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webrtc/audio
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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
..
test Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
utility Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
voip Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
audio_level.cc Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +00:00
audio_level.h Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +00:00
audio_receive_stream.cc Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
audio_receive_stream.h Add SetTransportCc to ReceiveStreamInterface. 2022-05-30 14:07:04 +00:00
audio_receive_stream_unittest.cc Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
audio_send_stream.cc Update audio/, media/, and video/ to not use implicit conversion 2022-04-21 09:00:14 +00:00
audio_send_stream.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
audio_send_stream_tests.cc Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
audio_send_stream_unittest.cc Remove rtc::Location from SendTask test helper 2022-08-11 12:55:32 +00:00
audio_state.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
audio_state.h Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
audio_state_unittest.cc Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
audio_transport_impl.cc Reland "Reland "Remove unused APM voice activity detection sub-module"" 2022-02-16 08:41:30 +00:00
audio_transport_impl.h Remove typing detection 2022-03-23 10:23:54 +00:00
BUILD.gn Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
channel_receive.cc Allow recursive check for RTC_DCHECK_RUN_ON macro 2022-07-26 09:27:23 +00:00
channel_receive.h Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
channel_receive_frame_transformer_delegate.cc Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
channel_receive_frame_transformer_delegate.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
channel_send.cc Make ChannelSend::OnUplinkPacketLossRate public 2022-07-18 13:42:01 +00:00
channel_send.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
channel_send_frame_transformer_delegate.cc Update audio code to not use implicit T* --> scoped_refptr<T> conversion 2022-01-13 15:49:49 +00:00
channel_send_frame_transformer_delegate.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
channel_send_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
conversion.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Cleanup of bwe_defines.h 2020-11-26 12:26:02 +00:00
mock_voe_channel_proxy.h Implement RTCOutboundRtpStreamStats.targetBitrate for audio. 2021-11-12 09:24:34 +00:00
null_audio_poller.cc Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
null_audio_poller.h Use SequenceChecker from public API 2021-02-10 15:04:55 +00:00
OWNERS Add jakobi@webrtc.org to audio/OWNERS 2021-08-19 15:30:32 +00:00
remix_resample.cc Reland "Rename FATAL() into RTC_FATAL()." 2020-11-18 20:49:08 +00:00
remix_resample.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
remix_resample_unittest.cc Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00