zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
640b51eeae487ff242bc6e982f04414d7c1c9905
platform-external-webrtc/call
History
Danil Chapovalov 5b298ab9dd For VP9 assume max number of spatial layers to simulate generic descriptor 
VP9 allows to increase number of spatial layers on delta frame, which
is not supported by dependency descriptor.
Thus to generate DD compatible generic header, simulator would set max
number of spatial layers, while number of active spatial layers would be
communicated with active_decode_target bitmask

Bug: webrtc:14042
Change-Id: I4da63fa7c38b0f17758a7a6243640f444470b40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265164
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37151}
2022-06-08 11:36:54 +00:00
..
adaptation
Delete TestListener and top-level thread wrapping.
2022-05-20 15:21:21 +00:00
test
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Add SetTransportCc to ReceiveStreamInterface.
2022-05-30 14:07:04 +00:00
audio_send_stream.cc
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
audio_send_stream.h
Remove typing detection
2022-03-23 10:23:54 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_state.cc
…
audio_state.h
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
2022-03-09 13:23:21 +00:00
bitrate_allocator.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
bitrate_estimator_tests.cc
Rename VideoReceiveStream to VideoReceiveStreamInterface
2022-05-22 10:54:38 +00:00
BUILD.gn
Delete TestListener and top-level thread wrapping.
2022-05-20 15:21:21 +00:00
call_config.cc
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
2021-06-01 06:57:31 +00:00
call_config.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
call_factory.cc
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
call_factory.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
call_perf_tests.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
call_unittest.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
call.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
call.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
degraded_call.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
degraded_call.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
fake_network_pipe_unittest.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
fake_network_pipe.cc
Delete some unneeded references to ProcessThread.
2022-01-03 15:36:02 +00:00
fake_network_pipe.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +00:00
flexfec_receive_stream_impl.cc
[FlexfecReceiveStream] Use explicit member variables for state.
2022-05-30 07:37:03 +00:00
flexfec_receive_stream_impl.h
Add SetTransportCc to ReceiveStreamInterface.
2022-05-30 14:07:04 +00:00
flexfec_receive_stream_unittest.cc
Delete TestListener and top-level thread wrapping.
2022-05-20 15:21:21 +00:00
flexfec_receive_stream.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h
Remove unused FlexfecReceiveStream::Stats struct
2022-05-25 07:02:39 +00:00
OWNERS
Update OWNERS for call/
2022-06-03 12:01:46 +00:00
packet_receiver.h
Remove DeliverPacketAsync.
2021-05-29 07:37:33 +00:00
rampup_tests.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
rampup_tests.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
receive_stream.h
Add SetTransportCc to ReceiveStreamInterface.
2022-05-30 14:07:04 +00:00
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
receive_time_calculator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +00:00
rtp_config.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
rtp_config.h
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
rtp_demuxer_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_payload_params.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_payload_params.h
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_stream_receiver_controller.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
rtp_transport_config.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_transport_controller_send_factory_interface.h
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
rtp_transport_controller_send_factory.h
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
rtp_transport_controller_send_interface.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_transport_controller_send.cc
Use ProbeClusterConfig in BitrateProber from GoogCC
2022-05-17 12:29:25 +00:00
rtp_transport_controller_send.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_video_sender_interface.h
Remove top-level const from parameters in function declarations.
2022-01-26 11:05:25 +00:00
rtp_video_sender_unittest.cc
Simulate generic dependency structure for VP8
2022-05-19 11:53:08 +00:00
rtp_video_sender.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_video_sender.h
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtx_receive_stream_unittest.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtx_receive_stream.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtx_receive_stream.h
IWYU: uint32_t is defined in cstdint
2020-05-07 17:04:15 +00:00
simulated_network_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
simulated_network.cc
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
simulated_network.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
simulated_packet_receiver.h
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
syncable.cc
…
syncable.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2022-06-08T04:01:49).
2022-06-08 05:36:22 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Remove unused VideoReceiveStreamInterface::Config::target_delay_ms field.
2022-05-30 09:30:23 +00:00
video_receive_stream.h
Remove deprecated VideoReceiveStream alias
2022-06-01 10:17:55 +00:00
video_send_stream.cc
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00
video_send_stream.h
RtpSenderInterface::SetEncoderSelector
2022-06-08 11:06:52 +00:00