webrtc/audio
Artem Titov a208861401 Reland "Fix data race for config_ in AudioSendStream"
This is a reland of 51e5c4b0f4

It may happen that user will pass config with min bitrate > max bitrate.
In such case we can't generate cached_constraints and will crash before.
The reland will handle this situation gracefully.

Original change's description:
> Fix data race for config_ in AudioSendStream
>
> config_ was written and read on different threads without sync. This CL
> moves config access on worker_thread_ with all other required fields.
> It keeps only bitrate allocator accessed from worker_queue_, because
> it is used from it in other classes and supposed to be single threaded.
>
> Bug: None
> Change-Id: I23ece4dc8b09b41a8c589412bedd36d63b76cbc5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203267
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33125}

Bug: None
Change-Id: I274ff15208d69c25fb25a0f1dd0a0e37b72480b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33162}
2021-02-04 12:33:56 +00:00
..
test Default enable sending transport sequence numbers on audio packets. 2020-11-24 09:19:54 +00:00
utility Rename several more tests that use EXPECT_DEATH to *DeathTest. 2020-05-18 16:10:04 +00:00
voip Replace rtc::ThreadChecker with webrtc::SequenceChecker 2021-02-02 14:56:27 +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 Reland "Prepare to avoid hops to worker for network events." 2021-02-03 17:44:47 +00:00
audio_receive_stream.h Replace rtc::ThreadChecker with webrtc::SequenceChecker 2021-02-02 14:56:27 +00:00
audio_receive_stream_unittest.cc Remove nesting of Naggy/Strict/NiceMock 2020-12-07 08:19:50 +00:00
audio_send_stream.cc Reland "Fix data race for config_ in AudioSendStream" 2021-02-04 12:33:56 +00:00
audio_send_stream.h Reland "Fix data race for config_ in AudioSendStream" 2021-02-04 12:33:56 +00:00
audio_send_stream_tests.cc Default enable sending transport sequence numbers on audio packets. 2020-11-24 09:19:54 +00:00
audio_send_stream_unittest.cc Remove nesting of Naggy/Strict/NiceMock 2020-12-07 08:19:50 +00:00
audio_state.cc Async audio processing API 2020-10-02 12:33:34 +00:00
audio_state.h Replace rtc::ThreadChecker with webrtc::SequenceChecker 2021-02-02 14:56:27 +00:00
audio_state_unittest.cc Async audio processing API 2020-10-02 12:33:34 +00:00
audio_transport_impl.cc Async audio processing API 2020-10-02 12:33:34 +00:00
audio_transport_impl.h Async audio processing API 2020-10-02 12:33:34 +00:00
BUILD.gn Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
channel_receive.cc Reland "Prepare to avoid hops to worker for network events." 2021-02-03 17:44:47 +00:00
channel_receive.h Fix standard GetStats to not modify NetEq state. 2020-09-14 09:51:21 +00:00
channel_receive_frame_transformer_delegate.cc Transform received audio frames in ChannelReceive. 2020-04-01 11:23:00 +00:00
channel_receive_frame_transformer_delegate.h Reland "Introduce RTC_NO_UNIQUE_ADDRESS." 2020-11-23 11:29:36 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Add unit tests for audio receive channel frame transformer delegate. 2020-05-04 15:44:08 +00:00
channel_send.cc Replace rtc::ThreadChecker with webrtc::SequenceChecker 2021-02-02 14:56:27 +00:00
channel_send.h Removes locking in TransportFeedbackProxy. 2020-06-29 16:52:34 +00:00
channel_send_frame_transformer_delegate.cc Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +00:00
channel_send_frame_transformer_delegate.h Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +00:00
channel_send_frame_transformer_delegate_unittest.cc Add unit tests for audio channel send frame transformer delegate. 2020-05-04 16:50:12 +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 Fix standard GetStats to not modify NetEq state. 2020-09-14 09:51:21 +00:00
null_audio_poller.cc Make MessageHandler be a pure virtual interface. 2020-09-25 11:44:02 +00:00
null_audio_poller.h Replace rtc::ThreadChecker with webrtc::SequenceChecker 2021-02-02 14:56:27 +00:00
OWNERS Remove wildcard ownership for build files. 2020-02-19 14:05:46 +00:00
remix_resample.cc Reland "Rename FATAL() into RTC_FATAL()." 2020-11-18 20:49:08 +00:00
remix_resample.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
remix_resample_unittest.cc Add RTC_ prefix to non-standard format specifier macro "PRIdNS" 2019-08-07 13:36:05 +00:00