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greatitman/webrtc-src

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文件
api
audio
build_overrides
call
adaptation
test
BUILD.gn
DEPS
OWNERS
audio_receive_stream.cc
audio_receive_stream.h
audio_send_stream.cc
audio_send_stream.h
audio_sender.h
audio_state.cc
audio_state.h
bitrate_allocator.cc
bitrate_allocator.h
bitrate_allocator_unittest.cc
bitrate_estimator_tests.cc
call.cc
call.h
call_config.cc
call_config.h
call_factory.cc
call_factory.h
call_perf_tests.cc
call_unittest.cc
degraded_call.cc
degraded_call.h
fake_network_pipe.cc
fake_network_pipe.h
fake_network_pipe_unittest.cc
flexfec_receive_stream.cc
flexfec_receive_stream.h
flexfec_receive_stream_impl.cc
flexfec_receive_stream_impl.h
flexfec_receive_stream_unittest.cc
packet_receiver.h
rampup_tests.cc
rampup_tests.h
receive_stream.h
receive_time_calculator.cc
receive_time_calculator.h
receive_time_calculator_unittest.cc
rtp_bitrate_configurator.cc
rtp_bitrate_configurator.h
rtp_bitrate_configurator_unittest.cc
rtp_config.cc
rtp_config.h
rtp_demuxer.cc
rtp_demuxer.h
rtp_demuxer_unittest.cc
rtp_packet_sink_interface.h
rtp_payload_params.cc
rtp_payload_params.h
rtp_payload_params_unittest.cc
rtp_stream_receiver_controller.cc
rtp_stream_receiver_controller.h
rtp_stream_receiver_controller_interface.h
rtp_transport_config.h
rtp_transport_controller_send.cc
rtp_transport_controller_send.h
rtp_transport_controller_send_factory.h
rtp_transport_controller_send_factory_interface.h
rtp_transport_controller_send_interface.h
rtp_video_sender.cc
rtp_video_sender.h
rtp_video_sender_interface.h
rtp_video_sender_unittest.cc
rtx_receive_stream.cc
rtx_receive_stream.h
rtx_receive_stream_unittest.cc
simulated_network.cc
simulated_network.h
simulated_network_unittest.cc
simulated_packet_receiver.h
syncable.cc
syncable.h
version.cc
version.h
video_receive_stream.cc
video_receive_stream.h
video_send_stream.cc
video_send_stream.h
common_audio
common_video
data
docs
examples
experiments
g3doc
infra
logging
media
modules
net/dcsctp
p2p
pc
resources
rtc_base
rtc_tools
sdk
stats
system_wrappers
test
tools_webrtc
video
.clang-format
.git-blame-ignore-revs
.gitignore
.gn
.mailmap
.style.yapf
.vpython
.vpython3
AUTHORS
BUILD.gn
CODE_OF_CONDUCT.md
DEPS
DIR_METADATA
ENG_REVIEW_OWNERS
LICENSE
OWNERS
OWNERS_INFRA
PATENTS
PRESUBMIT.py
README.chromium
README.md
WATCHLISTS
codereview.settings
license_template.txt
native-api.md
presubmit_test.py
presubmit_test_mocks.py
pylintrc
webrtc.gni
webrtc_lib_link_test.cc
whitespace.txt
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adaptation
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2年前
test
Pass rtcp message to RtpTransportController through newer interface
2年前
BUILD.gn
Reland "Remove dependency of video_replay on TestADM."
2年前
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateC…
4年前
OWNERS
Update OWNERS for call/
3年前
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
3年前
audio_receive_stream.h
Propagate time of the last received packet with Timestamp type
2年前
audio_send_stream.cc
Reland "Wire up non-sender RTT for audio, and implement related standa…
4年前
audio_send_stream.h
Replace "rcvd" with "received" for readability
2年前
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstractio…
5年前
audio_state.cc
Remove chromium clang style errors affecting sdk/android/media_jni
7年前
audio_state.h
Async audio processing API
5年前
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
3年前
bitrate_allocator.h
Use backticks not vertical bars to denote variables in comments for /call
4年前
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
3年前
bitrate_estimator_tests.cc
Reland "Remove dependency of video_replay on TestADM."
2年前
call.cc
Allow injecting packets of type Any to Call::DeliverRtpPacket
2年前
call.h
Delete Call dependency on ProcessThread as unused
3年前
call_config.cc
Add parameter to control the pacer's burst outside of field trials.
3年前
call_config.h
Introduce support for video packet batching.
2年前
call_factory.cc
Remove CoDel from webrtc::SimulatedNetwork.
3年前
call_factory.h
Delete Call dependency on ProcessThread as unused
3年前
call_perf_tests.cc
Reland "Remove dependency of video_replay on TestADM."
2年前
call_unittest.cc
Use Timestamp type in RtpState struct
2年前
degraded_call.cc
Migrate PostTask+Wait to BlockingCall to avoid deadlock in DegradedCall.
2年前
degraded_call.h
fake_network_pipe.cc
fake_network_pipe.h
fake_network_pipe_unittest.cc
flexfec_receive_stream.cc
flexfec_receive_stream.h
flexfec_receive_stream_impl.cc
flexfec_receive_stream_impl.h
flexfec_receive_stream_unittest.cc
packet_receiver.h
rampup_tests.cc
rampup_tests.h
receive_stream.h
receive_time_calculator.cc
receive_time_calculator.h
receive_time_calculator_unittest.cc
rtp_bitrate_configurator.cc
rtp_bitrate_configurator.h
rtp_bitrate_configurator_unittest.cc
rtp_config.cc
rtp_config.h
rtp_demuxer.cc
rtp_demuxer.h
rtp_demuxer_unittest.cc
rtp_packet_sink_interface.h
rtp_payload_params.cc
rtp_payload_params.h
rtp_payload_params_unittest.cc
rtp_stream_receiver_controller.cc
rtp_stream_receiver_controller.h
rtp_stream_receiver_controller_interf...
rtp_transport_config.h
rtp_transport_controller_send.cc
rtp_transport_controller_send.h
rtp_transport_controller_send_factory.h
rtp_transport_controller_send_factory...
rtp_transport_controller_send_interfa...
rtp_video_sender.cc
rtp_video_sender.h
rtp_video_sender_interface.h
rtp_video_sender_unittest.cc
rtx_receive_stream.cc
rtx_receive_stream.h
rtx_receive_stream_unittest.cc
simulated_network.cc
simulated_network.h
simulated_network_unittest.cc
simulated_packet_receiver.h
syncable.cc
syncable.h
version.cc
version.h
video_receive_stream.cc
video_receive_stream.h
video_send_stream.cc
video_send_stream.h
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