Per K b60f0ffbce Dont signal ReadyToSend in RtpTransport::SendPacket
Before this cl, ReadyToSend signaled false if sending a packet failed and transport->GetError() returns ECONN.
ECONN may be reported by the TCP connection (TcpConnection) if the remote closed the connection. TcpConnection will attempt to reconnect and should change the writable state if it fail.
Changing the state in the context of sending packets may cause recursive
calls and seems to cause problems with incorrect states.
It is simpler if RtpTransport::SendPacket ignore these failures and
upper layers treat these lost packets similar to if the packets had been
lost via UDP.
For safety, this change can be reverted by field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/.

Bug: webrtc:361124449 b/359989715
Change-Id: I8e7016dfb4301862286215c4512aa8ac03a16685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42868}
2024-08-27 14:16:53 +00:00
2024-08-27 13:50:35 +00:00
2024-07-09 10:30:26 +00:00
2024-07-11 20:26:16 +00:00
2024-08-27 13:50:35 +00:00
2024-08-22 10:37:00 +00:00
2023-09-25 15:56:09 +00:00
2024-05-27 12:46:57 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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