.classpath: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | -------------------------------------------------------------------------------- /talk/examples/android/res/menu/connect_menu.xml: -------------------------------------------------------------------------------- 1 | 2 | 8 | 9 | -------------------------------------------------------------------------------- /webrtc/modules/desktop_capture/OWNERS: -------------------------------------------------------------------------------- 1 | alexeypa@chromium.org 2 | jiayl@webrtc.org 3 | sergeyu@chromium.org 4 | wez@chromium.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 *.gyp=* 9 | per-file *.gypi=* 10 | 11 | per-file BUILD.gn=kjellander@webrtc.org 12 | -------------------------------------------------------------------------------- /webrtc/video_engine/OWNERS: -------------------------------------------------------------------------------- 1 | mflodman@webrtc.org 2 | stefan@webrtc.org 3 | 4 | per-file *.isolate=kjellander@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 *.gyp=* 9 | per-file *.gypi=* 10 | 11 | per-file BUILD.gn=kjellander@webrtc.org 12 | 13 | -------------------------------------------------------------------------------- /webrtc/libjingle/examples/call/Info.plist: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | CFBundleIdentifier 6 | com.google.call 7 | CFBundleName 8 | call 9 | 10 | 11 | 12 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt: -------------------------------------------------------------------------------- 1 | # Flakily fails or crashes on Dr Memory Full. 2 | # https://code.google.com/p/webrtc/issues/detail?id=3158 3 | DtmfSenderTest.* 4 | JsepPeerConnectionP2PTestClient.* 5 | PeerConnectionEndToEndTest.* 6 | PeerConnectionInterfaceTest.* 7 | # Issue 3453 8 | WebRtcSessionTest.TestReceiveSdesOfferCreateSdesAnswer 9 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/modules_tests.gtest-memcheck.txt: -------------------------------------------------------------------------------- 1 | # Tests that are too slow. 2 | AudioCodingModuleTest.TestAllCodecs* 3 | AudioCodingModuleTest.TestEncodeDecode* 4 | AudioCodingModuleTest.TestFEC* 5 | AudioCodingModuleTest.TestIsac* 6 | AudioCodingModuleTest.TwoWayCommunication* 7 | AudioCodingModuleTest.TestStereo* 8 | AudioCodingModuleTest.TestVADDTX* 9 | FecTest.FecTest 10 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq/OWNERS: -------------------------------------------------------------------------------- 1 | henrik.lundin@webrtc.org 2 | tina.legrand@webrtc.org 3 | turaj@webrtc.org 4 | minyue@webrtc.org 5 | 6 | per-file *.isolate=kjellander@webrtc.org 7 | 8 | # These are for the common case of adding or renaming files. If you're doing 9 | # structural changes, please get a review from a reviewer in this file. 10 | per-file *.gyp=* 11 | per-file *.gypi=* 12 | -------------------------------------------------------------------------------- /webrtc/modules/bitrate_controller/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | henrik.lundin@webrtc.org 3 | mflodman@webrtc.org 4 | asapersson@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 *.gyp=* 9 | per-file *.gypi=* 10 | 11 | per-file BUILD.gn=kjellander@webrtc.org 12 | -------------------------------------------------------------------------------- /webrtc/modules/remote_bitrate_estimator/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | henrik.lundin@webrtc.org 3 | mflodman@webrtc.org 4 | asapersson@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 *.gyp=* 9 | per-file *.gypi=* 10 | 11 | per-file BUILD.gn=kjellander@webrtc.org 12 | -------------------------------------------------------------------------------- /webrtc/README.chromium: -------------------------------------------------------------------------------- 1 | Name: WebRTC 2 | URL: http://www.webrtc.org 3 | Version: 90 4 | License: BSD 5 | License File: LICENSE 6 | 7 | Description: 8 | WebRTC provides real time voice and video processing 9 | functionality to enable the implementation of 10 | PeerConnection/MediaStream. 11 | 12 | Third party code used in this project is described 13 | in the file LICENSE_THIRD_PARTY. 14 | -------------------------------------------------------------------------------- /webrtc/common_video/OWNERS: -------------------------------------------------------------------------------- 1 | stefan@webrtc.org 2 | marpan@webrtc.org 3 | henrik.lundin@webrtc.org 4 | 5 | per-file *.isolate=kjellander@webrtc.org 6 | 7 | # These are for the common case of adding or renaming files. If you're doing 8 | # structural changes, please get a review from a reviewer in this file. 9 | per-file *.gyp=* 10 | per-file *.gypi=* 11 | 12 | per-file BUILD.gn=kjellander@webrtc.org 13 | -------------------------------------------------------------------------------- /webrtc/modules/video_render/OWNERS: -------------------------------------------------------------------------------- 1 | mflodman@webrtc.org 2 | perkj@webrtc.org 3 | tkchin@webrtc.org 4 | 5 | per-file *.isolate=kjellander@webrtc.org 6 | 7 | # These are for the common case of adding or renaming files. If you're doing 8 | # structural changes, please get a review from a reviewer in this file. 9 | per-file *.gyp=* 10 | per-file *.gypi=* 11 | 12 | per-file BUILD.gn=kjellander@webrtc.org 13 | -------------------------------------------------------------------------------- /tools/DEPS: -------------------------------------------------------------------------------- 1 | # Tools has its own dependencies, separate from the production code. 2 | # Use http rather than https; the latter can cause problems for users behind 3 | # proxies. 4 | 5 | deps = { 6 | # Used by tools/quality_tracking/dashboard and tools/python_charts. 7 | "tools/third_party/google-visualization-python": 8 | "http://google-visualization-python.googlecode.com/svn/trunk/@15", 9 | } 10 | 11 | -------------------------------------------------------------------------------- /webrtc/tools/e2e_quality/audio/default.pa: -------------------------------------------------------------------------------- 1 | # Place in ~/.pulse/ to add null sinks for the audio end-to-end quality test. 2 | 3 | .include /etc/pulse/default.pa 4 | 5 | load-module module-null-sink sink_name=render sink_properties=device.description=render format=s16 rate=48000 channels=1 6 | load-module module-null-sink sink_name=capture sink_properties=device.description=capture format=s16 rate=48000 channels=1 7 | -------------------------------------------------------------------------------- /OWNERS: -------------------------------------------------------------------------------- 1 | andrew@webrtc.org 2 | henrika@webrtc.org 3 | mflodman@webrtc.org 4 | niklas.enbom@webrtc.org 5 | tina.legrand@webrtc.org 6 | tommi@webrtc.org 7 | per-file .gitignore=* 8 | per-file AUTHORS=* 9 | per-file BUILD.gn=kjellander@webrtc.org 10 | per-file DEPS=* 11 | per-file PRESUBMIT.py=kjellander@webrtc.org 12 | per-file setup_links.py=* 13 | per-file sync_chromium.py=kjellander@webrtc.org 14 | per-file WATCHLISTS=* 15 | -------------------------------------------------------------------------------- /talk/codereview.settings: -------------------------------------------------------------------------------- 1 | # This file is used by gcl to get repository specific information. 2 | CODE_REVIEW_SERVER: webrtc-codereview.appspot.com 3 | CC_LIST: webrtc-reviews@webrtc.org 4 | VIEW_VC: http://code.google.com/p/webrtc/source/detail?r= 5 | TRY_ON_UPLOAD: False 6 | TRYSERVER_SVN_URL: svn://svn.chromium.org/chrome-try/try-webrtc 7 | TRYSERVER_ROOT: src/talk 8 | PROJECT: webrtc 9 | FORCE_HTTPS_COMMIT_URL: True 10 | -------------------------------------------------------------------------------- /webrtc/codereview.settings: -------------------------------------------------------------------------------- 1 | # This file is used by gcl to get repository specific information. 2 | CODE_REVIEW_SERVER: webrtc-codereview.appspot.com 3 | CC_LIST: webrtc-reviews@webrtc.org 4 | VIEW_VC: http://code.google.com/p/webrtc/source/detail?r= 5 | TRY_ON_UPLOAD: False 6 | TRYSERVER_SVN_URL: svn://svn.chromium.org/chrome-try/try-webrtc 7 | TRYSERVER_ROOT: src/webrtc 8 | PROJECT: webrtc 9 | FORCE_HTTPS_COMMIT_URL: True 10 | -------------------------------------------------------------------------------- /webrtc/modules/video_capture/OWNERS: -------------------------------------------------------------------------------- 1 | glaznev@webrtc.org 2 | mflodman@webrtc.org 3 | perkj@webrtc.org 4 | tkchin@webrtc.org 5 | 6 | per-file *.isolate=kjellander@webrtc.org 7 | 8 | # These are for the common case of adding or renaming files. If you're doing 9 | # structural changes, please get a review from a reviewer in this file. 10 | per-file *.gyp=* 11 | per-file *.gypi=* 12 | 13 | per-file BUILD.gn=kjellander@webrtc.org 14 | -------------------------------------------------------------------------------- /webrtc/voice_engine/OWNERS: -------------------------------------------------------------------------------- 1 | henrikg@webrtc.org 2 | henrika@webrtc.org 3 | niklas.enbom@webrtc.org 4 | xians@webrtc.org 5 | 6 | per-file *.isolate=kjellander@webrtc.org 7 | 8 | # These are for the common case of adding or renaming files. If you're doing 9 | # structural changes, please get a review from a reviewer in this file. 10 | per-file *.gyp=* 11 | per-file *.gypi=* 12 | 13 | per-file BUILD.gn=kjellander@webrtc.org 14 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/libjingle_unittest.gtest-memcheck_mac.txt: -------------------------------------------------------------------------------- 1 | # Crashes the kernel when run under memcheck on Mac. 2 | # No bug is filed in relation to this since it's unlikely we can fix it. 3 | # There are several tests disabled in the libjingle code with similar comments, 4 | # without bugs assigned to them. 5 | # Example: talk/base/physicalsocketserver_unittest.cc 6 | NatTest.TestVirtualIPv* 7 | PosixSignalDeliveryTest.* 8 | -------------------------------------------------------------------------------- /webrtc/common_audio/OWNERS: -------------------------------------------------------------------------------- 1 | bjornv@webrtc.org 2 | tina.legrand@webrtc.org 3 | jan.skoglund@webrtc.org 4 | andrew@webrtc.org 5 | 6 | per-file *.isolate=kjellander@webrtc.org 7 | 8 | # These are for the common case of adding or renaming files. If you're doing 9 | # structural changes, please get a review from a reviewer in this file. 10 | per-file *.gyp=* 11 | per-file *.gypi=* 12 | 13 | per-file BUILD.gn=kjellander@webrtc.org 14 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/libjingle_p2p_unittest.gtest-memcheck.txt: -------------------------------------------------------------------------------- 1 | # TODO(wu): Reenable P2PTransportChannelTest after issue 2158 is resolved. 2 | P2PTransportChannelTest.* 3 | P2PTransportChannelSameNatTest.TestConesBehindSameCone 4 | PortTest.TestSendStunMessageAsIce 5 | PseudoTcpTest.TestSendBothUseLargeWindowScale 6 | # Issue 3447 7 | P2PTransportChannelMultihomedTest.TestFailover 8 | P2PTransportChannelMultihomedTest.TestDrain 9 | -------------------------------------------------------------------------------- /codereview.settings: -------------------------------------------------------------------------------- 1 | # This file is used by gcl to get repository specific information. 2 | CODE_REVIEW_SERVER: webrtc-codereview.appspot.com 3 | CC_LIST: webrtc-reviews@webrtc.org 4 | VIEW_VC: http://code.google.com/p/webrtc/source/detail?r= 5 | TRY_ON_UPLOAD: False 6 | TRYSERVER_SVN_URL: svn://svn.chromium.org/chrome-try/try-webrtc 7 | TRYSERVER_PROJECT: webrtc 8 | TRYSERVER_ROOT: src 9 | PROJECT: webrtc 10 | FORCE_HTTPS_COMMIT_URL: True 11 | -------------------------------------------------------------------------------- /tools/cpu/README: -------------------------------------------------------------------------------- 1 | This directory contains a little utility for doing CPU measurements. 2 | It requires a Python package, psutil, to be installed. 3 | See: https://pypi.python.org/pypi/psutil 4 | 5 | On mac you can install this package like so: 6 | sudo ARCHFLAGS='-Wno-error=unused-command-line-argument-hard-error-in-future' easy_install psutil 7 | 8 | On Windows: 9 | - TBD (see link above) 10 | 11 | On Linux: 12 | - TBD (see link above) 13 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/OWNERS: -------------------------------------------------------------------------------- 1 | henrikg@webrtc.org 2 | henrika@webrtc.org 3 | niklas.enbom@webrtc.org 4 | tkchin@webrtc.org 5 | xians@webrtc.org 6 | 7 | per-file *.isolate=kjellander@webrtc.org 8 | 9 | # These are for the common case of adding or renaming files. If you're doing 10 | # structural changes, please get a review from a reviewer in this file. 11 | per-file *.gyp=* 12 | per-file *.gypi=* 13 | 14 | per-file BUILD.gn=kjellander@webrtc.org 15 | -------------------------------------------------------------------------------- /talk/OWNERS: -------------------------------------------------------------------------------- 1 | set noparent 2 | henrike@webrtc.org 3 | hta@webrtc.org 4 | jiayl@webrtc.org 5 | juberti@webrtc.org 6 | perkj@webrtc.org 7 | pthatcher@webrtc.org 8 | sergeyu@chromium.org 9 | tommi@webrtc.org 10 | per-file *.isolate=kjellander@webrtc.org 11 | 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 *.gyp=* 16 | per-file *.gypi=* 17 | -------------------------------------------------------------------------------- /license_template.txt: -------------------------------------------------------------------------------- 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 | -------------------------------------------------------------------------------- /tools/python_charts/webrtc/__init__.py: -------------------------------------------------------------------------------- 1 | #!/usr/bin/env python 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. -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /webrtc/tools/loopback_test/README: -------------------------------------------------------------------------------- 1 | Loopback test 2 | 3 | This is a simple html test framework to run a loopback test which can go via 4 | turn. For now the test is used to analyse bandwidth estimation and get records 5 | for bad scenarios. 6 | 7 | How to run: 8 | ./run-server.sh (to start python serving the tests) 9 | Access http://localhost:8080/loopback_test.html to run the test 10 | 11 | How to record: 12 | You can use record-test.sh to get a tcpdump of a test run. 13 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/android/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 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/win_test/res/WinTest.rc2: -------------------------------------------------------------------------------- 1 | // 2 | // WinTest.RC2 - resources Microsoft Visual C++ does not edit directly 3 | // 4 | 5 | #ifdef APSTUDIO_INVOKED 6 | #error this file is not editable by Microsoft Visual C++ 7 | #endif //APSTUDIO_INVOKED 8 | 9 | 10 | ///////////////////////////////////////////////////////////////////////////// 11 | // Add manually edited resources here... 12 | 13 | ///////////////////////////////////////////////////////////////////////////// 14 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/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-3 12 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/android/.classpath: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 | -------------------------------------------------------------------------------- /tools/msan/blacklist.txt: -------------------------------------------------------------------------------- 1 | # The rules in this file are only applied at compile time. 2 | # Because the Chrome buildsystem does not automatically touch the files 3 | # mentioned here, changing this file requires clobbering all MSan bots. 4 | # 5 | # Please think twice before you add or remove these rules. 6 | 7 | # This is a stripped down copy of Chromium's blacklist.txt, to enable 8 | # adding WebRTC-specific blacklist entries. 9 | 10 | # Uninit in zlib. http://crbug.com/116277 11 | fun:*MOZ_Z_deflate* 12 | 13 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/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, OpenSL ES requires API level 9 11 | target=android-9 12 | -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/transient/test/readPCM.m: -------------------------------------------------------------------------------- 1 | function [x, t] = readPCM(file, fs) 2 | %[x, t] = readPCM(file, fs) 3 | % 4 | %Reads a signal from a PCM file. 5 | % 6 | %x: The read signal after normalization. 7 | %t: The respective time vector. 8 | % 9 | %file: The PCM file where the signal is stored in int16 format. 10 | %fs: The signal sample rate in Hertz. 11 | fid = fopen(file); 12 | x = fread(fid, inf, 'int16'); 13 | fclose(fid); 14 | x = x - mean(x); 15 | x = x / max(abs(x)); 16 | t = 0:(1 / fs):((length(x) - 1) / fs); 17 | -------------------------------------------------------------------------------- /talk/examples/androidtests/README: -------------------------------------------------------------------------------- 1 | This directory contains an example unit test for Android AppRTCDemo. 2 | 3 | Example of building & using the app: 4 | 5 | - Build Android AppRTCDemo and AppRTCDemo unit test: 6 | cd /src 7 | ninja -C out/Debug AppRTCDemoTest 8 | 9 | - Install AppRTCDemo and AppRTCDemoTest: 10 | adb install -r out/Debug/AppRTCDemo-debug.apk 11 | adb install -r out/Debug/AppRTCDemoTest-debug.apk 12 | 13 | - Run unit tests: 14 | adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/res/layout/main.xml: -------------------------------------------------------------------------------- 1 | 2 | 6 | 7 | 12 | 13 | -------------------------------------------------------------------------------- /webrtc/tools/rtcbot/bot/browser/index.html: -------------------------------------------------------------------------------- 1 | 10 | 11 | 12 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/modules_tests.gtest-drmemory.txt: -------------------------------------------------------------------------------- 1 | # Tests that are too slow. 2 | AudioCodingModuleTest.TestAllCodecs* 3 | AudioCodingModuleTest.TestEncodeDecode* 4 | AudioCodingModuleTest.TestFEC* 5 | AudioCodingModuleTest.TestIsac* 6 | AudioCodingModuleTest.TwoWayCommunication* 7 | AudioCodingModuleTest.TestStereo* 8 | AudioCodingModuleTest.TestVADDTX* 9 | AudioCodingModuleTest.TestOpus* 10 | FecTest.FecTest 11 | TestVp8Impl.BaseUnitTest 12 | VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP8 13 | VideoProcessorIntegrationTest.*VP9 14 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/main/source/audio_device.gypi: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2012 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 | 'includes': [ 11 | '../../audio_device.gypi', 12 | ], 13 | } 14 | 15 | -------------------------------------------------------------------------------- /BUILD.gn: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2014 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 | # This file is copied and modified from Chromium (src/BUILD.gn). 10 | group("root") { 11 | deps = [ 12 | "//webrtc", 13 | ] 14 | } 15 | -------------------------------------------------------------------------------- /webrtc/build/no_op.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 | // No-op main() to provide a dummy executable target. 12 | int main() { 13 | return 0; 14 | } 15 | -------------------------------------------------------------------------------- /webrtc/tools/barcode_tools/DEPS: -------------------------------------------------------------------------------- 1 | # This is trimmed down version of the main tools DEPS file which is to be used 2 | # in Chromiums PyAuto WebRTC video quality measurement test. We will only 3 | # need the Zxing dependencies as we only use the barcode tools in this test. 4 | 5 | deps = { 6 | # Used by barcode_tools 7 | "barcode_tools/third_party/zxing/core": 8 | "http://zxing.googlecode.com/svn/trunk/core@2349", 9 | 10 | # Used by barcode_tools 11 | "barcode_tools/third_party/zxing/javase": 12 | "http://zxing.googlecode.com/svn/trunk/javase@2349", 13 | } 14 | -------------------------------------------------------------------------------- /webrtc/modules/video_processing/main/test/unit_test/writeYUV420file.m: -------------------------------------------------------------------------------- 1 | function writeYUV420file(filename, Y, U, V) 2 | % writeYUV420file(filename, Y, U, V) 3 | 4 | fid = fopen(filename,'wb'); 5 | if fid==-1 6 | error(['Cannot open file ' filename]); 7 | end 8 | 9 | numFrames=size(Y,3); 10 | 11 | for k=1:numFrames 12 | % Write luminance 13 | fwrite(fid,uint8(Y(:,:,k).'), 'uchar'); 14 | 15 | % Write U channel 16 | fwrite(fid,uint8(U(:,:,k).'), 'uchar'); 17 | 18 | % Write V channel 19 | fwrite(fid,uint8(V(:,:,k).'), 'uchar'); 20 | end 21 | 22 | fclose(fid); 23 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/rtc_unittests.gtest-memcheck.txt: -------------------------------------------------------------------------------- 1 | # Fails when run under memcheck 2 | # https://code.google.com/p/webrtc/issues/detail?id=1976 3 | PhysicalSocketTest.TestUdpReadyToSendIPv4 4 | ThreadTest.Release 5 | # TODO(wu): Reenable P2PTransportChannelTest after issue 2158 is resolved. 6 | P2PTransportChannelTest.* 7 | P2PTransportChannelSameNatTest.TestConesBehindSameCone 8 | PortTest.TestSendStunMessageAsIce 9 | PseudoTcpTest.TestSendBothUseLargeWindowScale 10 | # Issue 3447 11 | P2PTransportChannelMultihomedTest.TestFailover 12 | P2PTransportChannelMultihomedTest.TestDrain 13 | 14 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/jni/Application.mk: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2012 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 | # Build both ARMv5TE and ARMv7-A machine code. 10 | APP_ABI := armeabi armeabi-v7a x86 11 | APP_STL := stlport_shared 12 | -------------------------------------------------------------------------------- /third_party/winsdk_samples/README.webrtc: -------------------------------------------------------------------------------- 1 | Name: winsdk_samples 2 | URL: http://www.microsoft.com/en-us/download/details.aspx?id=8279 3 | Version: 7.1 4 | License: Microsoft Windows SDK license 5 | License File: src/License/License.htm 6 | Security Critical: yes 7 | 8 | Description: 9 | This contains a copy of a portion of the Microsoft Windows SDK 7.1 sample 10 | code. It is covered by the "Sample Code" section of the license. 11 | 12 | This would typically be installed to: 13 | C:\Program Files\Microsoft SDKs\Windows\v7.1 14 | 15 | It is used by WebRTC to capture video from a camera on Windows. 16 | -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/transient/test/plotDetection.m: -------------------------------------------------------------------------------- 1 | function [] = plotDetection(PCMfile, DATfile, fs, chunkSize) 2 | %[] = plotDetection(PCMfile, DATfile, fs, chunkSize) 3 | % 4 | %Plots the signal alongside the detection values. 5 | % 6 | %PCMfile: The file of the input signal in PCM format. 7 | %DATfile: The file containing the detection values in binary float format. 8 | %fs: The sample rate of the signal in Hertz. 9 | %chunkSize: The chunk size used to compute the detection values in seconds. 10 | [x, tx] = readPCM(PCMfile, fs); 11 | [d, td] = readDetection(DATfile, fs, chunkSize); 12 | plot(tx, x, td, d); 13 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/main/test/release_test.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 RELEASE_TEST_H 12 | #define RELEASE_TEST_H 13 | 14 | int ReleaseTest(); 15 | int ReleaseTestPart2(); 16 | 17 | #endif -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /talk/examples/androidtests/AndroidManifest.xml: -------------------------------------------------------------------------------- 1 | 2 | 6 | 7 | 8 | 9 | 12 | 13 | 14 | 15 | 16 | 17 | -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/res/values/integers.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | 11113 4 | 11113 5 | 0 6 | 1 7 | 2 8 | 0 9 | 0 10 | 0 11 | 11111 12 | 11111 13 | -------------------------------------------------------------------------------- /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/modules/audio_processing/transient/test/readDetection.m: -------------------------------------------------------------------------------- 1 | function [d, t] = readDetection(file, fs, chunkSize) 2 | %[d, t] = readDetection(file, fs, chunkSize) 3 | % 4 | %Reads a detection signal from a DAT file. 5 | % 6 | %d: The detection signal. 7 | %t: The respective time vector. 8 | % 9 | %file: The DAT file where the detection signal is stored in float format. 10 | %fs: The signal sample rate in Hertz. 11 | %chunkSize: The chunk size used for the detection in seconds. 12 | fid = fopen(file); 13 | d = fread(fid, inf, 'float'); 14 | fclose(fid); 15 | t = 0:(1 / fs):(length(d) * chunkSize - 1 / fs); 16 | d = d(floor(t / chunkSize) + 1); 17 | -------------------------------------------------------------------------------- /webrtc/base/asyncfile.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2010 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 "webrtc/base/asyncfile.h" 12 | 13 | namespace rtc { 14 | 15 | AsyncFile::AsyncFile() { 16 | } 17 | 18 | AsyncFile::~AsyncFile() { 19 | } 20 | 21 | } // namespace rtc 22 | -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/MediaEngineObserver.java: -------------------------------------------------------------------------------- 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 | package org.webrtc.webrtcdemo; 12 | 13 | public interface MediaEngineObserver { 14 | void newStats(String stats); 15 | } -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/MenuStateProvider.java: -------------------------------------------------------------------------------- 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 | package org.webrtc.webrtcdemo; 12 | 13 | public interface MenuStateProvider { 14 | public MediaEngine getEngine(); 15 | } -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/main/acm2/acm_neteq_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 | // This file contains unit tests for ACM's NetEQ wrapper (class ACMNetEQ). 12 | 13 | namespace webrtc { 14 | 15 | namespace acm2 {} // namespace 16 | -------------------------------------------------------------------------------- /webrtc/test/run_all_unittests.cc: -------------------------------------------------------------------------------- 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 | #include "webrtc/test/test_suite.h" 12 | 13 | int main(int argc, char** argv) { 14 | webrtc::test::TestSuite test_suite(argc, argv); 15 | return test_suite.Run(); 16 | } 17 | -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/project.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 | # "ant.properties", and override values to adapt the script to your 8 | # project structure. 9 | 10 | # To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home): 11 | #proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt 12 | 13 | # Project target. 14 | target=android-21 15 | -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/res/values/bools.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | false 4 | true 5 | true 6 | true 7 | true 8 | false 9 | true 10 | true 11 | true 12 | true 13 | -------------------------------------------------------------------------------- /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 -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/modules_unittests.gtest-drmemory_win32.txt: -------------------------------------------------------------------------------- 1 | # Too slow to run with Dr Memory on Windows. 2 | ApmTest.EchoCancellationReportsCorrectDelays 3 | ApmTest.FloatAndIntInterfacesGiveSimilarResults 4 | ApmTest.IdenticalInputChannelsResultInIdenticalOutputChannels 5 | ApmTest.VerifyDebugDumpFloat 6 | ApmTest.VerifyDebugDumpInt 7 | CommonFormats/AudioProcessingTest* 8 | TestScaler.PointScaleTest 9 | TestScaler.BiLinearScaleTest 10 | TestScaler.BoxScaleTest 11 | TestVideoSenderWithVp8.* 12 | VideoProcessingModuleTest.Denoising 13 | VideoSendersTest.* 14 | 15 | # https://code.google.com/p/webrtc/issues/detail?id=2323 16 | MouseCursorShapeTest.MatchCursors 17 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/memcheck/suppressions_mac.txt: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2012 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 | # This file is used in addition to the one already maintained in Chrome. 10 | # It acts as a place holder for future additions for WebRTC. 11 | # It must exist for the Python wrapper script to work properly. 12 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/memcheck/suppressions_win32.txt: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2012 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 | # This file is used in addition to the one already maintained in Chrome. 10 | # It acts as a place holder for future additions for WebRTC. 11 | # It must exist for the Python wrapper script to work properly. 12 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-drmemory_win32.txt: -------------------------------------------------------------------------------- 1 | # Never completes on Dr Memory Full. 2 | # https://code.google.com/p/webrtc/issues/detail?id=3159 3 | EndToEndTest.CanSwitchToUseAllSsrcs 4 | EndToEndTest.SendsAndReceivesMultipleStreams 5 | EndToEndTest.ReceivesAndRetransmitsNack 6 | # https://code.google.com/p/webrtc/issues/detail?id=3471 7 | VideoSendStreamTest.RetransmitsNackOverRtxWithPacing 8 | # Flaky: https://code.google.com/p/webrtc/issues/detail?id=3552 9 | EndToEndTest.RestartingSendStreamPreservesRtpState 10 | EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx 11 | EndToEndTest.SendsAndReceivesH264 12 | EndToEndTest.SendsAndReceivesVP9 13 | -------------------------------------------------------------------------------- /webrtc/base/mathutils.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2005 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 WEBRTC_BASE_MATHUTILS_H_ 12 | #define WEBRTC_BASE_MATHUTILS_H_ 13 | 14 | #include 15 | 16 | #ifndef M_PI 17 | #define M_PI 3.14159265359f 18 | #endif 19 | 20 | #endif // WEBRTC_BASE_MATHUTILS_H_ 21 | -------------------------------------------------------------------------------- /webrtc/tools/rtcbot/rtcBotReportVisualizer/index.html: -------------------------------------------------------------------------------- 1 | 10 | 11 | 12 | 13 | 14 | -------------------------------------------------------------------------------- /webrtc/voice_engine/channel_unittest.cc: -------------------------------------------------------------------------------- 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 | #include "testing/gtest/include/gtest/gtest.h" 12 | #include "webrtc/voice_engine/channel.h" 13 | 14 | // Empty test just to get coverage metrics. 15 | TEST(ChannelTest, EmptyTestToGetCodeCoverage) {} 16 | -------------------------------------------------------------------------------- /talk/examples/android/project.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 edit 7 | # "ant.properties", and override values to adapt the script to your 8 | # project structure. 9 | # 10 | # To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home): 11 | #proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt 12 | 13 | # Project target. 14 | target=android-21 15 | 16 | java.compilerargs=-Xlint:all -Werror 17 | -------------------------------------------------------------------------------- /talk/examples/objc/AppRTCDemo/third_party/SocketRocket/LICENSE: -------------------------------------------------------------------------------- 1 | 2 | Copyright 2012 Square Inc. 3 | 4 | Licensed under the Apache License, Version 2.0 (the "License"); 5 | you may not use this file except in compliance with the License. 6 | You may obtain a copy of the License at 7 | 8 | http://www.apache.org/licenses/LICENSE-2.0 9 | 10 | Unless required by applicable law or agreed to in writing, software 11 | distributed under the License is distributed on an "AS IS" BASIS, 12 | WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 | See the License for the specific language governing permissions and 14 | limitations under the License. 15 | 16 | -------------------------------------------------------------------------------- /webrtc/p2p/base/udpport.h: -------------------------------------------------------------------------------- 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 | #ifndef WEBRTC_P2P_BASE_UDPPORT_H_ 12 | #define WEBRTC_P2P_BASE_UDPPORT_H_ 13 | 14 | // StunPort will be handling UDPPort functionality. 15 | #include "webrtc/p2p/base/stunport.h" 16 | 17 | #endif // WEBRTC_P2P_BASE_UDPPORT_H_ 18 | -------------------------------------------------------------------------------- /talk/examples/androidtests/project.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 edit 7 | # "ant.properties", and override values to adapt the script to your 8 | # project structure. 9 | # 10 | # To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home): 11 | #proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt 12 | 13 | # Project target. 14 | target=android-21 15 | 16 | java.compilerargs=-Xlint:all -Werror 17 | -------------------------------------------------------------------------------- /webrtc/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 WEBRTC_BASE_WIN32SOCKETINIT_H_ 12 | #define WEBRTC_BASE_WIN32SOCKETINIT_H_ 13 | 14 | namespace rtc { 15 | 16 | void EnsureWinsockInit(); 17 | 18 | } // namespace rtc 19 | 20 | #endif // WEBRTC_BASE_WIN32SOCKETINIT_H_ 21 | -------------------------------------------------------------------------------- /webrtc/common.gyp: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2014 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 | 'includes': ['build/common.gypi'], 10 | 'targets': [ 11 | { 12 | 'target_name': 'webrtc_common', 13 | 'type': 'static_library', 14 | 'sources': [ 15 | 'config.h', 16 | 'config.cc', 17 | ], 18 | }, 19 | ], 20 | } 21 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/dummy/audio_device_utility_dummy.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 "webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h" 11 | 12 | namespace webrtc { 13 | int32_t AudioDeviceUtilityDummy::Init() { return 0; } 14 | } // namespace webrtc 15 | 16 | -------------------------------------------------------------------------------- /webrtc/modules/video_render/test/testAPI/testAPI_android.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 | int main(int argc, char* argv[]) { 12 | // TODO(leozwang): Video render test app is not ready on android, 13 | // make it dummy test now, will add android specific tests 14 | return 0; 15 | } 16 | -------------------------------------------------------------------------------- /webrtc/p2p/base/sessionid.h: -------------------------------------------------------------------------------- 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 | #ifndef WEBRTC_P2P_BASE_SESSIONID_H_ 12 | #define WEBRTC_P2P_BASE_SESSIONID_H_ 13 | 14 | // TODO: Remove this file. 15 | 16 | namespace cricket { 17 | 18 | } // namespace cricket 19 | 20 | #endif // WEBRTC_P2P_BASE_SESSIONID_H_ 21 | -------------------------------------------------------------------------------- /talk/app/webrtc/androidtests/project.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 edit 7 | # "ant.properties", and override values to adapt the script to your 8 | # project structure. 9 | # 10 | # To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home): 11 | #proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt 12 | 13 | # Project target. 14 | target=android-21 15 | 16 | java.compilerargs=-Xlint:all -Werror 17 | -------------------------------------------------------------------------------- /webrtc/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 "webrtc/base/messagehandler.h" 12 | #include "webrtc/base/messagequeue.h" 13 | 14 | namespace rtc { 15 | 16 | MessageHandler::~MessageHandler() { 17 | MessageQueueManager::Clear(this); 18 | } 19 | 20 | } // namespace rtc 21 | -------------------------------------------------------------------------------- /webrtc/tools/loopback_test/run-server.sh: -------------------------------------------------------------------------------- 1 | #!/bin/sh 2 | # 3 | # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 4 | # 5 | # Use of this source code is governed by a BSD-style license 6 | # that can be found in the LICENSE file in the root of the source 7 | # tree. An additional intellectual property rights grant can be found 8 | # in the file PATENTS. All contributing project authors may 9 | # be found in the AUTHORS file in the root of the source tree. 10 | # 11 | # This script is used to launch a simple http server for files in the same 12 | # location as the script itself. 13 | cd "`dirname \"$0\"`" 14 | echo "Starting http server in port 8080." 15 | exec python -m SimpleHTTPServer 8080 16 | -------------------------------------------------------------------------------- /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 "webrtc/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/base/stringdigest.h: -------------------------------------------------------------------------------- 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 | #ifndef WEBRTC_BASE_STRINGDIGEST_H_ 12 | #define WEBRTC_BASE_STRINGDIGEST_H_ 13 | 14 | // TODO: Update remaining callers to use messagedigest.h instead 15 | #include "webrtc/base/messagedigest.h" 16 | 17 | #endif // WEBRTC_BASE_STRINGDIGEST_H_ 18 | -------------------------------------------------------------------------------- /webrtc/examples/android/opensl_loopback/project.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 edit 7 | # "ant.properties", and override values to adapt the script to your 8 | # project structure. 9 | # 10 | # To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home): 11 | #proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt 12 | 13 | # Project target. 14 | target=android-21 15 | 16 | java.compilerargs=-Xlint:all -Werror 17 | -------------------------------------------------------------------------------- /chromium/.gclient: -------------------------------------------------------------------------------- 1 | solutions = [{ 2 | 'name': 'src', 3 | 'url': 'https://chromium.googlesource.com/chromium/src.git', 4 | 'deps_file': '.DEPS.git', 5 | 'managed': False, 6 | 'custom_deps': { 7 | # Skip syncing some large dependencies WebRTC will never need. 8 | 'src/chrome/tools/test/reference_build/chrome_linux': None, 9 | 'src/chrome/tools/test/reference_build/chrome_mac': None, 10 | 'src/chrome/tools/test/reference_build/chrome_win': None, 11 | 'src/native_client': None, 12 | 'src/third_party/ffmpeg': None, 13 | 'src/third_party/junit/src': None, 14 | 'src/third_party/WebKit': None, 15 | 'src/v8': None, 16 | }, 17 | 'safesync_url': '' 18 | }] 19 | 20 | cache_dir = None 21 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.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 "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" 12 | 13 | AfterStreamingFixture::AfterStreamingFixture() 14 | : BeforeStreamingFixture() { 15 | ResumePlaying(); 16 | } 17 | -------------------------------------------------------------------------------- /webrtc/base/basicdefs.h: -------------------------------------------------------------------------------- 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 | #ifndef WEBRTC_BASE_BASICDEFS_H_ 12 | #define WEBRTC_BASE_BASICDEFS_H_ 13 | 14 | #if HAVE_CONFIG_H 15 | #include "config.h" // NOLINT 16 | #endif 17 | 18 | #define ARRAY_SIZE(x) (static_cast(sizeof(x) / sizeof(x[0]))) 19 | 20 | #endif // WEBRTC_BASE_BASICDEFS_H_ 21 | -------------------------------------------------------------------------------- /webrtc/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 "webrtc/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 | } // namespace rtc 21 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/win_test/stdafx.cc: -------------------------------------------------------------------------------- 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 | // stdafx.cpp : source file that includes just the standard includes 12 | // WinTest.pch will be the pre-compiled header 13 | // stdafx.obj will contain the pre-compiled type information 14 | 15 | #include "webrtc/voice_engine/test/win_test/stdafx.h" 16 | -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/utility/fft4g.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 WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_ 12 | #define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_ 13 | 14 | void WebRtc_rdft(int, int, float *, int *, float *); 15 | void WebRtc_cdft(int, int, float *, int *, float *); 16 | 17 | #endif 18 | -------------------------------------------------------------------------------- /webrtc/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 WEBRTC_BASE_OPENSSL_H_ 12 | #define WEBRTC_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 // WEBRTC_BASE_OPENSSL_H_ 21 | -------------------------------------------------------------------------------- /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 "webrtc/modules/desktop_capture/screen_capturer.h" 12 | 13 | namespace webrtc { 14 | 15 | // static 16 | ScreenCapturer* ScreenCapturer::Create(const DesktopCaptureOptions& options) { 17 | return NULL; 18 | } 19 | 20 | } // namespace webrtc 21 | -------------------------------------------------------------------------------- /webrtc/supplement.gypi: -------------------------------------------------------------------------------- 1 | { 2 | 'variables': { 3 | 'variables': { 4 | 'webrtc_root%': '<(DEPTH)/webrtc', 5 | }, 6 | 'webrtc_root%': '<(webrtc_root)', 7 | 'build_with_chromium': 0, 8 | }, 9 | 'target_defaults': { 10 | 'target_conditions': [ 11 | ['_target_name=="sanitizer_options"', { 12 | 'conditions': [ 13 | ['tsan==1', { 14 | # Replace Chromium's TSan v2 suppressions with our own for WebRTC. 15 | 'sources/': [ 16 | ['exclude', 'tsan_suppressions.cc'], 17 | ], 18 | 'sources': [ 19 | '<(webrtc_root)/build/tsan_suppressions_webrtc.cc', 20 | ], 21 | }], 22 | ], 23 | }], 24 | ], 25 | }, 26 | } 27 | -------------------------------------------------------------------------------- /webrtc/common_video/interface/video_image.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 COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H 12 | #define COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H 13 | 14 | // TODO(pbos): Remove this file and include webrtc/video_frame.h instead. 15 | #include "webrtc/video_frame.h" 16 | 17 | #endif // COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H 18 | -------------------------------------------------------------------------------- /webrtc/modules/video_capture/ensure_initialized.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 | namespace webrtc { 12 | namespace videocapturemodule { 13 | 14 | // Ensure any necessary initialization of webrtc::videocapturemodule has 15 | // completed. 16 | void EnsureInitialized(); 17 | 18 | } // namespace videocapturemodule. 19 | } // namespace webrtc. 20 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/audio_encoder.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 "webrtc/modules/audio_coding/codecs/audio_encoder.h" 12 | 13 | namespace webrtc { 14 | 15 | AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() { 16 | } 17 | 18 | AudioEncoder::EncodedInfo::~EncodedInfo() { 19 | } 20 | 21 | } // namespace webrtc 22 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/res/values/strings.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 | 11 | 12 | WebRtc VoE 13 | 14 | -------------------------------------------------------------------------------- /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 "testing/gtest/include/gtest/gtest.h" 12 | #include "webrtc/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/modules/audio_device/main/interface/audio_device.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_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_ 12 | #define MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_ 13 | 14 | #include "webrtc/modules/audio_device/include/audio_device.h" 15 | 16 | #endif // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_ 17 | -------------------------------------------------------------------------------- /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 "webrtc/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/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/common_video/interface/i420_video_frame.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 COMMON_VIDEO_INTERFACE_I420_VIDEO_FRAME_H 12 | #define COMMON_VIDEO_INTERFACE_I420_VIDEO_FRAME_H 13 | 14 | // TODO(pbos): Remove this file and include webrtc/video_frame.h instead. 15 | #include "webrtc/video_frame.h" 16 | 17 | #endif // COMMON_VIDEO_INTERFACE_I420_VIDEO_FRAME_H 18 | -------------------------------------------------------------------------------- /webrtc/video_engine/video_engine.gyp: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2011 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 | 'includes': [ 11 | '../build/common.gypi', 12 | './video_engine_core.gypi', 13 | ], 14 | 15 | 'conditions': [ 16 | ['include_tests==1', { 17 | 'includes': [ 18 | 'test/libvietest/libvietest.gypi', 19 | 'test/auto_test/vie_auto_test.gypi', 20 | ], 21 | }], 22 | ], 23 | } 24 | 25 | -------------------------------------------------------------------------------- /resources/audio_coding/READ.ME: -------------------------------------------------------------------------------- 1 | Test files for Audio Coding Module 2 | 3 | testfile32kHz.pcm - mono speech file samples at 32 kHz 4 | teststereo32kHz.pcm - stereo speech file samples at 32 kHz 5 | 6 | Test and reference vectors to verify correct execution of PacketCable 7 | iLBC Fixed Point Reference Code 8 | 9 | Version 1.0.6 10 | Format: all .INP and .OUT files contain 16 bit sampled data using the 11 | Intel (PC) format. The .BIT files are stored in the appropriate byte 12 | sequence (big-endian format). 13 | 14 | *.INP - input files 15 | *.BIT20 - bit stream files 20 ms mode 16 | *.OUT20 - output files 20 ms mode (on a channel without packet loss) 17 | *.BIT30 - bit stream files 30 ms mode 18 | *.OUT30 - output files 30 ms mode (on a channel without packet loss) 19 | -------------------------------------------------------------------------------- /webrtc/sound/sound_tests.gypi: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2014 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 | 'includes': [ '../build/common.gypi', ], 11 | 'targets': [ 12 | { 13 | 'target_name': 'rtc_sound_tests', 14 | 'type': 'none', 15 | 'direct_dependent_settings': { 16 | 'sources': [ 17 | 'automaticallychosensoundsystem_unittest.cc', 18 | ], 19 | }, 20 | }, 21 | ], 22 | } 23 | 24 | -------------------------------------------------------------------------------- /talk/examples/android/ant.properties: -------------------------------------------------------------------------------- 1 | # This file is used to override default values used by the Ant build system. 2 | # 3 | # This file must be checked into Version Control Systems, as it is 4 | # integral to the build system of your project. 5 | 6 | # This file is only used by the Ant script. 7 | 8 | # You can use this to override default values such as 9 | # 'source.dir' for the location of your java source folder and 10 | # 'out.dir' for the location of your output folder. 11 | 12 | # You can also use it define how the release builds are signed by declaring 13 | # the following properties: 14 | # 'key.store' for the location of your keystore and 15 | # 'key.alias' for the name of the key to use. 16 | # The password will be asked during the build when you use the 'release' target. 17 | 18 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/automated_mode.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 SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_ 12 | #define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_ 13 | 14 | void InitializeGoogleTest(int* argc, char** argv); 15 | int RunInAutomatedMode(); 16 | 17 | #endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_ 18 | -------------------------------------------------------------------------------- /webrtc/rtc_unittests.isolate: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2014 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 | 'conditions': [ 10 | ['OS=="linux" or OS=="mac" or OS=="win"', { 11 | 'variables': { 12 | 'command': [ 13 | '<(PRODUCT_DIR)/rtc_unittests<(EXECUTABLE_SUFFIX)', 14 | ], 15 | 'files': [ 16 | '<(PRODUCT_DIR)/rtc_unittests<(EXECUTABLE_SUFFIX)', 17 | ], 18 | }, 19 | }], 20 | ], 21 | } 22 | -------------------------------------------------------------------------------- /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 WEBRTC_TEST_RUN_TEST_H 11 | #define WEBRTC_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 // WEBRTC_TEST_RUN_TEST_H 23 | -------------------------------------------------------------------------------- /webrtc/tools/rtcbot/test/ping_pong.js: -------------------------------------------------------------------------------- 1 | // Copyright (c) 2014 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 | function testPingPong(test, bot) { 10 | test.assert(typeof bot.ping === 'function', 'Bot does not exposes ping.'); 11 | 12 | bot.ping(gotAnswer); 13 | 14 | function gotAnswer(answer) { 15 | test.log('bot > ' + answer); 16 | test.done(); 17 | } 18 | } 19 | 20 | registerBotTest('testPingPong/chrome', testPingPong, ['chrome']); 21 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/main/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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_ 12 | #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_ 13 | 14 | class ACMTest { 15 | public: 16 | ACMTest() {} 17 | virtual ~ACMTest() {} 18 | virtual void Perform() = 0; 19 | }; 20 | 21 | #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_ 22 | -------------------------------------------------------------------------------- /webrtc/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 and uses basictypes.h 9 | 10 | #ifndef WEBRTC_BASE_SHA1_H_ 11 | #define WEBRTC_BASE_SHA1_H_ 12 | 13 | #include "webrtc/base/basictypes.h" 14 | 15 | namespace rtc { 16 | 17 | struct SHA1_CTX { 18 | uint32 state[5]; 19 | // TODO: Change bit count to uint64. 20 | uint32 count[2]; // Bit count of input. 21 | uint8 buffer[64]; 22 | }; 23 | 24 | #define SHA1_DIGEST_SIZE 20 25 | 26 | void SHA1Init(SHA1_CTX* context); 27 | void SHA1Update(SHA1_CTX* context, const uint8* data, size_t len); 28 | void SHA1Final(SHA1_CTX* context, uint8 digest[SHA1_DIGEST_SIZE]); 29 | 30 | #endif // WEBRTC_BASE_SHA1_H_ 31 | 32 | } // namespace rtc 33 | -------------------------------------------------------------------------------- /webrtc/test/testsupport/always_passing_unittest.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 "testing/gtest/include/gtest/gtest.h" 12 | 13 | namespace webrtc { 14 | 15 | // A test that always passes. Useful when all tests in a executable are 16 | // disabled, since a gtest returns exit code 1 if no tests have executed. 17 | TEST(AlwaysPassingTest, AlwaysPassingTest) {} 18 | 19 | } // namespace webrtc 20 | -------------------------------------------------------------------------------- /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 "webrtc/test/null_transport.h" 11 | 12 | namespace webrtc { 13 | namespace test { 14 | 15 | bool NullTransport::SendRtp(const uint8_t* packet, size_t length) { 16 | return true; 17 | } 18 | 19 | bool NullTransport::SendRtcp(const uint8_t* packet, size_t length) { 20 | return true; 21 | } 22 | 23 | } // namespace test 24 | } // namespace webrtc 25 | -------------------------------------------------------------------------------- /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 WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_ 11 | #define WEBRTC_VIDEO_ENGINE_TEST_COMMON_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 // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_ 23 | -------------------------------------------------------------------------------- /talk/examples/androidtests/ant.properties: -------------------------------------------------------------------------------- 1 | # This file is used to override default values used by the Ant build system. 2 | # 3 | # This file must be checked into Version Control Systems, as it is 4 | # integral to the build system of your project. 5 | 6 | # This file is only used by the Ant script. 7 | 8 | # You can use this to override default values such as 9 | # 'source.dir' for the location of your java source folder and 10 | # 'out.dir' for the location of your output folder. 11 | 12 | # You can also use it define how the release builds are signed by declaring 13 | # the following properties: 14 | # 'key.store' for the location of your keystore and 15 | # 'key.alias' for the name of the key to use. 16 | # The password will be asked during the build when you use the 'release' target. 17 | 18 | tested.project.dir=../android 19 | -------------------------------------------------------------------------------- /webrtc/base/scoped_autorelease_pool.mm: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright 2008 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 12 | 13 | #import "webrtc/base/scoped_autorelease_pool.h" 14 | 15 | namespace rtc { 16 | 17 | ScopedAutoreleasePool::ScopedAutoreleasePool() { 18 | pool_ = [[NSAutoreleasePool alloc] init]; 19 | } 20 | 21 | ScopedAutoreleasePool::~ScopedAutoreleasePool() { 22 | [pool_ drain]; 23 | } 24 | 25 | } // namespace rtc 26 | -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/NativeWebRtcContextRegistry.java: -------------------------------------------------------------------------------- 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 | package org.webrtc.webrtcdemo; 12 | 13 | import android.content.Context; 14 | 15 | public class NativeWebRtcContextRegistry { 16 | static { 17 | System.loadLibrary("webrtcdemo-jni"); 18 | } 19 | 20 | public native void register(Context context); 21 | public native void unRegister(); 22 | } -------------------------------------------------------------------------------- /webrtc/modules/desktop_capture/window_capturer.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 "webrtc/modules/desktop_capture/window_capturer.h" 12 | 13 | #include "webrtc/modules/desktop_capture/desktop_capture_options.h" 14 | 15 | namespace webrtc { 16 | 17 | // static 18 | WindowCapturer* WindowCapturer::Create() { 19 | return Create(DesktopCaptureOptions::CreateDefault()); 20 | } 21 | 22 | } // namespace webrtc 23 | -------------------------------------------------------------------------------- /talk/app/webrtc/androidtests/ant.properties: -------------------------------------------------------------------------------- 1 | # This file is used to override default values used by the Ant build system. 2 | # 3 | # This file must be checked into Version Control Systems, as it is 4 | # integral to the build system of your project. 5 | 6 | # This file is only used by the Ant script. 7 | 8 | # You can use this to override default values such as 9 | # 'source.dir' for the location of your java source folder and 10 | # 'out.dir' for the location of your output folder. 11 | 12 | # You can also use it define how the release builds are signed by declaring 13 | # the following properties: 14 | # 'key.store' for the location of your keystore and 15 | # 'key.alias' for the name of the key to use. 16 | # The password will be asked during the build when you use the 'release' target. 17 | 18 | source.dir=../java/testcommon/src;src -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/i420/main/source/i420.gypi: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2011 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 | 'targets': [ 11 | { 12 | 'target_name': 'webrtc_i420', 13 | 'type': 'static_library', 14 | 'dependencies': [ 15 | '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', 16 | ], 17 | 'sources': [ 18 | '../interface/i420.h', 19 | 'i420.cc', 20 | ], 21 | }, 22 | ], 23 | } 24 | -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/beamformer/mock_beamformer.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 "webrtc/modules/audio_processing/beamformer/mock_beamformer.h" 12 | 13 | #include 14 | 15 | namespace webrtc { 16 | 17 | MockBeamformer::MockBeamformer(const std::vector& array_geometry) 18 | : Beamformer(array_geometry) {} 19 | 20 | MockBeamformer::~MockBeamformer() {} 21 | 22 | } // namespace webrtc 23 | -------------------------------------------------------------------------------- /webrtc/sound/alsasymboltable.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 "webrtc/sound/alsasymboltable.h" 12 | 13 | namespace rtc { 14 | 15 | #define LATE_BINDING_SYMBOL_TABLE_CLASS_NAME ALSA_SYMBOLS_CLASS_NAME 16 | #define LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST ALSA_SYMBOLS_LIST 17 | #define LATE_BINDING_SYMBOL_TABLE_DLL_NAME "libasound.so.2" 18 | #include "webrtc/base/latebindingsymboltable.cc.def" 19 | 20 | } // namespace rtc 21 | -------------------------------------------------------------------------------- /webrtc/p2p/base/common.h: -------------------------------------------------------------------------------- 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 | #ifndef WEBRTC_P2P_BASE_COMMON_H_ 12 | #define WEBRTC_P2P_BASE_COMMON_H_ 13 | 14 | #include "webrtc/base/logging.h" 15 | 16 | // Common log description format for jingle messages 17 | #define LOG_J(sev, obj) LOG(sev) << "Jingle:" << obj->ToString() << ": " 18 | #define LOG_JV(sev, obj) LOG_V(sev) << "Jingle:" << obj->ToString() << ": " 19 | 20 | #endif // WEBRTC_P2P_BASE_COMMON_H_ 21 | -------------------------------------------------------------------------------- /webrtc/sound/platformsoundsystem.h: -------------------------------------------------------------------------------- 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 | #ifndef WEBRTC_SOUND_PLATFORMSOUNDSYSTEM_H_ 12 | #define WEBRTC_SOUND_PLATFORMSOUNDSYSTEM_H_ 13 | 14 | namespace rtc { 15 | 16 | class SoundSystemInterface; 17 | 18 | // Creates the sound system implementation for this platform. 19 | SoundSystemInterface *CreatePlatformSoundSystem(); 20 | 21 | } // namespace rtc 22 | 23 | #endif // WEBRTC_SOUND_PLATFORMSOUNDSYSTEM_H_ 24 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc: -------------------------------------------------------------------------------- 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 | #include "webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h" 11 | 12 | #include "webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h" 13 | 14 | ViEAutoTestWindowManagerInterface* 15 | ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() { 16 | return new ViEAutoTestWindowManager(); 17 | } 18 | -------------------------------------------------------------------------------- /webrtc/modules/desktop_capture/mouse_cursor_shape.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 WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_SHAPE_H_ 12 | #define WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_SHAPE_H_ 13 | 14 | // This file is no longer needed, but some code in chromium still includes it. 15 | // TODO(sergeyu): Cleanup dependencies in chromium and remove this file. 16 | 17 | #endif // WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_SHAPE_H_ 18 | -------------------------------------------------------------------------------- /webrtc/sound/linuxsoundsystem.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 "webrtc/sound/linuxsoundsystem.h" 12 | 13 | #include "webrtc/sound/alsasoundsystem.h" 14 | #include "webrtc/sound/pulseaudiosoundsystem.h" 15 | 16 | namespace rtc { 17 | 18 | const SoundSystemCreator kLinuxSoundSystemCreators[] = { 19 | #ifdef HAVE_LIBPULSE 20 | &PulseAudioSoundSystem::Create, 21 | #endif 22 | &AlsaSoundSystem::Create, 23 | }; 24 | 25 | } // namespace rtc 26 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc: -------------------------------------------------------------------------------- 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 | #include "webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h" 12 | 13 | #include "webrtc/video_engine/test/auto_test/interface/vie_autotest_linux.h" 14 | 15 | ViEAutoTestWindowManagerInterface* 16 | ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() { 17 | return new ViEAutoTestWindowManager(); 18 | } 19 | -------------------------------------------------------------------------------- /webrtc/examples/android/media_demo/res/layout/dropdownitems.xml: -------------------------------------------------------------------------------- 1 | 2 | 7 | 17 | -------------------------------------------------------------------------------- /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 "webrtc/modules/audio_coding/neteq/post_decode_vad.h" 14 | 15 | #include "testing/gtest/include/gtest/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/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 "webrtc/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/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 "webrtc/modules/audio_coding/neteq/random_vector.h" 14 | 15 | #include "testing/gtest/include/gtest/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 | -------------------------------------------------------------------------------- /webrtc/modules/video_render/test/testAPI/testAPI.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 WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H 12 | #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H 13 | 14 | #include "webrtc/modules/video_render/include/video_render_defines.h" 15 | 16 | void RunVideoRenderTests(void* window, webrtc::VideoRenderType windowType); 17 | 18 | #endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H 19 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/primitives/fake_stdin.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 | #ifndef FAKE_STDIN_H_ 11 | #define FAKE_STDIN_H_ 12 | 13 | #include 14 | 15 | #include 16 | 17 | #include "testing/gtest/include/gtest/gtest.h" 18 | 19 | namespace webrtc { 20 | 21 | // Creates a fake stdin-like FILE* for unit test usage. 22 | FILE* FakeStdin(const std::string& input); 23 | 24 | } // namespace webrtc 25 | 26 | #endif // FAKE_STDIN_H_ 27 | -------------------------------------------------------------------------------- /talk/examples/android/res/values/arrays.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | Default 5 | HD (1280 x 720) 6 | VGA (640 x 480) 7 | QVGA (320 x 240) 8 | 9 | 10 | 11 | Default 12 | 1280 x 720 13 | 640 x 480 14 | 320 x 240 15 | 16 | 17 | 18 | Default 19 | 30 fps 20 | 15 fps 21 | 22 | 23 | 24 | Default 25 | Manual 26 | 27 | 28 | 29 | -------------------------------------------------------------------------------- /talk/build/objc_app.plist: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | CFBundleDevelopmentRegion 6 | en 7 | CFBundleDisplayName 8 | ${PRODUCT_NAME} 9 | CFBundleExecutable 10 | ${EXECUTABLE_NAME} 11 | CFBundleIdentifier 12 | com.Google.${PRODUCT_NAME:rfc1034identifier} 13 | CFBundleInfoDictionaryVersion 14 | 6.0 15 | CFBundleName 16 | ${PRODUCT_NAME} 17 | CFBundlePackageType 18 | APPL 19 | CFBundleShortVersionString 20 | 1.0 21 | CFBundleVersion 22 | 1.0 23 | 24 | 25 | --------------------------------------------------------------------------------