├── tools ├── perf │ ├── __init__.py │ └── perf ├── e2e_quality │ ├── audio │ │ ├── perf │ │ ├── daemon.conf │ │ └── default.pa │ └── e2e_quality.gyp ├── valgrind-webrtc │ ├── memcheck │ │ ├── OWNERS │ │ ├── suppressions_mac.txt │ │ └── suppressions_win32.txt │ └── tsan │ │ ├── OWNERS │ │ ├── suppressions_mac.txt │ │ ├── suppressions_win32.txt │ │ └── suppressions.txt ├── python_charts │ ├── OWNERS │ ├── gviz_api.py │ ├── app.yaml │ └── webrtc │ │ └── __init__.py ├── quality_tracking │ ├── oauth2 │ ├── OWNERS │ └── dashboard │ │ ├── gaeunit.py │ │ ├── gviz_api.py │ │ ├── static │ │ └── google403c95edcde16425.html │ │ ├── index.yaml │ │ ├── app.yaml │ │ ├── lkgr_page.py │ │ ├── stylesheets │ │ └── stylesheet.css │ │ └── main.py ├── resources │ └── OWNERS ├── coverity │ └── OWNERS ├── OWNERS ├── perf_expectations │ ├── README │ └── webrtc_perf_expectations.cfg ├── .gitignore ├── codereview.settings ├── DEPS ├── create_supplement_gypi.py ├── refactoring │ ├── trim.py │ └── p4commands.py └── matlab │ └── maxUnwrap.m ├── PATENTS ├── LICENSE ├── samples └── js │ ├── apprtc │ ├── js │ │ └── adapter.js │ ├── images │ │ └── webrtc_black_20p.png │ ├── OWNERS │ ├── html │ │ └── help.html │ └── app.yaml │ └── demos │ ├── js │ └── demosite │ │ ├── favicon.ico │ │ └── app.yaml │ ├── images │ └── webrtc_black_20p.png │ ├── OWNERS │ ├── index.yaml │ └── html │ ├── gum1.html │ └── gum2.html ├── webrtc ├── modules │ ├── video_capture │ │ ├── mac │ │ │ ├── video_capture_mac.h │ │ │ └── qtkit │ │ │ │ ├── video_capture_recursive_lock.mm │ │ │ │ └── video_capture_recursive_lock.h │ │ ├── android │ │ │ ├── OWNERS │ │ │ └── java │ │ │ │ └── org │ │ │ │ └── webrtc │ │ │ │ └── videoengine │ │ │ │ └── CaptureCapabilityAndroid.java │ │ ├── OWNERS │ │ ├── test │ │ │ └── video_capture_main_mac.mm │ │ ├── external │ │ │ └── video_capture_external.cc │ │ ├── video_capture_delay.h │ │ └── video_capture_factory.cc │ ├── audio_device │ │ ├── android │ │ │ └── OWNERS │ │ ├── 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 │ │ │ │ │ ├── gen │ │ │ │ │ └── org │ │ │ │ │ │ └── webrtc │ │ │ │ │ │ └── voiceengine │ │ │ │ │ │ └── test │ │ │ │ │ │ └── R.java │ │ │ │ │ ├── AndroidManifest.xml │ │ │ │ │ ├── jni │ │ │ │ │ └── org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h │ │ │ │ │ └── .project │ │ │ └── README.txt │ │ ├── main │ │ │ ├── source │ │ │ │ └── audio_device.gypi │ │ │ └── interface │ │ │ │ └── audio_device.h │ │ ├── dummy │ │ │ └── audio_device_utility_dummy.h │ │ ├── audio_device_utility.h │ │ ├── mac │ │ │ └── audio_device_utility_mac.h │ │ └── linux │ │ │ └── audio_device_utility_linux.h │ ├── video_render │ │ ├── android │ │ │ └── OWNERS │ │ ├── OWNERS │ │ ├── test │ │ │ └── testAPI │ │ │ │ ├── renderStartImage.bmp │ │ │ │ ├── testAPI_android.cc │ │ │ │ └── testAPI.h │ │ └── mac │ │ │ ├── cocoa_full_screen_window.h │ │ │ └── cocoa_render_view.h │ ├── audio_processing │ │ ├── OWNERS │ │ ├── test │ │ │ └── android │ │ │ │ └── apmtest │ │ │ │ ├── jni │ │ │ │ ├── Application.mk │ │ │ │ └── Android.mk │ │ │ │ ├── res │ │ │ │ └── values │ │ │ │ │ └── strings.xml │ │ │ │ └── default.properties │ │ ├── utility │ │ │ └── fft4g.h │ │ └── debug.proto │ ├── audio_coding │ │ ├── neteq │ │ │ ├── OWNERS │ │ │ ├── test │ │ │ │ ├── ptypes.txt │ │ │ │ └── NETEQTEST_DummyRTPpacket.h │ │ │ ├── mute_signal.c │ │ │ └── mcu_address_init.c │ │ ├── codecs │ │ │ ├── OWNERS │ │ │ ├── isac │ │ │ │ ├── fix │ │ │ │ │ └── test │ │ │ │ │ │ └── QA │ │ │ │ │ │ ├── ChannelFiles.txt │ │ │ │ │ │ ├── InputFilesFew.txt │ │ │ │ │ │ ├── ListOfTestCases.xls │ │ │ │ │ │ ├── diffiSACPLC.txt │ │ │ │ │ │ ├── InputFiles.txt │ │ │ │ │ │ ├── runiSACPLC.txt │ │ │ │ │ │ └── runiSACfault.txt │ │ │ │ └── isacfix_test.gypi │ │ │ ├── cng │ │ │ │ ├── test │ │ │ │ │ └── StdAfx.cc │ │ │ │ ├── cng_helpfuns.h │ │ │ │ └── cng.gypi │ │ │ ├── pcm16b │ │ │ │ ├── pcm16b.gypi │ │ │ │ └── Android.mk │ │ │ ├── ilbc │ │ │ │ ├── index_conv_dec.h │ │ │ │ ├── poly_to_lsf.c │ │ │ │ ├── frame_classify.h │ │ │ │ └── get_cd_vec.h │ │ │ ├── opus │ │ │ │ └── opus.gypi │ │ │ ├── g711 │ │ │ │ └── Android.mk │ │ │ └── g722 │ │ │ │ └── Android.mk │ │ ├── main │ │ │ ├── OWNERS │ │ │ └── test │ │ │ │ ├── ACMTest.cc │ │ │ │ ├── ACMTest.h │ │ │ │ └── TimedTrace.h │ │ └── neteq4 │ │ │ ├── OWNERS │ │ │ ├── post_decode_vad_unittest.cc │ │ │ ├── random_vector_unittest.cc │ │ │ ├── test │ │ │ └── NETEQTEST_DummyRTPpacket.h │ │ │ ├── background_noise_unittest.cc │ │ │ ├── comfort_noise_unittest.cc │ │ │ ├── time_stretch_unittest.cc │ │ │ └── expand_unittest.cc │ ├── audio_conference_mixer │ │ ├── OWNERS │ │ └── source │ │ │ ├── audio_frame_manipulator.h │ │ │ └── level_indicator.h │ ├── desktop_capture │ │ ├── OWNERS │ │ ├── shared_memory.cc │ │ └── desktop_geometry.cc │ ├── utility │ │ ├── OWNERS │ │ └── interface │ │ │ └── process_thread.h │ ├── pacing │ │ └── OWNERS │ ├── media_file │ │ └── OWNERS │ ├── video_coding │ │ ├── OWNERS │ │ ├── codecs │ │ │ ├── OWNERS │ │ │ ├── i420 │ │ │ │ └── main │ │ │ │ │ └── source │ │ │ │ │ └── i420.gypi │ │ │ └── vp8 │ │ │ │ └── include │ │ │ │ ├── vp8.h │ │ │ │ └── vp8_common_types.h │ │ ├── main │ │ │ └── test │ │ │ │ ├── release_test.h │ │ │ │ ├── subfigure.m │ │ │ │ ├── pcap_file_reader.h │ │ │ │ ├── rtp_file_reader.h │ │ │ │ └── release_test_pt2.cc │ │ └── utility │ │ │ ├── video_coding_utility.gyp │ │ │ └── Android.mk │ ├── video_processing │ │ └── main │ │ │ ├── OWNERS │ │ │ ├── test │ │ │ ├── unit_test │ │ │ │ └── writeYUV420file.m │ │ │ └── vpm_tests.gypi │ │ │ └── source │ │ │ ├── color_enhancement.h │ │ │ └── brighten.h │ ├── rtp_rtcp │ │ ├── OWNERS │ │ └── source │ │ │ ├── rtp_format_video_generic.h │ │ │ ├── video_codec_information.h │ │ │ └── rtp_receiver_strategy.cc │ ├── remote_bitrate_estimator │ │ ├── OWNERS │ │ └── include │ │ │ └── mock │ │ │ └── mock_remote_bitrate_observer.h │ └── bitrate_controller │ │ └── OWNERS ├── tools │ ├── OWNERS │ └── barcode_tools │ │ └── DEPS ├── voice_engine │ ├── test │ │ ├── android │ │ │ ├── OWNERS │ │ │ └── android_test │ │ │ │ ├── res │ │ │ │ ├── drawable │ │ │ │ │ └── icon.png │ │ │ │ └── values │ │ │ │ │ └── strings.xml │ │ │ │ ├── .classpath │ │ │ │ ├── default.properties │ │ │ │ ├── jni │ │ │ │ └── Application.mk │ │ │ │ ├── Android.mk │ │ │ │ ├── .project │ │ │ │ └── gen │ │ │ │ └── org │ │ │ │ └── webrtc │ │ │ │ └── voiceengine │ │ │ │ └── test │ │ │ │ └── R.java │ │ ├── win_test │ │ │ ├── WinTest.aps │ │ │ ├── res │ │ │ │ ├── WinTest.ico │ │ │ │ └── WinTest.rc2 │ │ │ ├── stdafx.cc │ │ │ └── WinTest.h │ │ └── auto_test │ │ │ ├── automated_mode.h │ │ │ ├── automated_mode.cc │ │ │ ├── voe_cpu_test.h │ │ │ ├── resource_manager.cc │ │ │ ├── resource_manager.h │ │ │ └── standard │ │ │ ├── voe_base_misc_test.cc │ │ │ └── hardware_before_initializing_test.cc │ ├── OWNERS │ ├── channel_unittest.cc │ ├── voice_engine.gyp │ ├── include │ │ └── mock │ │ │ ├── mock_voe_connection_observer.h │ │ │ └── mock_voe_observer.h │ ├── voe_neteq_stats_impl.h │ └── voe_encryption_impl.h ├── video_engine │ ├── test │ │ ├── android │ │ │ ├── OWNERS │ │ │ ├── res │ │ │ │ ├── drawable │ │ │ │ │ └── logo.png │ │ │ │ └── layout │ │ │ │ │ ├── send.xml │ │ │ │ │ ├── row.xml │ │ │ │ │ └── tabhost.xml │ │ │ ├── project.properties │ │ │ ├── jni │ │ │ │ └── Application.mk │ │ │ └── src │ │ │ │ └── org │ │ │ │ └── webrtc │ │ │ │ └── videoengineapp │ │ │ │ └── IViEAndroidCallback.java │ │ └── auto_test │ │ │ ├── android │ │ │ ├── res │ │ │ │ ├── drawable │ │ │ │ │ └── logo.png │ │ │ │ └── values │ │ │ │ │ └── strings.xml │ │ │ ├── default.properties │ │ │ ├── .classpath │ │ │ └── .project │ │ │ ├── source │ │ │ ├── vie_window_manager_factory_win.cc │ │ │ ├── vie_window_manager_factory_linux.cc │ │ │ └── vie_window_manager_factory_mac.mm │ │ │ ├── primitives │ │ │ ├── fake_stdin.h │ │ │ └── fake_stdin.cc │ │ │ ├── automated │ │ │ └── legacy_fixture.cc │ │ │ └── interface │ │ │ ├── vie_autotest_android.h │ │ │ └── vie_window_manager_factory.h │ ├── OWNERS │ ├── video_engine.gyp │ └── vie_ref_count.h ├── .gitignore ├── test │ ├── OWNERS │ ├── channel_transport │ │ └── OWNERS │ ├── manual │ │ ├── README │ │ ├── iframe-video.html │ │ └── iframe-apprtc.html │ ├── run_all_unittests.cc │ ├── libtest │ │ └── libtest.gyp │ ├── testsupport │ │ ├── mock │ │ │ ├── mock_frame_writer.h │ │ │ └── mock_frame_reader.h │ │ ├── mac │ │ │ └── run_threaded_main_mac.h │ │ └── perf_test_unittest.cc │ └── Android.mk ├── common_video │ └── OWNERS ├── common_audio │ ├── OWNERS │ └── signal_processing │ │ ├── spl_version.c │ │ ├── webrtc_fft_t_rad.c │ │ └── energy.c ├── system_wrappers │ ├── OWNERS │ ├── source │ │ ├── cpu_features_android.c │ │ ├── logging_no_op.cc │ │ ├── critical_section.cc │ │ ├── critical_section_win.cc │ │ ├── clock_unittest.cc │ │ ├── tick_util.cc │ │ ├── sleep.cc │ │ ├── event.cc │ │ ├── thread.cc │ │ ├── critical_section_posix.h │ │ ├── rw_lock.cc │ │ ├── rw_lock_win.h │ │ └── rw_lock_posix.h │ └── interface │ │ ├── cpu_info.h │ │ ├── compile_assert.h │ │ └── sleep.h ├── README.chromium └── build │ ├── no_op.cc │ ├── arm_neon.gypi │ └── adb_shell.sh ├── 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_tiny11.wav │ ├── audio_tiny16.wav │ ├── audio_tiny22.wav │ ├── audio_tiny32.wav │ ├── audio_tiny44.wav │ ├── audio_tiny48.wav │ ├── audio_tiny8.wav │ ├── audio_short16.pcm │ ├── audio_long16noise.pcm │ ├── audio_long8mulaw.wav │ ├── audio_long16big_endian.pcm │ └── stereo_rtp_files │ │ ├── rtpplay.exe │ │ ├── HRTF_pcm16wb.rtp │ │ ├── stereo_g729.rtp │ │ ├── stereo_pcmu.rtp │ │ ├── hrtf_g722_1C_48.rtp │ │ ├── stereo_pcm16wb.rtp │ │ ├── stereo_pcmu_vad.rtp │ │ ├── HRTF_pcm16wb_jitter.rtp │ │ ├── stereo_g729_jitter.rtp │ │ ├── stereo_pcmu_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 ├── OWNERS ├── codereview.settings ├── AUTHORS ├── license_template.txt └── third_party ├── google-gflags └── README.webrtc └── winsdk_samples └── README.webrtc /tools/perf/__init__.py: -------------------------------------------------------------------------------- 1 | -------------------------------------------------------------------------------- /tools/perf/perf: -------------------------------------------------------------------------------- 1 | ../../perf -------------------------------------------------------------------------------- /PATENTS: -------------------------------------------------------------------------------- 1 | Refer to src/PATENTS. 2 | -------------------------------------------------------------------------------- /LICENSE: -------------------------------------------------------------------------------- 1 | Refer to webrtc/LICENSE. 2 | -------------------------------------------------------------------------------- /tools/e2e_quality/audio/perf: -------------------------------------------------------------------------------- 1 | ../../perf 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| tina.legrand@webrtc.org 3 | -------------------------------------------------------------------------------- /tools/quality_tracking/dashboard/gviz_api.py: -------------------------------------------------------------------------------- 1 | ../../third_party/google-visualization-python/gviz_api.py -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/test/android/apmtest/jni/Application.mk: -------------------------------------------------------------------------------- 1 | APP_PLATFORM := android-9 2 | -------------------------------------------------------------------------------- /webrtc/.gitignore: -------------------------------------------------------------------------------- 1 | # This file is for projects that checkout webrtc/ directly (e.g. Chromium). 2 | *.mk 3 | -------------------------------------------------------------------------------- /webrtc/modules/audio_conference_mixer/OWNERS: 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henrik.lundin@webrtc.org 4 | mflodman@webrtc.org 5 | asapersson@webrtc.org -------------------------------------------------------------------------------- /data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/rillian/webrtc/HEAD/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp -------------------------------------------------------------------------------- /data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/rillian/webrtc/HEAD/data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp -------------------------------------------------------------------------------- /data/voice_engine/stereo_rtp_files/stereo_pcmu_vad.rtp: -------------------------------------------------------------------------------- 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pwestin@webrtc.org 2 | stefan@webrtc.org 3 | henrik.lundin@webrtc.org 4 | mflodman@webrtc.org 5 | asapersson@webrtc.org -------------------------------------------------------------------------------- /data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/rillian/webrtc/HEAD/data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp -------------------------------------------------------------------------------- /data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/rillian/webrtc/HEAD/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp -------------------------------------------------------------------------------- /tools/perf_expectations/webrtc_perf_expectations.cfg: -------------------------------------------------------------------------------- 1 | { 2 | "base_url": "http://www.corp.google.com/~webrtc-cb/perf/", 3 | "perf_file": "perf_expectations.json" 4 | } 5 | -------------------------------------------------------------------------------- /webrtc/modules/bitrate_controller/OWNERS: -------------------------------------------------------------------------------- 1 | pwestin@webrtc.org 2 | stefan@webrtc.org 3 | henrik.lundin@webrtc.org 4 | mflodman@webrtc.org 5 | asapersson@webrtc.org 6 | -------------------------------------------------------------------------------- /webrtc/modules/video_render/test/testAPI/renderStartImage.bmp: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/rillian/webrtc/HEAD/webrtc/modules/video_render/test/testAPI/renderStartImage.bmp -------------------------------------------------------------------------------- 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3 | misc2.pcm 4 | sinesweep.pcm 5 | speechmusic.pcm 6 | tone_cisco.pcm 7 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ListOfTestCases.xls: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/rillian/webrtc/HEAD/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ListOfTestCases.xls -------------------------------------------------------------------------------- /webrtc/system_wrappers/OWNERS: -------------------------------------------------------------------------------- 1 | henrike@webrtc.org 2 | pwestin@webrtc.org 3 | perkj@webrtc.org 4 | henrika@webrtc.org 5 | henrikg@webrtc.org 6 | mflodman@webrtc.org 7 | niklas.enbom@webrtc.org -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/test/android/apmtest/res/values/strings.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | apmtest 4 | 5 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/src/org/webrtc/voiceengine/AudioDeviceAndroid.java: -------------------------------------------------------------------------------- 1 | ../../../../../../../source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java 2 | -------------------------------------------------------------------------------- /samples/js/demos/OWNERS: -------------------------------------------------------------------------------- 1 | juberti@webrtc.org 2 | braveyao@webrtc.org 3 | wu@webrtc.org 4 | hta@webrtc.org 5 | vikasmarwaha@webrtc.org 6 | phoglund@webrtc.org 7 | kjellander@webrtc.org 8 | henrika@webrtc.org 9 | 10 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/res/drawable/icon.png: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/rillian/webrtc/HEAD/webrtc/modules/audio_device/test/android/audio_device_android_test/res/drawable/icon.png -------------------------------------------------------------------------------- /OWNERS: -------------------------------------------------------------------------------- 1 | henrika@webrtc.org 2 | niklas.enbom@webrtc.org 3 | andrew@webrtc.org 4 | tina.legrand@webrtc.org 5 | tommi@webrtc.org 6 | per-file .gitignore=* 7 | per-file DEPS=* 8 | per-file AUTHORS=* 9 | per-file WATCHLISTS=* 10 | 11 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq/test/ptypes.txt: -------------------------------------------------------------------------------- 1 | pcmu 0 2 | pcma 8 3 | cn 13 4 | ilbc 102 5 | isac 103 6 | isacswb 104 7 | isacfb 124 8 | avt 106 9 | red 117 10 | cn_wb 98 11 | cn_swb32 99 12 | pcm16b 93 13 | pcm16b_wb 94 14 | pcm16b_swb32khz 95 15 | g722 9 16 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /samples/js/demos/js/demosite/app.yaml: -------------------------------------------------------------------------------- 1 | application: webrtc-demos 2 | version: 1 3 | runtime: python27 4 | api_version: 1 5 | threadsafe: yes 6 | 7 | handlers: 8 | - url: .* 9 | script: main.app 10 | secure: always 11 | 12 | libraries: 13 | - name: webapp2 14 | version: "2.5.1" 15 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /samples/js/apprtc/html/help.html: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 6 | WebRtc Demo App Help 7 | 8 | 9 | TODO 10 | 11 | 12 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /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 8 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/.classpath: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | -------------------------------------------------------------------------------- /webrtc/test/manual/README: -------------------------------------------------------------------------------- 1 | ================================================================ 2 | WEBRTC MANUAL TESTS 3 | ================================================================ 4 | 5 | You will need to serve these files off some kind of web server. Currently, 6 | GetUserMedia does not work when run off a file:// URL. 7 | 8 | Contact person: phoglund@webrtc.org -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/.classpath: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /AUTHORS: -------------------------------------------------------------------------------- 1 | # Names should be added to this file like so: 2 | # Name or Organization 3 | 4 | Google Inc. 5 | Mozilla Foundation 6 | Intel Corporation 7 | Vonage Holdings Corp. 8 | MIPS Technologies 9 | Ben Strong 10 | Martin Storsjo 11 | Jie Mao 12 | Anil Kumar 13 | Opera Software ASA 14 | Silviu Caragea 15 | -------------------------------------------------------------------------------- /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/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 | -------------------------------------------------------------------------------- /samples/js/apprtc/app.yaml: -------------------------------------------------------------------------------- 1 | application: apprtc 2 | version: 6 3 | runtime: python27 4 | threadsafe: true 5 | api_version: 1 6 | 7 | handlers: 8 | - url: /html 9 | static_dir: html 10 | 11 | - url: /images 12 | static_dir: images 13 | 14 | - url: /js 15 | static_dir: js 16 | 17 | - url: /.* 18 | script: apprtc.app 19 | secure: always 20 | 21 | inbound_services: 22 | - channel_presence 23 | 24 | libraries: 25 | - name: jinja2 26 | version: latest 27 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /third_party/google-gflags/README.webrtc: -------------------------------------------------------------------------------- 1 | URL: http://code.google.com/p/google-gflags/ 2 | Version: 1.5 3 | License: New BSD 4 | License File: LICENSE 5 | 6 | Description: 7 | The gflags package contains a library that implements commandline 8 | flags processing. As such it's a replacement for getopt(). It has 9 | increased flexibility, including built-in support for C++ types like 10 | string, and the ability to define flags in the source file in which 11 | they're used. 12 | 13 | Local Modifications: None 14 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /samples/js/demos/index.yaml: -------------------------------------------------------------------------------- 1 | indexes: 2 | 3 | # AUTOGENERATED 4 | 5 | # This index.yaml is automatically updated whenever the dev_appserver 6 | # detects that a new type of query is run. If you want to manage the 7 | # index.yaml file manually, remove the above marker line (the line 8 | # saying "# AUTOGENERATED"). If you want to manage some indexes 9 | # manually, move them above the marker line. The index.yaml file is 10 | # automatically uploaded to the admin console when you next deploy 11 | # your application using appcfg.py. 12 | 13 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/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 use, 7 | # "ant.properties", and override values to adapt the script to your 8 | # project structure. 9 | 10 | # Indicates whether an apk should be generated for each density. 11 | split.density=false 12 | # Project target. 13 | target=android-10 14 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/res/layout/main.xml: -------------------------------------------------------------------------------- 1 | 2 | 6 | 7 | 12 | 13 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/main/test/ACMTest.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 "ACMTest.h" 12 | 13 | ACMTest::~ACMTest() {} 14 | 15 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/android/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-v7a #armeabi armeabi-v7a x86 11 | APP_STL := stlport_static 12 | -------------------------------------------------------------------------------- /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/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/modules/video_capture/mac/qtkit/video_capture_recursive_lock.mm: -------------------------------------------------------------------------------- 1 | // 2 | // video_capture_recursive_lock.mm 3 | // 4 | // 5 | 6 | #import "video_capture_recursive_lock.h" 7 | 8 | @implementation VideoCaptureRecursiveLock 9 | 10 | @synthesize locked = _locked; 11 | 12 | - (id)init{ 13 | self = [super init]; 14 | if(nil == self){ 15 | return nil; 16 | } 17 | 18 | [self setLocked:NO]; 19 | return self; 20 | } 21 | 22 | - (void)lock{ 23 | [self setLocked:YES]; 24 | [super lock]; 25 | } 26 | 27 | - (void)unlock{ 28 | [self setLocked:NO]; 29 | [super unlock]; 30 | } 31 | 32 | 33 | @end 34 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /tools/quality_tracking/dashboard/index.yaml: -------------------------------------------------------------------------------- 1 | indexes: 2 | 3 | # AUTOGENERATED 4 | 5 | # This index.yaml is automatically updated whenever the dev_appserver 6 | # detects that a new type of query is run. If you want to manage the 7 | # index.yaml file manually, remove the above marker line (the line 8 | # saying "# AUTOGENERATED"). If you want to manage some indexes 9 | # manually, move them above the marker line. The index.yaml file is 10 | # automatically uploaded to the admin console when you next deploy 11 | # your application using appcfg.py. 12 | 13 | - kind: CoverageData 14 | properties: 15 | - name: report_category 16 | - name: date 17 | -------------------------------------------------------------------------------- /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 "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/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 "channel.h" 12 | #include "gtest/gtest.h" 13 | 14 | // Empty test just to get coverage metrics. 15 | TEST(ChannelTest, EmptyTestToGetCodeCoverage) {} 16 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/android/res/layout/send.xml: -------------------------------------------------------------------------------- 1 | 2 | 6 | 7 | 8 | 11 | 12 | 15 | 16 | 17 | 18 | -------------------------------------------------------------------------------- /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 ACMTEST_H 12 | #define ACMTEST_H 13 | 14 | class ACMTest { 15 | public: 16 | virtual ~ACMTest() = 0; 17 | virtual void Perform() = 0; 18 | }; 19 | 20 | #endif 21 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/tsan/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/tsan/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 | -------------------------------------------------------------------------------- /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/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 quality_tracking. 7 | "tools/third_party/gaeunit": 8 | "http://code.google.com/p/gaeunit.git@e16d5bd4", 9 | 10 | # Used by quality_tracking. 11 | "tools/third_party/oauth2": 12 | "http://github.com/simplegeo/python-oauth2.git@a83f4a29", 13 | 14 | # Used by tools/quality_tracking/dashboard and tools/python_charts. 15 | "tools/third_party/google-visualization-python": 16 | "http://google-visualization-python.googlecode.com/svn/trunk/@15", 17 | } 18 | 19 | -------------------------------------------------------------------------------- /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 | -------------------------------------------------------------------------------- /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/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 "stdafx.h" 16 | 17 | 18 | -------------------------------------------------------------------------------- /webrtc/modules/video_capture/android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java: -------------------------------------------------------------------------------- 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 | package org.webrtc.videoengine; 12 | 13 | public class CaptureCapabilityAndroid { 14 | public int width = 0; 15 | public int height = 0; 16 | public int maxFPS = 0; 17 | } 18 | -------------------------------------------------------------------------------- /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 "gtest/gtest.h" 12 | #include "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 "../../include/audio_device.h" 15 | 16 | #endif // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_ 17 | -------------------------------------------------------------------------------- /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/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 | #if defined(WEBRTC_CHROMIUM_BUILD) 12 | #include 13 | #else 14 | #include "android/cpu-features.h" 15 | #endif // defined(WEBRTC_CHROMIUM_BUILD) 16 | 17 | uint64_t WebRtc_GetCPUFeaturesARM(void) { 18 | return android_getCpuFeatures(); 19 | } 20 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/android/src/org/webrtc/videoengineapp/IViEAndroidCallback.java: -------------------------------------------------------------------------------- 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 | package org.webrtc.videoengineapp; 12 | 13 | public interface IViEAndroidCallback { 14 | public int updateStats(int frameRateI, int bitRateI, 15 | int packetLoss, int frameRateO, 16 | int bitRateO); 17 | } 18 | -------------------------------------------------------------------------------- /samples/js/demos/html/gum1.html: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | getUserMedia Demo 1 5 | 16 | 17 | 18 | 19 |
20 | 21 | 31 | 32 | 33 | 34 | -------------------------------------------------------------------------------- /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/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 "vie_window_manager_factory.h" 11 | 12 | #include "vie_autotest_windows.h" 13 | 14 | ViEAutoTestWindowManagerInterface* 15 | ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() { 16 | return new ViEAutoTestWindowManager(); 17 | } 18 | -------------------------------------------------------------------------------- /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 "vie_window_manager_factory.h" 12 | 13 | #include "vie_autotest_linux.h" 14 | 15 | ViEAutoTestWindowManagerInterface* 16 | ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() { 17 | return new ViEAutoTestWindowManager(); 18 | } 19 | -------------------------------------------------------------------------------- /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/test/manual/iframe-video.html: -------------------------------------------------------------------------------- 1 | 2 | 11 | 12 | 13 | IFRAME Single Local Preview (Video Only) 14 | 15 | 16 | 17 | 18 | 19 | -------------------------------------------------------------------------------- /webrtc/voice_engine/voice_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 | 'voice_engine_core.gypi', 13 | ], 14 | 15 | # Test targets, excluded when building with Chromium. 16 | 'conditions': [ 17 | ['include_tests==1', { 18 | 'includes': [ 19 | 'test/voice_engine_tests.gypi', 20 | ], 21 | }], 22 | ], 23 | } 24 | -------------------------------------------------------------------------------- /tools/quality_tracking/dashboard/app.yaml: -------------------------------------------------------------------------------- 1 | application: webrtc-dashboard 2 | version: 1 3 | runtime: python27 4 | api_version: 1 5 | threadsafe: false 6 | 7 | handlers: 8 | # Serve stylesheets statically. 9 | - url: /stylesheets 10 | static_dir: stylesheets 11 | # This magic file is here to prove to the Google Account Domain Management 12 | # that we own this domain. It needs to stay there so the domain management 13 | # doesn't get suspicious. 14 | - url: /google403c95edcde16425.html 15 | static_files: static/google403c95edcde16425.html 16 | upload: static/google403c95edcde16425.html 17 | 18 | # Note: tests should be disabled in production. 19 | # - url: /test.* 20 | # script: gaeunit.py 21 | 22 | # Redirect all other requests to our dynamic handlers. 23 | - url: /.* 24 | script: main.app -------------------------------------------------------------------------------- /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/system_wrappers/source/logging_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 | #include "webrtc/system_wrappers/interface/logging.h" 12 | 13 | namespace webrtc { 14 | 15 | LogMessage::LogMessage(const char*, int, LoggingSeverity) { 16 | // Avoid an unused-private-field warning. 17 | (void)severity_; 18 | } 19 | 20 | LogMessage::~LogMessage() { 21 | } 22 | 23 | } // namespace webrtc 24 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/android/res/layout/row.xml: -------------------------------------------------------------------------------- 1 | 2 | 7 | 18 | 19 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/cng/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 | // CNG.pch will be the pre-compiled header 13 | // stdafx.obj will contain the pre-compiled type information 14 | 15 | #include "stdafx.h" 16 | 17 | // TODO: reference any additional headers you need in STDAFX.H 18 | // and not in this file 19 | -------------------------------------------------------------------------------- /webrtc/test/manual/iframe-apprtc.html: -------------------------------------------------------------------------------- 1 | 2 | 11 | 12 | 13 | AppRTC web app in an IFRAME 14 | 15 | 16 | 17 | AppRTC in an <iframe> element:
18 | 19 | 20 | 21 | -------------------------------------------------------------------------------- /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 "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/modules/audio_coding/neteq4/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/neteq4/post_decode_vad.h" 14 | 15 | #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/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 | #include 15 | 16 | #include "gtest/gtest.h" 17 | 18 | namespace webrtc { 19 | 20 | // Creates a fake stdin-like FILE* for unit test usage. 21 | FILE* FakeStdin(const std::string& input); 22 | 23 | } // namespace webrtc 24 | 25 | #endif // FAKE_STDIN_H_ 26 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq4/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/neteq4/random_vector.h" 14 | 15 | #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/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef NETEQTEST_DUMMYRTPPACKET_H 12 | #define NETEQTEST_DUMMYRTPPACKET_H 13 | 14 | #include "NETEQTEST_RTPpacket.h" 15 | 16 | class NETEQTEST_DummyRTPpacket : public NETEQTEST_RTPpacket 17 | { 18 | public: 19 | virtual int readFromFile(FILE *fp); 20 | virtual int writeToFile(FILE *fp); 21 | }; 22 | 23 | #endif //NETEQTEST_DUMMYRTPPACKET_H 24 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq4/test/NETEQTEST_DummyRTPpacket.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef NETEQTEST_DUMMYRTPPACKET_H 12 | #define NETEQTEST_DUMMYRTPPACKET_H 13 | 14 | #include "NETEQTEST_RTPpacket.h" 15 | 16 | class NETEQTEST_DummyRTPpacket : public NETEQTEST_RTPpacket 17 | { 18 | public: 19 | virtual int readFromFile(FILE *fp); 20 | virtual int writeToFile(FILE *fp); 21 | }; 22 | 23 | #endif //NETEQTEST_DUMMYRTPPACKET_H 24 | -------------------------------------------------------------------------------- /tools/create_supplement_gypi.py: -------------------------------------------------------------------------------- 1 | #!/usr/bin/env python 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 | import sys 11 | 12 | supplement_gypi = """#!/usr/bin/env python 13 | # This file is generated by %s. Not for check-in. 14 | # Please see the WebRTC DEPS file for details. 15 | { 16 | 'variables': { 17 | 'build_with_chromium': 0, 18 | } 19 | } 20 | """ 21 | 22 | def main(argv): 23 | open(argv[1], 'w').write(supplement_gypi % argv[0]) 24 | 25 | if __name__ == '__main__': 26 | sys.exit(main(sys.argv)) 27 | -------------------------------------------------------------------------------- /tools/refactoring/trim.py: -------------------------------------------------------------------------------- 1 | #!/usr/bin/env python 2 | 3 | import sys 4 | import fileinput 5 | 6 | # Defaults 7 | TABSIZE = 4 8 | 9 | usage = """ 10 | Replaces all TAB characters with %(TABSIZE)d space characters. 11 | In addition, all trailing space characters are removed. 12 | usage: trim file ... 13 | file ... : files are changed in place without taking any backup. 14 | """ % vars() 15 | 16 | def main(): 17 | 18 | if len(sys.argv) == 1: 19 | sys.stderr.write(usage) 20 | sys.exit(2) 21 | 22 | # Iterate over the lines of all files listed in sys.argv[1:] 23 | for line in fileinput.input(sys.argv[1:], inplace=True): 24 | line = line.replace('\t',' '*TABSIZE); # replace TABs 25 | line = line.rstrip(None) # remove trailing whitespaces 26 | print line # modify the file 27 | 28 | if __name__ == '__main__': 29 | main() 30 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/automated_mode.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 "gtest/gtest.h" 12 | #include "test/testsupport/fileutils.h" 13 | 14 | void InitializeGoogleTest(int* argc, char** argv) { 15 | // Initialize WebRTC testing framework so paths to resources can be resolved. 16 | webrtc::test::SetExecutablePath(argv[0]); 17 | testing::InitGoogleTest(argc, argv); 18 | } 19 | 20 | int RunInAutomatedMode() { 21 | return RUN_ALL_TESTS(); 22 | } 23 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq4/background_noise_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // Unit tests for BackgroundNoise class. 12 | 13 | #include "webrtc/modules/audio_coding/neteq4/background_noise.h" 14 | 15 | #include "gtest/gtest.h" 16 | 17 | namespace webrtc { 18 | 19 | TEST(BackgroundNoise, CreateAndDestroy) { 20 | size_t channels = 1; 21 | BackgroundNoise bgn(channels); 22 | } 23 | 24 | // TODO(hlundin): Write more tests. 25 | 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc/test/libtest/libtest.gyp: -------------------------------------------------------------------------------- 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 | 'includes': [ 10 | '../../build/common.gypi' 11 | ], 12 | 'targets': [ 13 | { 14 | 'target_name': 'libtest', 15 | 'type': 'static_library', 16 | 'sources': [ 17 | # Helper classes 18 | 'include/bit_flip_encryption.h', 19 | 'include/random_encryption.h', 20 | 21 | 'helpers/bit_flip_encryption.cc', 22 | 'helpers/random_encryption.cc', 23 | ], 24 | }, 25 | ], 26 | } 27 | -------------------------------------------------------------------------------- /webrtc/modules/video_processing/main/source/color_enhancement.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /* 12 | * color_enhancement.h 13 | */ 14 | #ifndef VPM_COLOR_ENHANCEMENT_H 15 | #define VPM_COLOR_ENHANCEMENT_H 16 | 17 | #include "typedefs.h" 18 | #include "video_processing.h" 19 | 20 | namespace webrtc { 21 | 22 | namespace VideoProcessing 23 | { 24 | int32_t ColorEnhancement(I420VideoFrame* frame); 25 | } 26 | 27 | } //namespace 28 | 29 | #endif // VPM_COLOR_ENHANCEMENT_H 30 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/primitives/fake_stdin.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 "video_engine/test/auto_test/primitives/fake_stdin.h" 12 | 13 | namespace webrtc { 14 | 15 | FILE* FakeStdin(const std::string& input) { 16 | FILE* fake_stdin = tmpfile(); 17 | 18 | EXPECT_EQ(input.size(), 19 | fwrite(input.c_str(), sizeof(char), input.size(), fake_stdin)); 20 | rewind(fake_stdin); 21 | 22 | return fake_stdin; 23 | } 24 | 25 | } // namespace webrtc 26 | -------------------------------------------------------------------------------- /webrtc/common_audio/signal_processing/spl_version.c: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | 12 | /* 13 | * This file contains the function WebRtcSpl_get_version(). 14 | * The description header can be found in signal_processing_library.h 15 | * 16 | */ 17 | 18 | #include 19 | #include "signal_processing_library.h" 20 | 21 | int16_t WebRtcSpl_get_version(char* version, int16_t length_in_bytes) 22 | { 23 | strncpy(version, "1.2.0", length_in_bytes); 24 | return 0; 25 | } 26 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.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': 'PCM16B', 13 | 'type': 'static_library', 14 | 'include_dirs': [ 15 | 'include', 16 | ], 17 | 'direct_dependent_settings': { 18 | 'include_dirs': [ 19 | 'include', 20 | ], 21 | }, 22 | 'sources': [ 23 | 'include/pcm16b.h', 24 | 'pcm16b.c', 25 | ], 26 | }, 27 | ], # targets 28 | } 29 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.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 | #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_CNG_HELPFUNS_H_ 11 | #define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_CNG_HELPFUNS_H_ 12 | 13 | #include "typedefs.h" 14 | 15 | #ifdef __cplusplus 16 | extern "C" { 17 | #endif 18 | 19 | void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a); 20 | 21 | #ifdef __cplusplus 22 | } 23 | #endif 24 | 25 | #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_CNG_HELPFUNS_H_ 26 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/Android.mk: -------------------------------------------------------------------------------- 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 | LOCAL_PATH:= $(call my-dir) 10 | 11 | include $(CLEAR_VARS) 12 | 13 | LOCAL_MODULE_TAGS := tests 14 | 15 | LOCAL_SRC_FILES := \ 16 | src/org/webrtc/voiceengine/test/AndroidTest.java 17 | 18 | LOCAL_PACKAGE_NAME := webrtc-voice-demo 19 | LOCAL_CERTIFICATE := platform 20 | 21 | LOCAL_JNI_SHARED_LIBRARIES := libwebrtc-voice-demo-jni 22 | 23 | include $(BUILD_PACKAGE) 24 | 25 | include $(call all-makefiles-under,$(LOCAL_PATH)) 26 | -------------------------------------------------------------------------------- /tools/matlab/maxUnwrap.m: -------------------------------------------------------------------------------- 1 | function sequence = maxUnwrap(sequence, max) 2 | % 3 | % sequence = maxUnwrap(sequence, max) 4 | % Unwraps when a wrap around is detected. 5 | % 6 | % Arguments 7 | % 8 | % sequence: The vector to unwrap. 9 | % max: The maximum value that the sequence can take, 10 | % and after which it will wrap over to 0. 11 | % 12 | % Return value 13 | % 14 | % sequence: The unwrapped vector. 15 | % 16 | 17 | % Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 18 | % 19 | % Use of this source code is governed by a BSD-style license 20 | % that can be found in the LICENSE file in the root of the source 21 | % tree. An additional intellectual property rights grant can be found 22 | % in the file PATENTS. All contributing project authors may 23 | % be found in the AUTHORS file in the root of the source tree. 24 | 25 | sequence = round((unwrap(2 * pi * sequence / max) * max) / (2 * pi)); 26 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/interface/cpu_info.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_ 12 | #define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_ 13 | 14 | #include "typedefs.h" 15 | 16 | namespace webrtc { 17 | 18 | class CpuInfo { 19 | public: 20 | static uint32_t DetectNumberOfCores(); 21 | 22 | private: 23 | CpuInfo() {} 24 | static uint32_t number_of_cores_; 25 | }; 26 | 27 | } // namespace webrtc 28 | 29 | #endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_ 30 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm: -------------------------------------------------------------------------------- 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 "vie_window_manager_factory.h" 12 | 13 | #include "engine_configurations.h" 14 | #if defined(COCOA_RENDERING) 15 | #include "vie_autotest_mac_cocoa.h" 16 | #elif defined(CARBON_RENDERING) 17 | #include "vie_autotest_mac_carbon.h" 18 | #endif 19 | 20 | ViEAutoTestWindowManagerInterface* 21 | ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() { 22 | return new ViEAutoTestWindowManager(); 23 | } 24 | -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/debug.proto: -------------------------------------------------------------------------------- 1 | syntax = "proto2"; 2 | option optimize_for = LITE_RUNTIME; 3 | package webrtc.audioproc; 4 | 5 | message Init { 6 | optional int32 sample_rate = 1; 7 | optional int32 device_sample_rate = 2; 8 | optional int32 num_input_channels = 3; 9 | optional int32 num_output_channels = 4; 10 | optional int32 num_reverse_channels = 5; 11 | } 12 | 13 | message ReverseStream { 14 | optional bytes data = 1; 15 | } 16 | 17 | message Stream { 18 | optional bytes input_data = 1; 19 | optional bytes output_data = 2; 20 | optional int32 delay = 3; 21 | optional sint32 drift = 4; 22 | optional int32 level = 5; 23 | } 24 | 25 | message Event { 26 | enum Type { 27 | INIT = 0; 28 | REVERSE_STREAM = 1; 29 | STREAM = 2; 30 | } 31 | 32 | required Type type = 1; 33 | 34 | optional Init init = 2; 35 | optional ReverseStream reverse_stream = 3; 36 | optional Stream stream = 4; 37 | } 38 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/gen/org/webrtc/voiceengine/test/R.java: -------------------------------------------------------------------------------- 1 | /* AUTO-GENERATED FILE. DO NOT MODIFY. 2 | * 3 | * This class was automatically generated by the 4 | * aapt tool from the resource data it found. It 5 | * should not be modified by hand. 6 | */ 7 | 8 | package org.webrtc.voiceengine.test; 9 | 10 | public final class R { 11 | public static final class attr { 12 | } 13 | public static final class drawable { 14 | public static final int icon=0x7f020000; 15 | } 16 | public static final class id { 17 | public static final int Button01=0x7f050000; 18 | } 19 | public static final class layout { 20 | public static final int main=0x7f030000; 21 | } 22 | public static final class string { 23 | public static final int app_name=0x7f040000; 24 | public static final int run_button=0x7f040001; 25 | } 26 | } 27 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/voe_cpu_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 WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H 12 | #define WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H 13 | 14 | #include "voe_standard_test.h" 15 | 16 | namespace voetest { 17 | 18 | class VoETestManager; 19 | 20 | class VoECpuTest { 21 | public: 22 | VoECpuTest(VoETestManager& mgr); 23 | ~VoECpuTest() {} 24 | int DoTest(); 25 | private: 26 | VoETestManager& _mgr; 27 | }; 28 | 29 | } // namespace voetest 30 | 31 | #endif // WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H 32 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/main/test/subfigure.m: -------------------------------------------------------------------------------- 1 | function H = subfigure(m, n, p) 2 | % 3 | % H = SUBFIGURE(m, n, p) 4 | % 5 | % Create a new figure window and adjust position and size such that it will 6 | % become the p-th tile in an m-by-n matrix of windows. (The interpretation of 7 | % m, n, and p is the same as for SUBPLOT. 8 | % 9 | % Henrik Lundin, 2009-01-19 10 | % 11 | 12 | 13 | h = figure; 14 | 15 | [j, i] = ind2sub([n m], p); 16 | scrsz = get(0,'ScreenSize'); % get screen size 17 | %scrsz = [1, 1, 1600, 1200]; 18 | 19 | taskbarSize = 58; 20 | windowbarSize = 68; 21 | windowBorder = 4; 22 | 23 | scrsz(2) = scrsz(2) + taskbarSize; 24 | scrsz(4) = scrsz(4) - taskbarSize; 25 | 26 | set(h, 'position', [(j-1)/n * scrsz(3) + scrsz(1) + windowBorder,... 27 | (m-i)/m * scrsz(4) + scrsz(2) + windowBorder, ... 28 | scrsz(3)/n - (windowBorder + windowBorder),... 29 | scrsz(4)/m - (windowbarSize + windowBorder + windowBorder)]); 30 | 31 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/android/res/values/strings.xml: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | ViEAutotest 5 | ViEAutotest 6 | Run Test 7 | Test type... 8 | 9 | Standard 10 | API 11 | Extended 12 | Loopback 13 | Custom 14 | 15 | Run... 16 | 17 | All 18 | Base 19 | Capture 20 | Codec 21 | Mix 22 | Encryption 23 | External Codec 24 | File 25 | Image Process 26 | Network 27 | Render 28 | RTP/RTCP 29 | 30 | 31 | 32 | -------------------------------------------------------------------------------- /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 | 'conditions': [ 22 | ['OS=="android"', { 23 | 'includes': [ 24 | 'test/android/android_video_demo.gypi', 25 | ], 26 | }], 27 | ], 28 | }], 29 | ], 30 | } 31 | 32 | -------------------------------------------------------------------------------- /webrtc/voice_engine/include/mock/mock_voe_connection_observer.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 MOCK_VOE_CONNECTION_OBSERVER_H_ 12 | #define MOCK_VOE_CONNECTION_OBSERVER_H_ 13 | 14 | #include "voice_engine/include/voe_network.h" 15 | 16 | namespace webrtc { 17 | 18 | class MockVoeConnectionObserver : public VoEConnectionObserver { 19 | public: 20 | MOCK_METHOD2(OnPeriodicDeadOrAlive, void(const int channel, 21 | const bool alive)); 22 | }; 23 | 24 | } 25 | 26 | #endif // MOCK_VOE_CONNECTION_OBSERVER_H_ 27 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/main/test/pcap_file_reader.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_VIDEO_CODING_TEST_PCAP_FILE_READER_H_ 12 | #define WEBRTC_MODULES_VIDEO_CODING_TEST_PCAP_FILE_READER_H_ 13 | 14 | #include 15 | 16 | namespace webrtc { 17 | namespace rtpplayer { 18 | 19 | class RtpPacketSourceInterface; 20 | 21 | RtpPacketSourceInterface* CreatePcapFileReader(const std::string& filename); 22 | 23 | } // namespace rtpplayer 24 | } // namespace webrtc 25 | 26 | #endif // WEBRTC_MODULES_VIDEO_CODING_TEST_PCAP_FILE_READER_H_ 27 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/interface/compile_assert.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 WEBRTC_SYSTEM_WRAPPERS_INTERFACE_COMPILE_ASSERT_H_ 12 | #define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_COMPILE_ASSERT_H_ 13 | 14 | /* Use this macro to verify at compile time that certain restrictions are met. 15 | * The argument is the boolean expression to evaluate. 16 | * Example: 17 | * COMPILE_ASSERT(sizeof(foo) < 128); 18 | */ 19 | #define COMPILE_ASSERT(expression) switch(0){case 0: case expression:;} 20 | 21 | #endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_COMPILE_ASSERT_H_ 22 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/critical_section.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 | #if defined(_WIN32) 12 | #include 13 | #include "webrtc/system_wrappers/source/critical_section_win.h" 14 | #else 15 | #include "webrtc/system_wrappers/source/critical_section_posix.h" 16 | #endif 17 | 18 | namespace webrtc { 19 | 20 | CriticalSectionWrapper* CriticalSectionWrapper::CreateCriticalSection() { 21 | #ifdef _WIN32 22 | return new CriticalSectionWindows(); 23 | #else 24 | return new CriticalSectionPosix(); 25 | #endif 26 | } 27 | 28 | } // namespace webrtc 29 | -------------------------------------------------------------------------------- /webrtc/build/arm_neon.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 | # This file sets correct neon flags. Include it if you want to build 10 | # source with neon intrinsics. 11 | # To use this, create a gyp target with the following form: 12 | # { 13 | # 'target_name': 'my_lib', 14 | # 'type': 'static_library', 15 | # 'sources': [ 16 | # 'foo.c', 17 | # 'bar.cc', 18 | # ], 19 | # 'includes': ['path/to/this/gypi/file'], 20 | # } 21 | 22 | { 23 | 'cflags!': [ 24 | '-mfpu=vfpv3-d16', 25 | ], 26 | 'cflags': [ 27 | '-mfpu=neon', 28 | '-flax-vector-conversions', 29 | ], 30 | } 31 | -------------------------------------------------------------------------------- /webrtc/modules/desktop_capture/shared_memory.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "webrtc/modules/desktop_capture/shared_memory.h" 12 | 13 | namespace webrtc { 14 | 15 | #if defined(WEBRTC_WIN) 16 | const SharedMemory::Handle SharedMemory::kInvalidHandle = NULL; 17 | #else 18 | const SharedMemory::Handle SharedMemory::kInvalidHandle = -1; 19 | #endif 20 | 21 | SharedMemory::SharedMemory(void* data, size_t size, Handle handle, int id) 22 | : data_(data), 23 | size_(size), 24 | handle_(handle), 25 | id_(id) { 26 | } 27 | 28 | } // namespace webrtc 29 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/interface/sleep.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | // An OS-independent sleep function. 11 | 12 | #ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SLEEP_H_ 13 | #define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SLEEP_H_ 14 | 15 | namespace webrtc { 16 | 17 | // This function sleeps for the specified number of milliseconds. 18 | // It may return early if the thread is woken by some other event, 19 | // such as the delivery of a signal on Unix. 20 | void SleepMs(int msecs); 21 | 22 | } // namespace webrtc 23 | 24 | #endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SLEEP_H_ 25 | -------------------------------------------------------------------------------- /webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.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_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ 11 | #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ 12 | 13 | #include "webrtc/typedefs.h" 14 | 15 | namespace webrtc { 16 | namespace RtpFormatVideoGeneric { 17 | static const uint8_t kKeyFrameBit = 0x01; 18 | static const uint8_t kFirstPacketBit = 0x02; 19 | } // namespace RtpFormatVideoGeneric 20 | } // namespace webrtc 21 | 22 | #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ 23 | -------------------------------------------------------------------------------- /webrtc/modules/video_capture/external/video_capture_external.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "../video_capture_impl.h" 12 | #include "ref_count.h" 13 | 14 | namespace webrtc { 15 | 16 | namespace videocapturemodule { 17 | 18 | VideoCaptureModule* VideoCaptureImpl::Create( 19 | const int32_t id, 20 | const char* deviceUniqueIdUTF8) { 21 | RefCountImpl* implementation = 22 | new RefCountImpl(id); 23 | return implementation; 24 | } 25 | 26 | } // namespace videocapturemodule 27 | 28 | } // namespace webrtc 29 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/main/test/rtp_file_reader.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_VIDEO_CODING_MAIN_TEST_RTP_FILE_READER_H_ 12 | #define WEBRTC_MODULES_VIDEO_CODING_MAIN_TEST_RTP_FILE_READER_H_ 13 | 14 | #include 15 | 16 | namespace webrtc { 17 | namespace rtpplayer { 18 | 19 | class RtpPacketSourceInterface; 20 | 21 | RtpPacketSourceInterface* CreateRtpFileReader(const std::string& filename); 22 | 23 | } // namespace rtpplayer 24 | } // namespace webrtc 25 | 26 | #endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_TEST_RTP_FILE_READER_H_ 27 | -------------------------------------------------------------------------------- /webrtc/modules/video_processing/main/source/brighten.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_ 12 | #define MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_ 13 | 14 | #include "typedefs.h" 15 | #include "modules/video_processing/main/interface/video_processing.h" 16 | 17 | namespace webrtc { 18 | namespace VideoProcessing { 19 | 20 | int32_t Brighten(I420VideoFrame* frame, int delta); 21 | 22 | } // namespace VideoProcessing 23 | } // namespace webrtc 24 | 25 | #endif // MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_ 26 | -------------------------------------------------------------------------------- /webrtc/build/adb_shell.sh: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | 3 | # Copyright (c) 2012 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 | # 'adb shell' always returns "0" regardless of executable return code. 12 | # This handy script will return executable return code to shell which 13 | # can be used by buildbots. 14 | 15 | adb_shell () { 16 | local RET ADB_LOG 17 | ADB_LOG=$(mktemp "${TMPDIR:-/tmp}/adb-XXXXXXXX") 18 | adb "$1" "$2" shell "$3" "$4" ";" echo \$? | tee "$ADB_LOG" 19 | sed -i -e 's![[:cntrl:]]!!g' "$ADB_LOG" # Remove \r. 20 | RET=$(sed -e '$!d' "$ADB_LOG") # Last line contains status code. 21 | rm -f "$ADB_LOG" 22 | return $RET 23 | } 24 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/utility/video_coding_utility.gyp: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2013 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 | ], 13 | 'targets': [ 14 | { 15 | 'target_name': 'video_coding_utility', 16 | 'type': 'static_library', 17 | 'dependencies': [ 18 | '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', 19 | ], 20 | 'sources': [ 21 | 'include/exp_filter.h', 22 | 'include/frame_dropper.h', 23 | 'exp_filter.cc', 24 | 'frame_dropper.cc', 25 | ], 26 | }, 27 | ], # targets 28 | } 29 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/main/test/TimedTrace.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 TIMED_TRACE_H 12 | #define TIMED_TRACE_H 13 | 14 | #include "typedefs.h" 15 | 16 | #include 17 | #include 18 | 19 | class TimedTrace { 20 | public: 21 | TimedTrace(); 22 | ~TimedTrace(); 23 | 24 | void SetTimeEllapsed(double myTime); 25 | double TimeEllapsed(); 26 | void Tick10Msec(); 27 | int16_t SetUp(char* fileName); 28 | void TimedLogg(char* message); 29 | 30 | private: 31 | static double _timeEllapsedSec; 32 | static FILE* _timedTraceFile; 33 | 34 | }; 35 | 36 | #endif 37 | -------------------------------------------------------------------------------- /webrtc/voice_engine/include/mock/mock_voe_observer.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 WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_ 12 | #define WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_ 13 | 14 | #include "gmock/gmock.h" 15 | #include "voice_engine/include/voe_base.h" 16 | 17 | namespace webrtc { 18 | 19 | class MockVoEObserver: public VoiceEngineObserver { 20 | public: 21 | MockVoEObserver() {} 22 | virtual ~MockVoEObserver() {} 23 | 24 | MOCK_METHOD2(CallbackOnError, void(const int channel, const int error_code)); 25 | }; 26 | 27 | } 28 | 29 | #endif // WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_ 30 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/critical_section_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 | 11 | #include "webrtc/system_wrappers/source/critical_section_win.h" 12 | 13 | namespace webrtc { 14 | 15 | CriticalSectionWindows::CriticalSectionWindows() { 16 | InitializeCriticalSection(&crit); 17 | } 18 | 19 | CriticalSectionWindows::~CriticalSectionWindows() { 20 | DeleteCriticalSection(&crit); 21 | } 22 | 23 | void 24 | CriticalSectionWindows::Enter() { 25 | EnterCriticalSection(&crit); 26 | } 27 | 28 | void 29 | CriticalSectionWindows::Leave() { 30 | LeaveCriticalSection(&crit); 31 | } 32 | 33 | } // namespace webrtc 34 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/win_test/WinTest.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 | #pragma once 12 | 13 | #ifndef __AFXWIN_H__ 14 | #error "include 'stdafx.h' before including this file for PCH" 15 | #endif 16 | 17 | #include "resource.h" // main symbols 18 | 19 | 20 | // CWinTestApp: 21 | // See WinTest.cpp for the implementation of this class 22 | // 23 | 24 | class CWinTestApp : public CWinApp 25 | { 26 | public: 27 | CWinTestApp(); 28 | 29 | // Overrides 30 | public: 31 | virtual BOOL InitInstance(); 32 | 33 | // Implementation 34 | 35 | DECLARE_MESSAGE_MAP() 36 | }; 37 | 38 | extern CWinTestApp theApp; 39 | -------------------------------------------------------------------------------- /webrtc/modules/rtp_rtcp/source/video_codec_information.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ 12 | #define WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ 13 | 14 | #include "rtp_rtcp_config.h" 15 | #include "rtp_utility.h" 16 | 17 | namespace webrtc { 18 | class VideoCodecInformation 19 | { 20 | public: 21 | virtual void Reset() = 0; 22 | 23 | virtual RtpVideoCodecTypes Type() = 0; 24 | virtual ~VideoCodecInformation(){}; 25 | }; 26 | } // namespace webrtc 27 | 28 | #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_ 29 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/android/.project: -------------------------------------------------------------------------------- 1 | 2 | 3 | ViEAutotest 4 | 5 | 6 | 7 | 8 | 9 | com.android.ide.eclipse.adt.ResourceManagerBuilder 10 | 11 | 12 | 13 | 14 | com.android.ide.eclipse.adt.PreCompilerBuilder 15 | 16 | 17 | 18 | 19 | org.eclipse.jdt.core.javabuilder 20 | 21 | 22 | 23 | 24 | com.android.ide.eclipse.adt.ApkBuilder 25 | 26 | 27 | 28 | 29 | 30 | com.android.ide.eclipse.adt.AndroidNature 31 | org.eclipse.jdt.core.javanature 32 | 33 | 34 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/automated/legacy_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/video_engine/test/auto_test/automated/legacy_fixture.h" 12 | 13 | #include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h" 14 | 15 | void LegacyFixture::SetUpTestCase() { 16 | TwoWindowsFixture::SetUpTestCase(); 17 | 18 | // Create the test cases 19 | tests_ = new ViEAutoTest(window_1_, window_2_); 20 | } 21 | 22 | void LegacyFixture::TearDownTestCase() { 23 | delete tests_; 24 | 25 | TwoWindowsFixture::TearDownTestCase(); 26 | } 27 | 28 | ViEAutoTest* LegacyFixture::tests_ = NULL; 29 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/.project: -------------------------------------------------------------------------------- 1 | 2 | 3 | AndroidTest 4 | 5 | 6 | 7 | 8 | 9 | com.android.ide.eclipse.adt.ResourceManagerBuilder 10 | 11 | 12 | 13 | 14 | com.android.ide.eclipse.adt.PreCompilerBuilder 15 | 16 | 17 | 18 | 19 | org.eclipse.jdt.core.javabuilder 20 | 21 | 22 | 23 | 24 | com.android.ide.eclipse.adt.ApkBuilder 25 | 26 | 27 | 28 | 29 | 30 | com.android.ide.eclipse.adt.AndroidNature 31 | org.eclipse.jdt.core.javanature 32 | 33 | 34 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/AndroidManifest.xml: -------------------------------------------------------------------------------- 1 | 2 | 5 | 8 | 10 | 11 | 12 | 13 | 14 | 15 | 16 | 17 | 18 | 19 | 20 | 21 | 22 | 23 | -------------------------------------------------------------------------------- /tools/e2e_quality/e2e_quality.gyp: -------------------------------------------------------------------------------- 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': ['../../webrtc/build/common.gypi'], 11 | 'targets': [ 12 | { 13 | 'target_name': 'audio_e2e_harness', 14 | 'type': 'executable', 15 | 'dependencies': [ 16 | '<(webrtc_root)/test/channel_transport.gyp:channel_transport', 17 | '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core', 18 | '<(DEPTH)/testing/gtest.gyp:gtest', 19 | '<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags', 20 | ], 21 | 'sources': [ 22 | 'audio/audio_e2e_harness.cc', 23 | ], 24 | }, 25 | ], 26 | } 27 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/clock_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 | #include "webrtc/system_wrappers/interface/clock.h" 12 | 13 | #include "testing/gtest/include/gtest/gtest.h" 14 | 15 | namespace webrtc { 16 | 17 | TEST(ClockTest, NtpTime) { 18 | Clock* clock = Clock::GetRealTimeClock(); 19 | uint32_t seconds; 20 | uint32_t fractions; 21 | clock->CurrentNtp(seconds, fractions); 22 | int64_t milliseconds = clock->CurrentNtpInMilliseconds(); 23 | EXPECT_GE(milliseconds, Clock::NtpToMs(seconds, fractions)); 24 | EXPECT_NEAR(milliseconds, Clock::NtpToMs(seconds, fractions), 5); 25 | } 26 | } // namespace webrtc 27 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/tick_util.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/system_wrappers/interface/tick_util.h" 12 | 13 | #include 14 | 15 | namespace webrtc { 16 | 17 | bool TickTime::use_fake_clock_ = false; 18 | int64_t TickTime::fake_ticks_ = 0; 19 | 20 | void TickTime::UseFakeClock(int64_t start_millisecond) { 21 | use_fake_clock_ = true; 22 | fake_ticks_ = MillisecondsToTicks(start_millisecond); 23 | } 24 | 25 | void TickTime::AdvanceFakeClock(int64_t milliseconds) { 26 | assert(use_fake_clock_); 27 | fake_ticks_ += MillisecondsToTicks(milliseconds); 28 | } 29 | 30 | } // namespace webrtc 31 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/interface/vie_autotest_android.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_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_ 12 | #define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_ 13 | 14 | class ViEAutoTestAndroid 15 | { 16 | public: 17 | static int RunAutotest(int testSelection, 18 | int subTestSelection, 19 | void* window1, 20 | void* window2, 21 | void* javaVM, 22 | void* env, 23 | void* context); 24 | }; 25 | 26 | #endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_ 27 | -------------------------------------------------------------------------------- /webrtc/common_audio/signal_processing/webrtc_fft_t_rad.c: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | 12 | /* 13 | * This file contains the Q14 radix-2 tables used in ARM9E optimization routines. 14 | * 15 | */ 16 | 17 | extern const unsigned short t_Q14S_rad8[2]; 18 | const unsigned short t_Q14S_rad8[2] = { 0x0000,0x2d41 }; 19 | 20 | //extern const int t_Q30S_rad8[2]; 21 | //const int t_Q30S_rad8[2] = { 0x00000000,0x2d413ccd }; 22 | 23 | extern const unsigned short t_Q14R_rad8[2]; 24 | const unsigned short t_Q14R_rad8[2] = { 0x2d41,0x2d41 }; 25 | 26 | //extern const int t_Q30R_rad8[2]; 27 | //const int t_Q30R_rad8[2] = { 0x2d413ccd,0x2d413ccd }; 28 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/jni/org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h: -------------------------------------------------------------------------------- 1 | /* DO NOT EDIT THIS FILE - it is machine generated */ 2 | #include 3 | /* Header for class org_webrtc_voiceengine_test_AudioDeviceAndroidTest */ 4 | 5 | #ifndef _Included_org_webrtc_voiceengine_test_AudioDeviceAndroidTest 6 | #define _Included_org_webrtc_voiceengine_test_AudioDeviceAndroidTest 7 | #ifdef __cplusplus 8 | extern "C" { 9 | #endif 10 | /* 11 | * Class: org_webrtc_voiceengine_test_AudioDeviceAndroidTest 12 | * Method: NativeInit 13 | * Signature: ()Z 14 | */ 15 | JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_NativeInit 16 | (JNIEnv *, jclass); 17 | 18 | /* 19 | * Class: org_webrtc_voiceengine_test_AudioDeviceAndroidTest 20 | * Method: RunTest 21 | * Signature: (I)I 22 | */ 23 | JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_RunTest 24 | (JNIEnv *, jobject, jint); 25 | 26 | #ifdef __cplusplus 27 | } 28 | #endif 29 | #endif 30 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/main/test/release_test_pt2.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 "ReleaseTest.h" 12 | #include "ReceiverTests.h" 13 | #include "TestMacros.h" 14 | #include "MediaOptTest.h" 15 | #include "CodecDataBaseTest.h" 16 | #include "GenericCodecTest.h" 17 | 18 | 19 | 20 | 21 | int ReleaseTestPart2() 22 | { 23 | printf("Verify that TICK_TIME_DEBUG and EVENT_DEBUG are uncommented"); 24 | // Tests requiring verification 25 | 26 | printf("Testing Generic Codecs...\n"); 27 | TEST(VCMGenericCodecTest() == 0); 28 | printf("Verify by viewing output file GCTest_out.yuv \n"); 29 | 30 | return 0; 31 | } -------------------------------------------------------------------------------- /webrtc/modules/audio_processing/test/android/apmtest/jni/Android.mk: -------------------------------------------------------------------------------- 1 | # Copyright (C) 2010 The Android Open Source Project 2 | # 3 | # Licensed under the Apache License, Version 2.0 (the "License"); 4 | # you may not use this file except in compliance with the License. 5 | # You may obtain a copy of the License at 6 | # 7 | # http://www.apache.org/licenses/LICENSE-2.0 8 | # 9 | # Unless required by applicable law or agreed to in writing, software 10 | # distributed under the License is distributed on an "AS IS" BASIS, 11 | # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 12 | # See the License for the specific language governing permissions and 13 | # limitations under the License. 14 | # 15 | LOCAL_PATH := $(call my-dir) 16 | 17 | include $(CLEAR_VARS) 18 | 19 | LOCAL_MODULE := apmtest-activity 20 | LOCAL_SRC_FILES := main.c 21 | LOCAL_LDLIBS := -llog -landroid -lEGL -lGLESv1_CM 22 | LOCAL_STATIC_LIBRARIES := android_native_app_glue 23 | 24 | include $(BUILD_SHARED_LIBRARY) 25 | 26 | $(call import-module,android/native_app_glue) 27 | -------------------------------------------------------------------------------- /tools/refactoring/p4commands.py: -------------------------------------------------------------------------------- 1 | import os 2 | import filemanagement 3 | 4 | # checks out entire p4 repository 5 | def checkoutallfiles(): 6 | os.system('p4 edit //depotGoogle/...') 7 | return 8 | 9 | # reverts all unchanged files, this is completely innoculus 10 | def revertunchangedfiles(): 11 | os.system('p4 revert -a //depotGoogle/...') 12 | return 13 | 14 | def integratefile( old_name, new_name): 15 | if(old_name == new_name): 16 | return 17 | if(not filemanagement.fileexist(old_name)): 18 | return 19 | integrate_command = 'p4 integrate -o -f ' +\ 20 | old_name +\ 21 | ' ' +\ 22 | new_name +\ 23 | ' > p4summary.txt 2> error.txt' 24 | os.system(integrate_command) 25 | #print integrate_command 26 | delete_command = 'p4 delete -c default ' +\ 27 | old_name +\ 28 | ' > p4summary.txt 2> error.txt' 29 | os.system(delete_command) 30 | #print delete_command 31 | return 32 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/android/audio_device_android_test/.project: -------------------------------------------------------------------------------- 1 | 2 | 3 | AudioDeviceAndroidTest 4 | 5 | 6 | 7 | 8 | 9 | com.android.ide.eclipse.adt.ResourceManagerBuilder 10 | 11 | 12 | 13 | 14 | com.android.ide.eclipse.adt.PreCompilerBuilder 15 | 16 | 17 | 18 | 19 | org.eclipse.jdt.core.javabuilder 20 | 21 | 22 | 23 | 24 | com.android.ide.eclipse.adt.ApkBuilder 25 | 26 | 27 | 28 | 29 | 30 | com.android.ide.eclipse.adt.AndroidNature 31 | org.eclipse.jdt.core.javanature 32 | 33 | 34 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/cng/cng.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': 'CNG', 13 | 'type': 'static_library', 14 | 'dependencies': [ 15 | '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', 16 | ], 17 | 'include_dirs': [ 18 | 'include', 19 | ], 20 | 'direct_dependent_settings': { 21 | 'include_dirs': [ 22 | 'include', 23 | ], 24 | }, 25 | 'sources': [ 26 | 'include/webrtc_cng.h', 27 | 'webrtc_cng.c', 28 | 'cng_helpfuns.c', 29 | 'cng_helpfuns.h', 30 | ], 31 | }, 32 | ], # targets 33 | } 34 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/sleep.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | // An OS-independent sleep function. 11 | 12 | #include "system_wrappers/interface/sleep.h" 13 | 14 | #ifdef _WIN32 15 | // For Sleep() 16 | #include 17 | #else 18 | // For nanosleep() 19 | #include 20 | #endif 21 | 22 | namespace webrtc { 23 | 24 | void SleepMs(int msecs) { 25 | #ifdef _WIN32 26 | Sleep(msecs); 27 | #else 28 | struct timespec short_wait; 29 | struct timespec remainder; 30 | short_wait.tv_sec = msecs / 1000; 31 | short_wait.tv_nsec = (msecs % 1000) * 1000 * 1000; 32 | nanosleep(&short_wait, &remainder); 33 | #endif 34 | } 35 | 36 | } // namespace webrtc 37 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/resource_manager.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 "resource_manager.h" 12 | 13 | #include "testsupport/fileutils.h" 14 | 15 | ResourceManager::ResourceManager() { 16 | std::string filename = "audio_long16.pcm"; 17 | #if defined(WEBRTC_ANDROID) 18 | long_audio_file_path_ = "/sdcard/" + filename; 19 | #else 20 | std::string resource_path = webrtc::test::ProjectRootPath(); 21 | if (resource_path == webrtc::test::kCannotFindProjectRootDir) { 22 | long_audio_file_path_ = ""; 23 | } else { 24 | long_audio_file_path_ = 25 | resource_path + "data/voice_engine/" + filename; 26 | } 27 | #endif 28 | } 29 | 30 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/resource_manager.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_RESOURCE_MANAGER_H_ 12 | #define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_ 13 | 14 | #include 15 | 16 | class ResourceManager { 17 | public: 18 | ResourceManager(); 19 | 20 | // Returns the full path to a long audio file. 21 | // Returns the empty string on failure. 22 | const std::string& long_audio_file_path() const { 23 | return long_audio_file_path_; 24 | } 25 | 26 | private: 27 | std::string long_audio_file_path_; 28 | }; 29 | 30 | #endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_ 31 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /****************************************************************** 12 | 13 | iLBC Speech Coder ANSI-C Source Code 14 | 15 | WebRtcIlbcfix_IndexConvDec.h 16 | 17 | ******************************************************************/ 18 | 19 | #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_ 20 | #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_ 21 | 22 | #include "defines.h" 23 | 24 | void WebRtcIlbcfix_IndexConvDec( 25 | int16_t *index /* (i/o) Codebook indexes */ 26 | ); 27 | 28 | #endif 29 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/opus/opus.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 | 'targets': [ 11 | { 12 | 'target_name': 'webrtc_opus', 13 | 'type': 'static_library', 14 | 'conditions': [ 15 | ['build_with_mozilla==1', { 16 | # Mozilla provides its own build of the opus library. 17 | 'include_dirs': [ 18 | '$(DIST)/include/opus', 19 | ] 20 | }, { 21 | 'dependencies': [ 22 | '<(DEPTH)/third_party/opus/opus.gyp:opus' 23 | ], 24 | }], 25 | ], 26 | 'sources': [ 27 | 'interface/opus_interface.h', 28 | 'opus_interface.c', 29 | ], 30 | }, 31 | ], 32 | } 33 | -------------------------------------------------------------------------------- /webrtc/modules/video_render/mac/cocoa_full_screen_window.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // 12 | // cocoa_full_screen_window.h 13 | // 14 | // 15 | 16 | #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_ 17 | #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_ 18 | 19 | #import 20 | //#define GRAB_ALL_SCREENS 1 21 | 22 | @interface CocoaFullScreenWindow : NSObject { 23 | NSWindow* _window; 24 | } 25 | 26 | -(id)init; 27 | -(void)grabFullScreen; 28 | -(void)releaseFullScreen; 29 | -(NSWindow*)window; 30 | 31 | @end 32 | 33 | #endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_ 34 | -------------------------------------------------------------------------------- /tools/valgrind-webrtc/tsan/suppressions.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 must exist for the Python wrapper script to work properly. 11 | 12 | { 13 | bug_300 14 | ThreadSanitizer:Race 15 | fun:webrtc::TraceImpl::SetTraceCallbackImpl 16 | fun:webrtc::Trace::SetTraceCallback 17 | ... 18 | } 19 | 20 | # Known bugs we don't care about / problems in third parties 21 | { 22 | bug_884 (Wider timezone filter than Chromium's suppressions) 23 | ThreadSanitizer:Race 24 | ... 25 | fun:__tz* 26 | } 27 | { 28 | don't care about issues in trace functions 29 | ThreadSanitizer:Race 30 | ... 31 | fun:webrtc::Trace::Add 32 | ... 33 | } 34 | -------------------------------------------------------------------------------- /tools/quality_tracking/dashboard/lkgr_page.py: -------------------------------------------------------------------------------- 1 | #!/usr/bin/env python 2 | #-*- coding: utf-8 -*- 3 | # Copyright (c) 2012 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 | """Implements the LKGR page.""" 12 | 13 | import webapp2 14 | 15 | import load_build_status 16 | 17 | class ShowLkgr(webapp2.RequestHandler): 18 | """This handler shows the LKGR in the simplest possible way. 19 | 20 | The page is intended to be used by automated tools. 21 | """ 22 | def get(self): 23 | build_status_loader = load_build_status.BuildStatusLoader() 24 | 25 | lkgr = build_status_loader.compute_lkgr() 26 | if lkgr is None: 27 | self.response.out.write('No data has been uploaded to the dashboard.') 28 | else: 29 | self.response.out.write(lkgr) 30 | -------------------------------------------------------------------------------- /webrtc/modules/video_capture/mac/qtkit/video_capture_recursive_lock.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // 12 | // video_capture_recursive_lock.h 13 | // 14 | // 15 | 16 | #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_ 17 | #define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_ 18 | 19 | #import 20 | 21 | @interface VideoCaptureRecursiveLock : NSRecursiveLock { 22 | BOOL _locked; 23 | } 24 | 25 | @property BOOL locked; 26 | 27 | - (void)lock; 28 | - (void)unlock; 29 | 30 | @end 31 | 32 | #endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_ 33 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/android/res/layout/tabhost.xml: -------------------------------------------------------------------------------- 1 | 2 | 6 | 10 | 14 | 18 | 19 | 20 | 21 | 22 | 23 | 24 | 25 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq/mute_signal.c: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /* 12 | * This function mutes a signal linearly on a sample by sample basis. 13 | */ 14 | 15 | #include "dsp_helpfunctions.h" 16 | 17 | #include "signal_processing_library.h" 18 | 19 | void WebRtcNetEQ_MuteSignal(int16_t *pw16_inout, int16_t muteSlope, 20 | int16_t N) 21 | { 22 | int i; 23 | int32_t w32_tmp = 1048608; /* (16384<<6 + 32) */ 24 | 25 | for (i = 0; i < N; i++) 26 | { 27 | pw16_inout[i] 28 | = (int16_t) ((WEBRTC_SPL_MUL_16_16((int16_t)(w32_tmp>>6), pw16_inout[i]) 29 | + 8192) >> 14); 30 | w32_tmp -= muteSlope; 31 | } 32 | } 33 | 34 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/dummy/audio_device_utility_dummy.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 WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H 12 | #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H 13 | 14 | #include "audio_device_utility.h" 15 | #include "audio_device.h" 16 | 17 | namespace webrtc 18 | { 19 | class CriticalSectionWrapper; 20 | 21 | class AudioDeviceUtilityDummy: public AudioDeviceUtility 22 | { 23 | public: 24 | AudioDeviceUtilityDummy(const int32_t id) {} 25 | ~AudioDeviceUtilityDummy() {} 26 | 27 | virtual int32_t Init() { return 0; } 28 | }; 29 | 30 | } // namespace webrtc 31 | 32 | #endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_DUMMY_H_ 33 | -------------------------------------------------------------------------------- /webrtc/test/testsupport/mock/mock_frame_writer.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_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_ 12 | #define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_ 13 | 14 | #include "testsupport/frame_writer.h" 15 | 16 | #include "gmock/gmock.h" 17 | 18 | namespace webrtc { 19 | namespace test { 20 | 21 | class MockFrameWriter : public FrameWriter { 22 | public: 23 | MOCK_METHOD0(Init, bool()); 24 | MOCK_METHOD1(WriteFrame, bool(uint8_t* frame_buffer)); 25 | MOCK_METHOD0(Close, void()); 26 | MOCK_METHOD0(FrameLength, size_t()); 27 | }; 28 | 29 | } // namespace test 30 | } // namespace webrtc 31 | 32 | #endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_ 33 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/standard/voe_base_misc_test.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 "voice_engine/test/auto_test/fixtures/before_initialization_fixture.h" 12 | 13 | #include 14 | 15 | class VoeBaseMiscTest : public BeforeInitializationFixture { 16 | }; 17 | 18 | using namespace testing; 19 | 20 | TEST_F(VoeBaseMiscTest, MaxNumChannelsIs100) { 21 | EXPECT_EQ(100, voe_base_->MaxNumOfChannels()); 22 | } 23 | 24 | TEST_F(VoeBaseMiscTest, GetVersionPrintsSomeUsefulInformation) { 25 | char char_buffer[1024]; 26 | memset(char_buffer, 0, sizeof(char_buffer)); 27 | EXPECT_EQ(0, voe_base_->GetVersion(char_buffer)); 28 | EXPECT_THAT(char_buffer, ContainsRegex("VoiceEngine")); 29 | } 30 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/isac/isacfix_test.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 | 'targets': [ 11 | # kenny 12 | { 13 | 'target_name': 'iSACFixtest', 14 | 'type': 'executable', 15 | 'dependencies': [ 16 | 'iSACFix', 17 | '<(webrtc_root)/test/test.gyp:test_support', 18 | ], 19 | 'include_dirs': [ 20 | './fix/test', 21 | './fix/interface', 22 | ], 23 | 'sources': [ 24 | './fix/test/kenny.cc', 25 | ], 26 | # Disable warnings to enable Win64 build, issue 1323. 27 | 'msvs_disabled_warnings': [ 28 | 4267, # size_t to int truncation. 29 | ], 30 | }, 31 | ], 32 | } 33 | 34 | # TODO(kma): Add bit-exact test for iSAC-fix. 35 | -------------------------------------------------------------------------------- /webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.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_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_ 12 | #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_ 13 | 14 | namespace webrtc { 15 | class AudioFrame; 16 | 17 | // Updates the audioFrame's energy (based on its samples). 18 | void CalculateEnergy(AudioFrame& audioFrame); 19 | 20 | // Apply linear step function that ramps in/out the audio samples in audioFrame 21 | void RampIn(AudioFrame& audioFrame); 22 | void RampOut(AudioFrame& audioFrame); 23 | 24 | } // namespace webrtc 25 | 26 | #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_ 27 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/audio_device_utility.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H 12 | #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H 13 | 14 | #include "typedefs.h" 15 | 16 | namespace webrtc 17 | { 18 | 19 | class AudioDeviceUtility 20 | { 21 | public: 22 | static uint32_t GetTimeInMS(); 23 | static void WaitForKey(); 24 | static bool StringCompare(const char* str1, 25 | const char* str2, 26 | const uint32_t length); 27 | virtual int32_t Init() = 0; 28 | 29 | virtual ~AudioDeviceUtility() {} 30 | }; 31 | 32 | } // namespace webrtc 33 | 34 | #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H 35 | 36 | -------------------------------------------------------------------------------- /webrtc/modules/video_capture/video_capture_delay.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_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_ 12 | #define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_ 13 | 14 | namespace webrtc 15 | { 16 | namespace videocapturemodule 17 | { 18 | 19 | struct DelayValue 20 | { 21 | int32_t width; 22 | int32_t height; 23 | int32_t delay; 24 | }; 25 | 26 | enum { NoOfDelayValues = 40 }; 27 | struct DelayValues 28 | { 29 | char * deviceName; 30 | char* productId; 31 | DelayValue delayValues[NoOfDelayValues]; 32 | }; 33 | 34 | } //namespace videocapturemodule 35 | } //namespace webrtc 36 | #endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_ 37 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq4/comfort_noise_unittest.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // Unit tests for ComfortNoise class. 12 | 13 | #include "webrtc/modules/audio_coding/neteq4/comfort_noise.h" 14 | 15 | #include "gtest/gtest.h" 16 | #include "webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h" 17 | #include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" 18 | 19 | namespace webrtc { 20 | 21 | TEST(ComfortNoise, CreateAndDestroy) { 22 | int fs = 8000; 23 | MockDecoderDatabase db; 24 | SyncBuffer sync_buffer(1, 1000); 25 | ComfortNoise cn(fs, &db, &sync_buffer); 26 | EXPECT_CALL(db, Die()); // Called when |db| goes out of scope. 27 | } 28 | 29 | // TODO(hlundin): Write more tests. 30 | 31 | } // namespace webrtc 32 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/test/README.txt: -------------------------------------------------------------------------------- 1 | INSTRUCTIONS: 2 | 3 | - Start with test #3 (Device enumeration) to get an overview of the available 4 | audio devices. 5 | - Next, proceed with test #4 (Device selection) to get more details about 6 | the supported functions for each audio device. 7 | - Verify two-way audio in test #5. 8 | Repeat this test for different selections of playout and recording devices. 9 | - More detailed tests (volume, mute etc.) can also be performed using #6-#11. 10 | 11 | NOTE: 12 | 13 | - Some tests requires that the user opens up the audio mixer dialog and 14 | verifies that a certain action (e.g. Mute ON/OFF) is executed correctly. 15 | - Files can be recorded during some tests to enable off-line analysis. 16 | - Full support of 'Default Communication' devices requires Windows 7. 17 | - If a test consists of several sub tests, press any key to start a new sub test. 18 | 19 | KNOWN ISSUES: 20 | 21 | - Microphone Boost control is not supported on Windows Vista or Windows 7. 22 | - Speaker and microphone volume controls will not work as intended on Windows 23 | Vista if a 'Default Communication' device is selected in any direction. 24 | -------------------------------------------------------------------------------- /webrtc/test/testsupport/mock/mock_frame_reader.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_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_ 12 | #define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_ 13 | 14 | #include "testsupport/frame_reader.h" 15 | 16 | #include "gmock/gmock.h" 17 | 18 | namespace webrtc { 19 | namespace test { 20 | 21 | class MockFrameReader : public FrameReader { 22 | public: 23 | MOCK_METHOD0(Init, bool()); 24 | MOCK_METHOD1(ReadFrame, bool(uint8_t* source_buffer)); 25 | MOCK_METHOD0(Close, void()); 26 | MOCK_METHOD0(FrameLength, size_t()); 27 | MOCK_METHOD0(NumberOfFrames, int()); 28 | }; 29 | 30 | } // namespace test 31 | } // namespace webrtc 32 | 33 | #endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_ 34 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java: -------------------------------------------------------------------------------- 1 | /* AUTO-GENERATED FILE. DO NOT MODIFY. 2 | * 3 | * This class was automatically generated by the 4 | * aapt tool from the resource data it found. It 5 | * should not be modified by hand. 6 | */ 7 | 8 | package org.webrtc.voiceengine.test; 9 | 10 | public final class R { 11 | public static final class attr { 12 | } 13 | public static final class drawable { 14 | public static final int icon=0x7f020000; 15 | } 16 | public static final class id { 17 | public static final int Button01=0x7f050002; 18 | public static final int Button02=0x7f050005; 19 | public static final int EditText01=0x7f050001; 20 | public static final int Spinner01=0x7f050003; 21 | public static final int Spinner02=0x7f050004; 22 | public static final int TextView01=0x7f050000; 23 | } 24 | public static final class layout { 25 | public static final int main=0x7f030000; 26 | } 27 | public static final class string { 28 | public static final int app_name=0x7f040000; 29 | } 30 | } 31 | -------------------------------------------------------------------------------- /webrtc/voice_engine/voe_neteq_stats_impl.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 WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H 12 | #define WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H 13 | 14 | #include "voe_neteq_stats.h" 15 | 16 | #include "shared_data.h" 17 | 18 | namespace webrtc { 19 | 20 | class VoENetEqStatsImpl : public VoENetEqStats 21 | { 22 | public: 23 | virtual int GetNetworkStatistics(int channel, 24 | NetworkStatistics& stats); 25 | 26 | protected: 27 | VoENetEqStatsImpl(voe::SharedData* shared); 28 | virtual ~VoENetEqStatsImpl(); 29 | 30 | private: 31 | voe::SharedData* _shared; 32 | }; 33 | 34 | } // namespace webrtc 35 | 36 | #endif // WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H 37 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /****************************************************************** 12 | 13 | iLBC Speech Coder ANSI-C Source Code 14 | 15 | WebRtcIlbcfix_Poly2Lsf.c 16 | 17 | ******************************************************************/ 18 | 19 | #include "defines.h" 20 | #include "constants.h" 21 | #include "poly_to_lsp.h" 22 | #include "lsp_to_lsf.h" 23 | 24 | void WebRtcIlbcfix_Poly2Lsf( 25 | int16_t *lsf, /* (o) lsf coefficients (Q13) */ 26 | int16_t *a /* (i) A coefficients (Q12) */ 27 | ) { 28 | int16_t lsp[10]; 29 | WebRtcIlbcfix_Poly2Lsp(a, lsp, (int16_t*)WebRtcIlbcfix_kLspMean); 30 | WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10); 31 | } 32 | -------------------------------------------------------------------------------- /webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.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_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_ 12 | #define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_ 13 | 14 | class ViEAutoTestWindowManagerInterface; 15 | 16 | class ViEWindowManagerFactory { 17 | public: 18 | // This method is implemented in different files depending on platform. 19 | // The caller is responsible for freeing the resulting object using 20 | // the delete operator. 21 | static ViEAutoTestWindowManagerInterface* 22 | CreateWindowManagerForCurrentPlatform(); 23 | }; 24 | 25 | #endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_ 26 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/mac/audio_device_utility_mac.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 WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H 12 | #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H 13 | 14 | #include "audio_device_utility.h" 15 | #include "audio_device.h" 16 | 17 | namespace webrtc 18 | { 19 | class CriticalSectionWrapper; 20 | 21 | class AudioDeviceUtilityMac: public AudioDeviceUtility 22 | { 23 | public: 24 | AudioDeviceUtilityMac(const int32_t id); 25 | ~AudioDeviceUtilityMac(); 26 | 27 | virtual int32_t Init(); 28 | 29 | private: 30 | CriticalSectionWrapper& _critSect; 31 | int32_t _id; 32 | }; 33 | 34 | } // namespace webrtc 35 | 36 | #endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_UTILITY_MAC_H_ 37 | -------------------------------------------------------------------------------- /webrtc/modules/desktop_capture/desktop_geometry.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/desktop_geometry.h" 12 | 13 | #include 14 | 15 | namespace webrtc { 16 | 17 | void DesktopRect::IntersectWith(const DesktopRect& rect) { 18 | left_ = std::max(left(), rect.left()); 19 | top_ = std::max(top(), rect.top()); 20 | right_ = std::min(right(), rect.right()); 21 | bottom_ = std::min(bottom(), rect.top()); 22 | if (is_empty()) { 23 | left_ = 0; 24 | top_ = 0; 25 | right_ = 0; 26 | bottom_ = 0; 27 | } 28 | } 29 | 30 | void DesktopRect::Translate(int32_t dx, int32_t dy) { 31 | left_ += dx; 32 | top_ += dy; 33 | right_ += dx; 34 | bottom_ += dy; 35 | } 36 | 37 | } // namespace webrtc 38 | 39 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/event.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/system_wrappers/interface/event_wrapper.h" 12 | 13 | #if defined(_WIN32) 14 | #include 15 | #include "webrtc/system_wrappers/source/event_win.h" 16 | #elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) 17 | #include 18 | #include 19 | #include "webrtc/system_wrappers/source/event_posix.h" 20 | #else 21 | #include 22 | #include "webrtc/system_wrappers/source/event_posix.h" 23 | #endif 24 | 25 | namespace webrtc { 26 | EventWrapper* EventWrapper::Create() { 27 | #if defined(_WIN32) 28 | return new EventWindows(); 29 | #else 30 | return EventPosix::Create(); 31 | #endif 32 | } 33 | } // namespace webrtc 34 | -------------------------------------------------------------------------------- /webrtc/modules/video_render/mac/cocoa_render_view.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | // 12 | // cocoa_render_view.h 13 | // 14 | 15 | #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_ 16 | #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_ 17 | 18 | #import 19 | #import 20 | #import 21 | #import 22 | 23 | @interface CocoaRenderView : NSOpenGLView { 24 | NSOpenGLContext* _nsOpenGLContext; 25 | } 26 | 27 | -(void)initCocoaRenderView:(NSOpenGLPixelFormat*)fmt; 28 | -(void)initCocoaRenderViewFullScreen:(NSOpenGLPixelFormat*)fmt; 29 | -(NSOpenGLContext*)nsOpenGLContext; 30 | @end 31 | 32 | #endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_ 33 | -------------------------------------------------------------------------------- /webrtc/test/testsupport/mac/run_threaded_main_mac.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 | /** 12 | * This file and its corresponding .mm file implement a main function on Mac. 13 | * It's useful if you need to access a webcam in your Mac application. The code 14 | * forks a worker thread which runs the below ImplementThisToRunYourTest 15 | * function, and uses the main thread to pump messages. That way we can run our 16 | * code in a regular sequential fashion and still pump events, which are 17 | * necessary to access the webcam for instance. 18 | */ 19 | 20 | // Implement this method to do whatever you want to do in the worker thread. 21 | // The argc and argv variables are the unmodified command line from main. 22 | int ImplementThisToRunYourTest(int argc, char** argv); 23 | -------------------------------------------------------------------------------- /webrtc/voice_engine/test/auto_test/standard/hardware_before_initializing_test.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 "common_types.h" 12 | #include "before_initialization_fixture.h" 13 | 14 | using namespace webrtc; 15 | 16 | class HardwareBeforeInitializingTest : public BeforeInitializationFixture { 17 | }; 18 | 19 | TEST_F(HardwareBeforeInitializingTest, 20 | SetAudioDeviceLayerAcceptsPlatformDefaultBeforeInitializing) { 21 | AudioLayers wanted_layer = kAudioPlatformDefault; 22 | AudioLayers given_layer; 23 | EXPECT_EQ(0, voe_hardware_->SetAudioDeviceLayer(wanted_layer)); 24 | EXPECT_EQ(0, voe_hardware_->GetAudioDeviceLayer(given_layer)); 25 | EXPECT_EQ(wanted_layer, given_layer) << 26 | "These should be the same before initializing."; 27 | } 28 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/pcm16b/Android.mk: -------------------------------------------------------------------------------- 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 | LOCAL_PATH := $(call my-dir) 10 | 11 | include $(CLEAR_VARS) 12 | 13 | include $(LOCAL_PATH)/../../../../../android-webrtc.mk 14 | 15 | LOCAL_ARM_MODE := arm 16 | LOCAL_MODULE_CLASS := STATIC_LIBRARIES 17 | LOCAL_MODULE := libwebrtc_pcm16b 18 | LOCAL_MODULE_TAGS := optional 19 | LOCAL_SRC_FILES := pcm16b.c 20 | 21 | # Flags passed to both C and C++ files. 22 | LOCAL_CFLAGS := \ 23 | $(MY_WEBRTC_COMMON_DEFS) 24 | 25 | LOCAL_C_INCLUDES := \ 26 | $(LOCAL_PATH)/include \ 27 | $(LOCAL_PATH)/../../../.. 28 | 29 | LOCAL_SHARED_LIBRARIES := \ 30 | libcutils \ 31 | libdl \ 32 | libstlport 33 | 34 | ifndef NDK_ROOT 35 | include external/stlport/libstlport.mk 36 | endif 37 | include $(BUILD_STATIC_LIBRARY) 38 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/thread.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/system_wrappers/interface/thread_wrapper.h" 12 | 13 | #if defined(_WIN32) 14 | #include "webrtc/system_wrappers/source/thread_win.h" 15 | #else 16 | #include "webrtc/system_wrappers/source/thread_posix.h" 17 | #endif 18 | 19 | namespace webrtc { 20 | 21 | ThreadWrapper* ThreadWrapper::CreateThread(ThreadRunFunction func, 22 | ThreadObj obj, ThreadPriority prio, 23 | const char* thread_name) { 24 | #if defined(_WIN32) 25 | return new ThreadWindows(func, obj, prio, thread_name); 26 | #else 27 | return ThreadPosix::Create(func, obj, prio, thread_name); 28 | #endif 29 | } 30 | 31 | } // namespace webrtc 32 | -------------------------------------------------------------------------------- /webrtc/modules/audio_device/linux/audio_device_utility_linux.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_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H 12 | #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H 13 | 14 | #include "audio_device_utility.h" 15 | #include "audio_device.h" 16 | 17 | namespace webrtc 18 | { 19 | class CriticalSectionWrapper; 20 | 21 | class AudioDeviceUtilityLinux: public AudioDeviceUtility 22 | { 23 | public: 24 | AudioDeviceUtilityLinux(const int32_t id); 25 | ~AudioDeviceUtilityLinux(); 26 | 27 | virtual int32_t Init(); 28 | 29 | private: 30 | CriticalSectionWrapper& _critSect; 31 | int32_t _id; 32 | }; 33 | 34 | } // namespace webrtc 35 | 36 | #endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_LINUX_H_ 37 | -------------------------------------------------------------------------------- /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/source/system_wrappers.gyp:system_wrappers', 16 | ], 17 | 'include_dirs': [ 18 | '../interface', 19 | '../../../interface', 20 | '../../../../../../common_video/interface', 21 | ], 22 | 'direct_dependent_settings': { 23 | 'include_dirs': [ 24 | '../interface', 25 | '../../../../../../common_video/interface', 26 | ], 27 | }, 28 | 'sources': [ 29 | '../interface/i420.h', 30 | 'i420.cc', 31 | ], 32 | }, 33 | ], 34 | } 35 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq/mcu_address_init.c: -------------------------------------------------------------------------------- 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 "mcu.h" 12 | 13 | #include /* to define NULL */ 14 | 15 | /* 16 | * Initializes MCU with read address and write address 17 | */ 18 | int WebRtcNetEQ_McuAddressInit(MCUInst_t *inst, void * Data2McuAddress, 19 | void * Data2DspAddress, void *main_inst) 20 | { 21 | 22 | inst->pw16_readAddress = (int16_t*) Data2McuAddress; 23 | inst->pw16_writeAddress = (int16_t*) Data2DspAddress; 24 | inst->main_inst = main_inst; 25 | 26 | inst->millisecondsPerCall = 10; 27 | 28 | /* Do expansions in the beginning */ 29 | if (inst->pw16_writeAddress != NULL) inst->pw16_writeAddress[0] = DSP_INSTR_EXPAND; 30 | 31 | return (0); 32 | } 33 | 34 | -------------------------------------------------------------------------------- /webrtc/test/Android.mk: -------------------------------------------------------------------------------- 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 | LOCAL_PATH := $(call my-dir) 11 | 12 | include $(CLEAR_VARS) 13 | include $(LOCAL_PATH)/../../android-webrtc.mk 14 | 15 | LOCAL_ARM_MODE := arm 16 | LOCAL_MODULE:= libwebrtc_test_support 17 | LOCAL_MODULE_TAGS := optional 18 | LOCAL_CPP_EXTENSION := .cc 19 | LOCAL_SRC_FILES:= \ 20 | testsupport/fileutils.cc \ 21 | testsupport/perf_test.cc 22 | 23 | # Flags passed to both C and C++ files. 24 | LOCAL_CFLAGS := \ 25 | $(MY_WEBRTC_COMMON_DEFS) 26 | 27 | LOCAL_C_INCLUDES := \ 28 | external/gtest/include \ 29 | external/webrtc \ 30 | external/webrtc/webrtc 31 | 32 | LOCAL_STATIC_LIBRARIES := \ 33 | libgtest 34 | 35 | ifndef NDK_ROOT 36 | include external/stlport/libstlport.mk 37 | endif 38 | include $(BUILD_STATIC_LIBRARY) 39 | 40 | -------------------------------------------------------------------------------- /webrtc/test/testsupport/perf_test_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 | #include "webrtc/test/testsupport/perf_test.h" 12 | 13 | #include 14 | 15 | #include "gtest/gtest.h" 16 | 17 | namespace webrtc { 18 | namespace test { 19 | 20 | TEST(PerfTest, AppendResult) { 21 | std::string expected = "RESULT measurementmodifier: trace= 42 units\n"; 22 | std::string output; 23 | AppendResult(output, "measurement", "modifier", "trace", 42, "units", false); 24 | EXPECT_EQ(expected, output); 25 | std::cout << output; 26 | 27 | expected += "*RESULT foobar: baz= 7 widgets\n"; 28 | AppendResult(output, "foo", "bar", "baz", 7, "widgets", true); 29 | EXPECT_EQ(expected, output); 30 | std::cout << output; 31 | } 32 | 33 | } // namespace test 34 | } // namespace webrtc 35 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/critical_section_posix.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_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_ 12 | #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_ 13 | 14 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 15 | 16 | #include 17 | 18 | namespace webrtc { 19 | 20 | class CriticalSectionPosix : public CriticalSectionWrapper { 21 | public: 22 | CriticalSectionPosix(); 23 | 24 | virtual ~CriticalSectionPosix(); 25 | 26 | virtual void Enter(); 27 | virtual void Leave(); 28 | 29 | private: 30 | pthread_mutex_t mutex_; 31 | friend class ConditionVariablePosix; 32 | }; 33 | 34 | } // namespace webrtc 35 | 36 | #endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_ 37 | -------------------------------------------------------------------------------- /tools/quality_tracking/dashboard/stylesheets/stylesheet.css: -------------------------------------------------------------------------------- 1 | /******************************************************************** 2 | * 3 | * Copyright (c) 2012 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 | *********************************************************************/ 12 | 13 | .status_OK { 14 | color: #FFFFFF; 15 | background-color: #8fdf5f; 16 | } 17 | 18 | .status_failed { 19 | color: #FFFFFF; 20 | background-color: #e98080; 21 | } 22 | 23 | .status_building { 24 | color: #666666; 25 | background-color: #fffc6c; 26 | } 27 | 28 | .status_warnings { 29 | color: #000000; 30 | background-color: #FFC343; 31 | } 32 | 33 | .last_known_good_revision { 34 | font-size: 800%; 35 | } 36 | 37 | .status_cell { 38 | width: 100px; 39 | text-align: center; 40 | } 41 | 42 | body { 43 | margin-left: 35px; 44 | margin-top: 25px; 45 | } -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq4/time_stretch_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 Accelerate and PreemptiveExpand classes. 12 | 13 | #include "webrtc/modules/audio_coding/neteq4/accelerate.h" 14 | #include "webrtc/modules/audio_coding/neteq4/preemptive_expand.h" 15 | 16 | #include "gtest/gtest.h" 17 | #include "webrtc/modules/audio_coding/neteq4/background_noise.h" 18 | 19 | namespace webrtc { 20 | 21 | TEST(TimeStretch, CreateAndDestroy) { 22 | int sample_rate = 8000; 23 | size_t num_channels = 1; 24 | BackgroundNoise bgn(num_channels); 25 | Accelerate accelerate(sample_rate, num_channels, bgn); 26 | PreemptiveExpand preemptive_expand(sample_rate, num_channels, bgn); 27 | } 28 | 29 | // TODO(hlundin): Write more tests. 30 | 31 | } // namespace webrtc 32 | -------------------------------------------------------------------------------- /webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.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/modules/rtp_rtcp/source/rtp_receiver_strategy.h" 12 | 13 | #include 14 | 15 | namespace webrtc { 16 | 17 | RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback) 18 | : data_callback_(data_callback) { 19 | memset(&last_payload_, 0, sizeof(last_payload_)); 20 | } 21 | 22 | void RTPReceiverStrategy::GetLastMediaSpecificPayload( 23 | ModuleRTPUtility::PayloadUnion* payload) const { 24 | memcpy(payload, &last_payload_, sizeof(*payload)); 25 | } 26 | 27 | void RTPReceiverStrategy::SetLastMediaSpecificPayload( 28 | const ModuleRTPUtility::PayloadUnion& payload) { 29 | memcpy(&last_payload_, &payload, sizeof(last_payload_)); 30 | } 31 | 32 | } // namespace webrtc 33 | -------------------------------------------------------------------------------- /samples/js/demos/html/gum2.html: -------------------------------------------------------------------------------- 1 | 2 | 3 | 4 | getUserMedia Demo 2 5 | 21 | 22 | 23 | 24 | 25 |
26 | 27 | 28 | 46 | 47 | 48 | 49 | -------------------------------------------------------------------------------- /tools/quality_tracking/dashboard/main.py: -------------------------------------------------------------------------------- 1 | #!/usr/bin/env python 2 | #-*- coding: utf-8 -*- 3 | # Copyright (c) 2012 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 | """Connects all URLs with their respective handlers.""" 12 | 13 | from google.appengine.ext.webapp import template 14 | import webapp2 15 | 16 | import add_build_status_data 17 | import add_coverage_data 18 | import dashboard 19 | import lkgr_page 20 | 21 | app = webapp2.WSGIApplication([('/', dashboard.ShowDashboard), 22 | ('/lkgr', lkgr_page.ShowLkgr), 23 | ('/add_coverage_data', 24 | add_coverage_data.AddCoverageData), 25 | ('/add_build_status_data', 26 | add_build_status_data.AddBuildStatusData)], 27 | debug=True) -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/neteq4/expand_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 Expand class. 12 | 13 | #include "webrtc/modules/audio_coding/neteq4/expand.h" 14 | 15 | #include "gtest/gtest.h" 16 | #include "webrtc/modules/audio_coding/neteq4/background_noise.h" 17 | #include "webrtc/modules/audio_coding/neteq4/random_vector.h" 18 | #include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" 19 | 20 | namespace webrtc { 21 | 22 | TEST(Expand, CreateAndDestroy) { 23 | int fs = 8000; 24 | size_t channels = 1; 25 | BackgroundNoise bgn(channels); 26 | SyncBuffer sync_buffer(1, 1000); 27 | RandomVector random_vector; 28 | Expand expand(&bgn, &sync_buffer, &random_vector, fs, channels); 29 | } 30 | 31 | // TODO(hlundin): Write more tests. 32 | 33 | } // namespace webrtc 34 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/rw_lock.cc: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" 12 | 13 | #include 14 | 15 | #if defined(_WIN32) 16 | #include "webrtc/system_wrappers/source/rw_lock_generic.h" 17 | #include "webrtc/system_wrappers/source/rw_lock_win.h" 18 | #else 19 | #include "webrtc/system_wrappers/source/rw_lock_posix.h" 20 | #endif 21 | 22 | namespace webrtc { 23 | 24 | RWLockWrapper* RWLockWrapper::CreateRWLock() { 25 | #ifdef _WIN32 26 | // Native implementation is faster, so use that if available. 27 | RWLockWrapper* lock = RWLockWin::Create(); 28 | if (lock) { 29 | return lock; 30 | } 31 | return new RWLockGeneric(); 32 | #else 33 | return RWLockPosix::Create(); 34 | #endif 35 | } 36 | 37 | } // namespace webrtc 38 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /****************************************************************** 12 | 13 | iLBC Speech Coder ANSI-C Source Code 14 | 15 | WebRtcIlbcfix_FrameClassify.h 16 | 17 | ******************************************************************/ 18 | 19 | #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_ 20 | #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_ 21 | 22 | int16_t WebRtcIlbcfix_FrameClassify( 23 | /* (o) Index to the max-energy sub frame */ 24 | iLBC_Enc_Inst_t *iLBCenc_inst, 25 | /* (i/o) the encoder state structure */ 26 | int16_t *residualFIX /* (i) lpc residual signal */ 27 | ); 28 | 29 | #endif 30 | -------------------------------------------------------------------------------- /webrtc/modules/utility/interface/process_thread.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_UTILITY_INTERFACE_PROCESS_THREAD_H_ 12 | #define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_ 13 | 14 | #include "typedefs.h" 15 | 16 | namespace webrtc { 17 | class Module; 18 | 19 | class ProcessThread 20 | { 21 | public: 22 | static ProcessThread* CreateProcessThread(); 23 | static void DestroyProcessThread(ProcessThread* module); 24 | 25 | virtual int32_t Start() = 0; 26 | virtual int32_t Stop() = 0; 27 | 28 | virtual int32_t RegisterModule(const Module* module) = 0; 29 | virtual int32_t DeRegisterModule(const Module* module) = 0; 30 | protected: 31 | virtual ~ProcessThread(); 32 | }; 33 | } // namespace webrtc 34 | #endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_ 35 | -------------------------------------------------------------------------------- /webrtc/video_engine/vie_ref_count.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 | // TODO(mflodman) Remove this class and use ref count class in system_wrappers. 12 | 13 | #ifndef WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_ 14 | #define WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_ 15 | 16 | #include "system_wrappers/interface/scoped_ptr.h" 17 | 18 | namespace webrtc { 19 | 20 | class CriticalSectionWrapper; 21 | 22 | class ViERefCount { 23 | public: 24 | ViERefCount(); 25 | ~ViERefCount(); 26 | 27 | ViERefCount& operator++(int); // NOLINT 28 | ViERefCount& operator--(int); // NOLINT 29 | 30 | void Reset(); 31 | int GetCount() const; 32 | 33 | private: 34 | volatile int count_; 35 | scoped_ptr crit_; 36 | }; 37 | 38 | } // namespace webrtc 39 | 40 | #endif // WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_ 41 | -------------------------------------------------------------------------------- /webrtc/common_audio/signal_processing/energy.c: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | 12 | /* 13 | * This file contains the function WebRtcSpl_Energy(). 14 | * The description header can be found in signal_processing_library.h 15 | * 16 | */ 17 | 18 | #include "signal_processing_library.h" 19 | 20 | int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor) 21 | { 22 | int32_t en = 0; 23 | int i; 24 | int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length); 25 | int looptimes = vector_length; 26 | int16_t *vectorptr = vector; 27 | 28 | for (i = 0; i < looptimes; i++) 29 | { 30 | en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling); 31 | vectorptr++; 32 | } 33 | *scale_factor = scaling; 34 | 35 | return en; 36 | } 37 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/vp8/include/vp8.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 | * WEBRTC VP8 wrapper interface 11 | */ 12 | 13 | #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_ 14 | #define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_ 15 | 16 | #include "modules/video_coding/codecs/interface/video_codec_interface.h" 17 | 18 | namespace webrtc { 19 | 20 | class VP8Encoder : public VideoEncoder { 21 | public: 22 | static VP8Encoder* Create(); 23 | 24 | virtual ~VP8Encoder() {}; 25 | }; // end of VP8Encoder class 26 | 27 | 28 | class VP8Decoder : public VideoDecoder { 29 | public: 30 | static VP8Decoder* Create(); 31 | 32 | virtual ~VP8Decoder() {}; 33 | }; // end of VP8Decoder class 34 | } // namespace webrtc 35 | 36 | #endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_ 37 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/rw_lock_win.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 WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_ 12 | #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_ 13 | 14 | #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" 15 | 16 | #include 17 | 18 | namespace webrtc { 19 | 20 | class RWLockWin : public RWLockWrapper { 21 | public: 22 | static RWLockWin* Create(); 23 | ~RWLockWin() {} 24 | 25 | virtual void AcquireLockExclusive(); 26 | virtual void ReleaseLockExclusive(); 27 | 28 | virtual void AcquireLockShared(); 29 | virtual void ReleaseLockShared(); 30 | 31 | private: 32 | RWLockWin(); 33 | static bool LoadModule(); 34 | 35 | SRWLOCK lock_; 36 | }; 37 | 38 | } // namespace webrtc 39 | 40 | #endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_ 41 | -------------------------------------------------------------------------------- /webrtc/modules/video_capture/video_capture_factory.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 "video_capture_factory.h" 12 | #include "video_capture_impl.h" 13 | 14 | namespace webrtc 15 | { 16 | 17 | VideoCaptureModule* VideoCaptureFactory::Create(const int32_t id, 18 | const char* deviceUniqueIdUTF8) { 19 | return videocapturemodule::VideoCaptureImpl::Create(id, deviceUniqueIdUTF8); 20 | } 21 | 22 | VideoCaptureModule* VideoCaptureFactory::Create(const int32_t id, 23 | VideoCaptureExternal*& externalCapture) { 24 | return videocapturemodule::VideoCaptureImpl::Create(id, externalCapture); 25 | } 26 | 27 | VideoCaptureModule::DeviceInfo* VideoCaptureFactory::CreateDeviceInfo( 28 | const int32_t id) { 29 | return videocapturemodule::VideoCaptureImpl::CreateDeviceInfo(id); 30 | } 31 | 32 | } // namespace webrtc 33 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACfault.txt: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | (set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character 3 | 4 | LOGFILE=logfault.txt 5 | echo "START FAULT TEST" > $LOGFILE 6 | 7 | ISAC=../Release/kenny.exe 8 | ISACFIXFLOAT=../Release/testFixFloat.exe 9 | 10 | INFILES=$(cat InputFiles.txt) 11 | SUBSET=$(cat InputFilesFew.txt) 12 | CHANNELFILES=$(cat ChannelFiles.txt) 13 | CHANNELLIST=($(cat ChannelFiles.txt)) 14 | INDIR=../data/orig 15 | OUTDIR=../dataqaft 16 | mkdir -p $OUTDIR 17 | 18 | TARGETRATE=(10000 15000 20000 25000 30000 32000) 19 | FAULTTEST=(1 2 3 4 5 6 7 9) 20 | 21 | index1=0 22 | 23 | file=wb_contspeech.pcm 24 | 25 | # Fault test 26 | for testnr in ${FAULTTEST[*]} 27 | do 28 | $ISAC 32000 -F $testnr $INDIR/"$file" $OUTDIR/ft$testnr"$file" >> $LOGFILE 29 | done 30 | 31 | # Fault test number 10, error in bitstream 32 | $ISAC 32000 -F 10 $INDIR/"$file" $OUTDIR/ft10_"$file" >> $LOGFILE 33 | $ISAC 32000 -F 10 -PL 10 $INDIR/"$file" $OUTDIR/ft10plc_"$file" >> $LOGFILE 34 | $ISAC 32000 -F 10 -NB 1 $INDIR/"$file" $OUTDIR/ft10nb1_"$file" >> $LOGFILE 35 | $ISAC 32000 -F 10 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/ft10nb2_"$file" >> $LOGFILE 36 | 37 | echo DONE! 38 | 39 | 40 | 41 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.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_CODING_CODECS_VP8_COMMON_TYPES_H_ 12 | #define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_ 13 | 14 | #include "common_types.h" 15 | 16 | namespace webrtc { 17 | 18 | // Ratio allocation between temporal streams: 19 | // Values as required for the VP8 codec (accumulating). 20 | static const float 21 | kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = { 22 | {1.0f, 0, 0, 0}, // 1 layer 23 | {0.6f, 1.0f , 0 , 0}, // 2 layers {60%, 40%} 24 | {0.4f, 0.6f , 1.0f, 0}, // 3 layers {40%, 20%, 40%} 25 | {0.25f, 0.4f, 0.6f, 1.0f} // 4 layers {25%, 15%, 20%, 40%} 26 | }; 27 | 28 | } // namespace webrtc 29 | #endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_ 30 | -------------------------------------------------------------------------------- /webrtc/modules/video_processing/main/test/vpm_tests.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': 'video_processing_unittests', 13 | 'type': 'executable', 14 | 'dependencies': [ 15 | 'video_processing', 16 | 'webrtc_utility', 17 | '<(webrtc_root)/test/test.gyp:test_support_main', 18 | '<(DEPTH)/testing/gtest.gyp:gtest', 19 | ], 20 | 'sources': [ 21 | # headers 22 | 'unit_test/unit_test.h', 23 | # sources 24 | 'unit_test/brightness_detection_test.cc', 25 | 'unit_test/color_enhancement_test.cc', 26 | 'unit_test/content_metrics_test.cc', 27 | 'unit_test/deflickering_test.cc', 28 | 'unit_test/denoising_test.cc', 29 | 'unit_test/unit_test.cc', 30 | ], # sources 31 | }, 32 | ], 33 | } 34 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/g711/Android.mk: -------------------------------------------------------------------------------- 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 | LOCAL_PATH := $(call my-dir) 10 | 11 | include $(CLEAR_VARS) 12 | 13 | include $(LOCAL_PATH)/../../../../../android-webrtc.mk 14 | 15 | LOCAL_ARM_MODE := arm 16 | LOCAL_MODULE_CLASS := STATIC_LIBRARIES 17 | LOCAL_MODULE := libwebrtc_g711 18 | LOCAL_MODULE_TAGS := optional 19 | LOCAL_GENERATED_SOURCES := 20 | LOCAL_SRC_FILES := \ 21 | g711_interface.c \ 22 | g711.c 23 | 24 | # Flags passed to both C and C++ files. 25 | LOCAL_CFLAGS := \ 26 | $(MY_WEBRTC_COMMON_DEFS) 27 | 28 | LOCAL_C_INCLUDES := \ 29 | $(LOCAL_PATH)/include \ 30 | $(LOCAL_PATH)/../../../.. 31 | 32 | LOCAL_SHARED_LIBRARIES := \ 33 | libcutils \ 34 | libdl \ 35 | libstlport 36 | 37 | ifndef NDK_ROOT 38 | include external/stlport/libstlport.mk 39 | endif 40 | include $(BUILD_STATIC_LIBRARY) 41 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/g722/Android.mk: -------------------------------------------------------------------------------- 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 | LOCAL_PATH := $(call my-dir) 10 | 11 | include $(CLEAR_VARS) 12 | 13 | include $(LOCAL_PATH)/../../../../../android-webrtc.mk 14 | 15 | LOCAL_ARM_MODE := arm 16 | LOCAL_MODULE_CLASS := STATIC_LIBRARIES 17 | LOCAL_MODULE := libwebrtc_g722 18 | LOCAL_MODULE_TAGS := optional 19 | LOCAL_SRC_FILES := \ 20 | g722_interface.c \ 21 | g722_encode.c \ 22 | g722_decode.c 23 | 24 | # Flags passed to both C and C++ files. 25 | LOCAL_CFLAGS := \ 26 | $(MY_WEBRTC_COMMON_DEFS) 27 | 28 | LOCAL_C_INCLUDES := \ 29 | $(LOCAL_PATH)/include \ 30 | $(LOCAL_PATH)/../../../.. 31 | 32 | LOCAL_SHARED_LIBRARIES := \ 33 | libcutils \ 34 | libdl \ 35 | libstlport 36 | 37 | ifndef NDK_ROOT 38 | include external/stlport/libstlport.mk 39 | endif 40 | include $(BUILD_STATIC_LIBRARY) 41 | -------------------------------------------------------------------------------- /webrtc/modules/audio_conference_mixer/source/level_indicator.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_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_ 12 | #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_ 13 | 14 | #include "typedefs.h" 15 | 16 | namespace webrtc { 17 | class LevelIndicator 18 | { 19 | public: 20 | enum{TICKS_BEFORE_CALCULATION = 10}; 21 | 22 | LevelIndicator(); 23 | ~LevelIndicator(); 24 | 25 | // Updates the level. 26 | void ComputeLevel(const int16_t* speech, 27 | const uint16_t nrOfSamples); 28 | 29 | int32_t GetLevel(); 30 | private: 31 | int32_t _max; 32 | uint32_t _count; 33 | uint32_t _currentLevel; 34 | }; 35 | } // namespace webrtc 36 | 37 | #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_ 38 | -------------------------------------------------------------------------------- /webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.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 WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_ 12 | #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_ 13 | 14 | #include 15 | 16 | #include 17 | 18 | #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" 19 | 20 | namespace webrtc { 21 | 22 | class MockRemoteBitrateObserver : public RemoteBitrateObserver { 23 | public: 24 | MOCK_METHOD2(OnReceiveBitrateChanged, 25 | void(std::vector* ssrcs, unsigned int bitrate)); 26 | }; 27 | 28 | } // namespace webrtc 29 | 30 | #endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_ 31 | -------------------------------------------------------------------------------- /webrtc/modules/video_coding/utility/Android.mk: -------------------------------------------------------------------------------- 1 | # Copyright (c) 2013 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 | LOCAL_PATH := $(call my-dir) 10 | 11 | include $(CLEAR_VARS) 12 | 13 | include $(LOCAL_PATH)/../../../../android-webrtc.mk 14 | 15 | LOCAL_ARM_MODE := arm 16 | LOCAL_MODULE_CLASS := STATIC_LIBRARIES 17 | LOCAL_MODULE := libvideo_coding_utility 18 | LOCAL_MODULE_TAGS := optional 19 | LOCAL_CPP_EXTENSION := .cc 20 | LOCAL_SRC_FILES := \ 21 | exp_filter.cc \ 22 | frame_dropper.cc \ 23 | 24 | # Flags passed to both C and C++ files. 25 | LOCAL_CFLAGS := \ 26 | $(MY_WEBRTC_COMMON_DEFS) 27 | 28 | LOCAL_C_INCLUDES := \ 29 | $(LOCAL_PATH)/../../../../system_wrappers/interface 30 | 31 | LOCAL_SHARED_LIBRARIES := \ 32 | libcutils \ 33 | libdl \ 34 | libstlport 35 | 36 | ifndef NDK_ROOT 37 | include external/stlport/libstlport.mk 38 | endif 39 | include $(BUILD_STATIC_LIBRARY) 40 | -------------------------------------------------------------------------------- /webrtc/system_wrappers/source/rw_lock_posix.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_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_ 12 | #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_ 13 | 14 | #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" 15 | 16 | #include 17 | 18 | namespace webrtc { 19 | 20 | class RWLockPosix : public RWLockWrapper { 21 | public: 22 | static RWLockPosix* Create(); 23 | virtual ~RWLockPosix(); 24 | 25 | virtual void AcquireLockExclusive(); 26 | virtual void ReleaseLockExclusive(); 27 | 28 | virtual void AcquireLockShared(); 29 | virtual void ReleaseLockShared(); 30 | 31 | private: 32 | RWLockPosix(); 33 | bool Init(); 34 | 35 | pthread_rwlock_t lock_; 36 | }; 37 | 38 | } // namespace webrtc 39 | 40 | #endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_ 41 | -------------------------------------------------------------------------------- /webrtc/voice_engine/voe_encryption_impl.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 WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H 12 | #define WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H 13 | 14 | #include "voe_encryption.h" 15 | 16 | #include "shared_data.h" 17 | 18 | namespace webrtc { 19 | 20 | class VoEEncryptionImpl : public VoEEncryption 21 | { 22 | public: 23 | // External encryption 24 | virtual int RegisterExternalEncryption( 25 | int channel, 26 | Encryption& encryption); 27 | 28 | virtual int DeRegisterExternalEncryption(int channel); 29 | 30 | protected: 31 | VoEEncryptionImpl(voe::SharedData* shared); 32 | virtual ~VoEEncryptionImpl(); 33 | 34 | private: 35 | voe::SharedData* _shared; 36 | }; 37 | 38 | } // namespace webrtc 39 | 40 | #endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H 41 | -------------------------------------------------------------------------------- /webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h: -------------------------------------------------------------------------------- 1 | /* 2 | * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 | * 4 | * Use of this source code is governed by a BSD-style license 5 | * that can be found in the LICENSE file in the root of the source 6 | * tree. An additional intellectual property rights grant can be found 7 | * in the file PATENTS. All contributing project authors may 8 | * be found in the AUTHORS file in the root of the source tree. 9 | */ 10 | 11 | /****************************************************************** 12 | 13 | iLBC Speech Coder ANSI-C Source Code 14 | 15 | WebRtcIlbcfix_GetCbVec.h 16 | 17 | ******************************************************************/ 18 | 19 | #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_ 20 | #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_ 21 | 22 | void WebRtcIlbcfix_GetCbVec( 23 | int16_t *cbvec, /* (o) Constructed codebook vector */ 24 | int16_t *mem, /* (i) Codebook buffer */ 25 | int16_t index, /* (i) Codebook index */ 26 | int16_t lMem, /* (i) Length of codebook buffer */ 27 | int16_t cbveclen /* (i) Codebook vector length */ 28 | ); 29 | 30 | #endif 31 | --------------------------------------------------------------------------------