gecko-dev/third_party/libwebrtc/README.md

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Bug 1250356: Webrtc updated to upstream branch 49; pull made 2016-02-22 by pkerr rs=jesup --HG-- rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/OWNERS => media/webrtc/trunk/webrtc/audio/OWNERS rename : media/webrtc/trunk/webrtc/video_engine/OWNERS => media/webrtc/trunk/webrtc/call/OWNERS rename : media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator.cc => media/webrtc/trunk/webrtc/call/bitrate_allocator.cc rename : media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_allocator.h => media/webrtc/trunk/webrtc/call/bitrate_allocator.h rename : media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc => media/webrtc/trunk/webrtc/call/bitrate_allocator_unittest.cc rename : media/webrtc/trunk/webrtc/video/call_perf_tests.cc => media/webrtc/trunk/webrtc/call/call_perf_tests.cc rename : media/webrtc/trunk/webrtc/common_video/interface/i420_buffer_pool.h => media/webrtc/trunk/webrtc/common_video/include/i420_buffer_pool.h rename : media/webrtc/trunk/webrtc/common_video/interface/video_image.h => media/webrtc/trunk/webrtc/common_video/include/video_image.h rename : media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.cc => media/webrtc/trunk/webrtc/common_video/video_render_frames.cc rename : media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.h => media/webrtc/trunk/webrtc/common_video/video_render_frames.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/audio_codec_speed_tests.isolate => media/webrtc/trunk/webrtc/modules/audio_codec_speed_tests.isolate rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/interface/neteq.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/include/neteq.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ACMTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/ACMTest.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Tester.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/Tester.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/delay_test.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/delay_test.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/target_delay_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.cc rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.h rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate => media/webrtc/trunk/webrtc/modules/audio_decoder_unittests.isolate rename : media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_device/audio_device_tests.isolate => media/webrtc/trunk/webrtc/modules/audio_device_tests.isolate rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/include/echo_cancellation.h => media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression.h => media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h => media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/common.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/common.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.cc rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.cc => 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media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.cc rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.h rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc rename : media/webrtc/trunk/webrtc/modules/interface/module.h => media/webrtc/trunk/webrtc/modules/include/module.h rename : media/webrtc/trunk/webrtc/modules/interface/module_common_types.h => media/webrtc/trunk/webrtc/modules/include/module_common_types.h rename : media/webrtc/trunk/webrtc/modules/media_file/interface/media_file.h => media/webrtc/trunk/webrtc/modules/media_file/media_file.h rename : media/webrtc/trunk/webrtc/modules/media_file/interface/media_file_defines.h => media/webrtc/trunk/webrtc/modules/media_file/media_file_defines.h rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.cc => media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.cc rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.h => media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.h rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_unittest.cc => media/webrtc/trunk/webrtc/modules/media_file/media_file_unittest.cc rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.cc => media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.cc rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.h => media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.h rename : media/webrtc/trunk/webrtc/modules/pacing/include/mock/mock_paced_sender.h => media/webrtc/trunk/webrtc/modules/pacing/mock/mock_paced_sender.h rename : media/webrtc/trunk/webrtc/modules/pacing/include/paced_sender.h => media/webrtc/trunk/webrtc/modules/pacing/paced_sender.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/fec_receiver.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/fec_receiver.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/receive_statistics.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/receive_statistics.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_cvo.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_receiver.h rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h rename : media/webrtc/trunk/webrtc/modules/utility/interface/audio_frame_operations.h => media/webrtc/trunk/webrtc/modules/utility/include/audio_frame_operations.h rename : media/webrtc/trunk/webrtc/modules/utility/interface/file_player.h => media/webrtc/trunk/webrtc/modules/utility/include/file_player.h rename : media/webrtc/trunk/webrtc/modules/utility/interface/file_recorder.h => media/webrtc/trunk/webrtc/modules/utility/include/file_recorder.h rename : media/webrtc/trunk/webrtc/modules/utility/interface/mock/mock_process_thread.h => media/webrtc/trunk/webrtc/modules/utility/include/mock/mock_process_thread.h rename : media/webrtc/trunk/webrtc/modules/utility/interface/process_thread.h => media/webrtc/trunk/webrtc/modules/utility/include/process_thread.h rename : media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture.h => media/webrtc/trunk/webrtc/modules/video_capture/video_capture.h rename : media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_defines.h => 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media/webrtc/trunk/webrtc/voice_engine/mock/mock_voe_observer.h
2016-12-28 03:40:48 +03:00
**WebRTC is a free, open software project** that provides browsers and mobile
applications with Real-Time Communications (RTC) capabilities via simple APIs.
The WebRTC components have been optimized to best serve this purpose.
**Our mission:** To enable rich, high-quality RTC applications to be
developed for the browser, mobile platforms, and IoT devices, and allow them
all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera,
amongst others. This page is maintained by the Google Chrome team.
### Development
See http://www.webrtc.org/native-code/development for instructions on how to get
started developing with the native code.
### More info
* Official web site: http://www.webrtc.org
* Master source code repo: https://chromium.googlesource.com/external/webrtc
* Samples and reference apps: https://github.com/webrtc
* Mailing list: http://groups.google.com/group/discuss-webrtc
* Continuous build: http://build.chromium.org/p/client.webrtc