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webrtc/audio
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Guy Hershenbaum f009e38fe0 Fix AudioSendStream reconfigure - stop processing during unconfigured state 
When Reconfiguring there's a call to ResetSenderCongestionControlObjects followed by a later call to
RegisterSenderCongestionControlObjects which happen on the worker thread, while enqueuing packets is
happening on a different thread.
If packets are enqueued in between these calls we get a null dereference of the `rtp_packet_pacer_`
This change fixes it by pausing processing of incoming audio in the interim state

Bug: webrtc:358290775
Change-Id: I77cebfb131545ce2a6fdb26105dd999da3f7c443
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42815}
2024-08-20 16:22:04 +00:00
..
test Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
utility Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch 2024-07-23 13:23:26 +00:00
voip Pass Environment instead of just clock to AcmReceiver at construction 2024-08-06 08:28:23 +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 Pass Clock and RtcEventLog as Environment into AudioReceiveStream 2024-08-02 11:58:23 +00:00
audio_receive_stream.h Pass Clock and RtcEventLog as Environment into AudioReceiveStream 2024-08-02 11:58:23 +00:00
audio_receive_stream_unittest.cc Pass Clock and RtcEventLog as Environment into AudioReceiveStream 2024-08-02 11:58:23 +00:00
audio_send_stream.cc Add accounting of actual audio bit usage 2024-08-06 18:04:46 +00:00
audio_send_stream.h Add variables to lend unused audio bits to video 2024-07-30 18:42:16 +00:00
audio_send_stream_tests.cc Expose AudioLevel as an absl::optional struct in api/rtp_headers.h 2024-03-22 10:07:47 +00:00
audio_send_stream_unittest.cc Add accounting of actual audio bit usage 2024-08-06 18:04:46 +00:00
audio_state.cc Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
audio_state.h Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +00:00
audio_state_unittest.cc Implement support for Chrome task origin tracing. #3.5/4 2023-03-01 11:11:37 +00:00
audio_transport_impl.cc Remove PushResampler<T>::InitializeIfNeeded 2024-07-04 10:33:21 +00:00
audio_transport_impl.h Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
BUILD.gn Pass Clock and RtcEventLog as Environment into AudioReceiveStream 2024-08-02 11:58:23 +00:00
channel_receive.cc Pass Environment instead of just clock to AcmReceiver at construction 2024-08-06 08:28:23 +00:00
channel_receive.h Pass Clock and RtcEventLog as Environment into AudioReceiveStream 2024-08-02 11:58:23 +00:00
channel_receive_frame_transformer_delegate.cc Pass receive_time through frame transformer 2024-08-02 07:01:33 +00:00
channel_receive_frame_transformer_delegate.h Pass receive_time through frame transformer 2024-08-02 07:01:33 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Pass receive_time through frame transformer 2024-08-02 07:01:33 +00:00
channel_receive_unittest.cc Pass Clock and RtcEventLog as Environment into AudioReceiveStream 2024-08-02 11:58:23 +00:00
channel_send.cc Fix AudioSendStream reconfigure - stop processing during unconfigured state 2024-08-20 16:22:04 +00:00
channel_send.h Add accounting of actual audio bit usage 2024-08-06 18:04:46 +00:00
channel_send_frame_transformer_delegate.cc Pass receive_time through frame transformer 2024-08-02 07:01:33 +00:00
channel_send_frame_transformer_delegate.h Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +00:00
channel_send_frame_transformer_delegate_unittest.cc Reland "Run IWYU on some files I intend to work on" 2024-06-05 08:59:49 +00:00
channel_send_unittest.cc Fix AudioSendStream reconfigure - stop processing during unconfigured state 2024-08-20 16:22:04 +00:00
conversion.h Make header files self contained. 2022-10-08 08:38:36 +00:00
DEPS pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
mock_voe_channel_proxy.h Add accounting of actual audio bit usage 2024-08-06 18:04:46 +00:00
OWNERS Add alessiob@webrtc.org in audio/OWNERS 2022-09-09 07:33:11 +00:00
remix_resample.cc Remove PushResampler<T>::InitializeIfNeeded 2024-07-04 10:33:21 +00:00
remix_resample.h Update RemixAndResample to use audio views 2024-07-03 09:52:24 +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