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webrtc/call
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Danil Chapovalov e519f38eaa Remove rtc::Location from SendTask test helper 
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.

Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
2022-08-11 12:55:32 +00:00
..
adaptation Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +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 Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +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 Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_state.h Async audio processing API 2020-10-02 12:33:34 +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_allocator_unittest.cc Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
bitrate_estimator_tests.cc Remove rtc::Location from SendTask test helper 2022-08-11 12:55:32 +00:00
BUILD.gn Migrate call/ to absl::AnyInvocable based TaskQueueBase interface 2022-07-07 10:32:26 +00:00
call.cc Fix Event Log For Video Receiver 2022-08-11 12:15:52 +00:00
call.h Delete Call dependency on ProcessThread as unused 2022-06-21 08:59:38 +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 WebRTCFakeNetwork field trials. 2022-07-27 07:33:15 +00:00
call_factory.h Delete Call dependency on ProcessThread as unused 2022-06-21 08:59:38 +00:00
call_perf_tests.cc Remove rtc::Location from SendTask test helper 2022-08-11 12:55:32 +00:00
call_unittest.cc Delete Call dependency on ProcessThread as unused 2022-06-21 08:59:38 +00:00
degraded_call.cc Migrate call/ to absl::AnyInvocable based TaskQueueBase interface 2022-07-07 10:32:26 +00:00
degraded_call.h Move to_queued_task.h and pending_task_safety_flag.h into public API 2022-06-17 09:20:39 +00:00
DEPS SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. 2021-08-30 10:20:55 +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
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
flexfec_receive_stream.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h Add SetPayloadType to FlexfecReceiveStream. 2022-07-28 21:24:50 +00:00
flexfec_receive_stream_impl.cc Add SetPayloadType to FlexfecReceiveStream. 2022-07-28 21:24:50 +00:00
flexfec_receive_stream_impl.h Add SetPayloadType to FlexfecReceiveStream. 2022-07-28 21:24:50 +00:00
flexfec_receive_stream_unittest.cc test: fix flexfec test 2022-07-06 10:37:19 +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 Remove rtc::Location from SendTask test helper 2022-08-11 12:55:32 +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.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
receive_time_calculator_unittest.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 2022-03-09 22:17:52 +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_bitrate_configurator_unittest.cc Revert "In RtpBitrateConfigurator ignore new parameters when set to default values." 2020-01-10 16:39:51 +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 Update old TODO comments 2022-07-05 09:59:33 +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_demuxer_unittest.cc Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
rtp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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_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_stream_receiver_controller.cc Demote RtpStreamReceiverController AddSink/RemoveSink to private 2022-07-06 09:31:54 +00:00
rtp_stream_receiver_controller.h Demote RtpStreamReceiverController AddSink/RemoveSink to private 2022-07-06 09:31:54 +00:00
rtp_stream_receiver_controller_interface.h Demote RtpStreamReceiverController AddSink/RemoveSink to private 2022-07-06 09:31:54 +00:00
rtp_transport_config.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_transport_controller_send.cc Remove unused field trial kill-switch WebRTC-LazyPacerStart. 2022-07-07 11:21:55 +00:00
rtp_transport_controller_send.h Update/delete old TODOs 2022-07-06 07:49:04 +00:00
rtp_transport_controller_send_factory.h Remove legacy PacedSender. 2022-05-13 20:31:06 +00:00
rtp_transport_controller_send_factory_interface.h Delete Call dependency on ProcessThread as unused 2022-06-21 08:59:38 +00:00
rtp_transport_controller_send_interface.h Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
rtp_video_sender.cc Drop frames in RtpVideoSender::OnEncodedImage if stream disabled 2022-08-11 08:40:01 +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
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 Drop frames in RtpVideoSender::OnEncodedImage if stream disabled 2022-08-11 08:40:01 +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
rtx_receive_stream_unittest.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +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_network_unittest.cc clean up misc TimeDelta use 2022-08-02 13:52:36 +00:00
simulated_packet_receiver.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
version.cc Update WebRTC code version (2022-08-11T04:04:56). 2022-08-11 06:47:01 +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 Make nack history configurable. 2022-08-05 22:58:43 +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