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webrtc_m130/call
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Ali Tofigh 641a1b11b6 Adopt absl::string_view in call/ 
Bug: webrtc:13579
Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36910}
2022-05-17 12:00:45 +00:00
..
adaptation
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
test
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
audio_receive_stream.cc
…
audio_receive_stream.h
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +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
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
BUILD.gn
Adopt absl::string_view in call/
2022-05-17 12:00:45 +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
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
call_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
call.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
call.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
degraded_call.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
degraded_call.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +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
Reland: Update local_ssrc without needing to recreate video streams.
2022-05-17 10:59:54 +00:00
flexfec_receive_stream_impl.h
Reland: Update local_ssrc without needing to recreate video streams.
2022-05-17 10:59:54 +00:00
flexfec_receive_stream_unittest.cc
Deduplicate set of the rtp header extension uri constants
2021-09-14 13:38:44 +00:00
flexfec_receive_stream.cc
…
flexfec_receive_stream.h
Add a transport_cc() getter and remove rtp_config().
2022-05-09 20:21:14 +00:00
OWNERS
Fix WebRTC version updater bot.
2021-11-22 08:44:15 +00:00
packet_receiver.h
Remove DeliverPacketAsync.
2021-05-29 07:37:33 +00:00
rampup_tests.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rampup_tests.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
receive_stream.h
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +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
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
rtp_payload_params.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_payload_params.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +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
Adopt absl::string_view in call/
2022-05-17 12:00:45 +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
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
rtp_video_sender.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_video_sender.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +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
…
syncable.cc
…
syncable.h
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
version.cc
Update WebRTC code version (2022-05-17T04:04:40).
2022-05-17 05:17:46 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Prepare WebRtcVideoReceiveStream for configuration changes.
2021-07-01 11:23:51 +00:00
video_receive_stream.h
Add a transport_cc() getter and remove rtp_config().
2022-05-09 20:21:14 +00:00
video_send_stream.cc
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00
video_send_stream.h
Remove more top-level const from parameters in function declarations
2022-02-01 09:15:50 +00:00