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android
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Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199)
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2018-02-22 13:58:58 +00:00 |
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fuzzers
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Add saza to fuzzer owners
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2018-07-12 11:12:00 +00:00 |
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gl
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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ios
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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linux
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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mac
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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testsupport
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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win
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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BUILD.gn
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Revert "Add Profile 2 configuration to VP9 Encoder and Decoder"
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2018-07-11 06:04:49 +00:00 |
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call_test.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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call_test.h
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Reland "Move creating encoder to VideoStreamEncoder."
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2018-04-19 08:48:58 +00:00 |
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configurable_frame_size_encoder.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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configurable_frame_size_encoder.h
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Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
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2018-04-23 15:31:27 +00:00 |
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constants.cc
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Remove voe_auto_test and add new tests to cover the missing cases.
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2017-09-15 16:56:08 +00:00 |
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constants.h
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Remove voe_auto_test and add new tests to cover the missing cases.
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2017-09-15 16:56:08 +00:00 |
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DEPS
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Reland "Add multiplex case to webrtc_perf_tests"
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2018-03-10 01:21:04 +00:00 |
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direct_transport.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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direct_transport.h
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Moving demux from FakeNetworkPipe to DirectTransport.
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2018-04-25 10:13:03 +00:00 |
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direct_transport_unittest.cc
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Moving demux from FakeNetworkPipe to DirectTransport.
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2018-04-25 10:13:03 +00:00 |
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drifting_clock.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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drifting_clock.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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encoder_proxy_factory.h
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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encoder_settings.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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encoder_settings.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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fake_decoder.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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fake_decoder.h
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Delete pre_decode_callback.
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2018-06-20 07:04:09 +00:00 |
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fake_encoder.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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fake_encoder.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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fake_texture_frame.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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fake_texture_frame.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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fake_videorenderer.h
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New file api/video/BUILD.gn
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2018-05-14 06:57:38 +00:00 |
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field_trial.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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field_trial.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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frame_generator.cc
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Revert "Add Profile 2 configuration to VP9 Encoder and Decoder"
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2018-07-11 06:04:49 +00:00 |
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frame_generator.h
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Revert "Add Profile 2 configuration to VP9 Encoder and Decoder"
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2018-07-11 06:04:49 +00:00 |
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frame_generator_capturer.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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frame_generator_capturer.h
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Replace rtc::Optional with absl::optional in test and rtc_tools
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2018-06-18 13:15:23 +00:00 |
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frame_generator_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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frame_utils.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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frame_utils.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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function_video_decoder_factory.h
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Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl}
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2018-05-31 11:48:17 +00:00 |
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function_video_encoder_factory.h
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Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl}
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2018-05-31 11:48:17 +00:00 |
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gmock.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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gtest.h
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Reland "Adding gtest-spi.h in webrtc/test/gtest.h"
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2018-04-05 08:21:23 +00:00 |
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layer_filtering_transport.cc
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Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
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2018-07-05 14:29:07 +00:00 |
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layer_filtering_transport.h
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Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted.
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2018-06-21 17:53:35 +00:00 |
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mock_audio_decoder.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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mock_audio_decoder.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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mock_audio_decoder_factory.h
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Replace rtc::Optional with absl::optional in test and rtc_tools
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2018-06-18 13:15:23 +00:00 |
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mock_audio_encoder.cc
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Removed Die mock from MockAudioEncoder
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2018-02-22 12:53:38 +00:00 |
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mock_audio_encoder.h
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Replace rtc::Optional with absl::optional in test and rtc_tools
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2018-06-18 13:15:23 +00:00 |
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mock_audio_encoder_factory.h
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Replace rtc::Optional with absl::optional in test and rtc_tools
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2018-06-18 13:15:23 +00:00 |
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mock_transport.cc
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Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
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2018-03-09 16:04:35 +00:00 |
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mock_transport.h
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Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
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2018-03-09 16:04:35 +00:00 |
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null_platform_renderer.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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null_transport.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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null_transport.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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OWNERS
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Remove pbos@webrtc.org from all OWNERS.
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2017-11-01 08:03:46 +00:00 |
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rtcp_packet_parser.cc
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Stop using LOG macros in favor of RTC_ prefixed macros.
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2017-11-09 11:56:32 +00:00 |
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rtcp_packet_parser.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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rtp_file_reader.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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rtp_file_reader.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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rtp_file_reader_unittest.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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rtp_file_writer.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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rtp_file_writer.h
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Adding NOLINT for typedefs.h and common_types.h
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2017-09-15 13:03:51 +00:00 |
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rtp_file_writer_unittest.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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rtp_rtcp_observer.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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run_loop.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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run_loop.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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run_test.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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run_test.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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single_threaded_task_queue.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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single_threaded_task_queue.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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single_threaded_task_queue_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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statistics.cc
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Reland "Updated analysis in videoprocessor."
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2018-01-18 08:37:27 +00:00 |
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statistics.h
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Reland "Updated analysis in videoprocessor."
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2018-01-18 08:37:27 +00:00 |
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test_main.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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vcm_capturer.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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vcm_capturer.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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video_capturer.cc
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Replace rtc::Optional with absl::optional in test and rtc_tools
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2018-06-18 13:15:23 +00:00 |
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video_capturer.h
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Replace rtc::Optional with absl::optional in test and rtc_tools
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2018-06-18 13:15:23 +00:00 |
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video_codec_settings.h
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Delete RTP-specific values from the VideoCodecType enum.
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2018-06-07 07:49:27 +00:00 |
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video_renderer.cc
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Adding NOLINT for typedefs.h and common_types.h
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2017-09-15 13:03:51 +00:00 |
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video_renderer.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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