Fix simulcast using hardware encoder on Android #7
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kanat
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kanat
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Apr 5, 2024
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io>
kanat
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Apr 5, 2024
* Audio Device Optimization allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io> * fix compilation errors --------- Co-authored-by: CloudWebRTC <duanweiwei1982@gmail.com> Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io>
santhoshvai
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Nov 20, 2024
This was a fun bug which proved to be challenging to find a good solution for. The issue comes from the combination of partial reliability and stream resetting, which are covered in different RFCs, and where they don't refer to each other... Stream resetting (RFC 6525) is used in WebRTC for closing a Data Channel, and is done by signaling to the receiver that the stream sequence number (SSN) should be set to zero (0) at some time. Partial reliability (RFC 3758) - and expiring messages that will not be retransmitted - is done by signaling that the SSN should be set to a certain value at a certain TSN, as the messages up until the provided SSN are not to be expected to be sent again. As these two functionalities both work by signaling to the receiver what the next expected SSN should be, they need to do it correctly not to overwrite each others' intent. And here was the bug. An example scenario where this caused issues, where we are Z (the receiver), getting packets from the sender (A): 5 A->Z DATA (TSN=30, B, SID=2, SSN=0) 6 Z->A SACK (Ack=30) 7 A->Z DATA (TSN=31, E, SID=2, SSN=0) 8 A->Z RE_CONFIG (REQ=30, TSN=31, SID=2) 9 Z->A RE_CONFIG (RESP=30, Performed) 10 Z->A SACK (Ack=31) 11 A->Z DATA (TSN=32, SID=1) 12 A->Z FORWARD_TSN (TSN=32, SID=2, SSN=0) Let's assume that the path Z->A had packet loss and A never really received our responses (#6, #9, #10) in time. At #5, Z receives a DATA fragment, which it acks, and at #7 the end of that message. The stream is then reset (#8) which it signals that it was performed (#9) and acked (#10), and data on another stream (2) was received (#11). Since A hasn't received any ACKS yet, and those chunks on SID=2 all expired, A sends a FORWARD-TSN saying that "Skip to TSN=32, and don't expect SID=2, SSN=0". That makes the receiver expect the SSN on SID=2 to be SSN=1 next time at TSN=32. But that's not good at all - A reset the stream at #8 and will want to send the next message on SID=2 using SSN=0 - not 1. The FORWARD-TSN clearly can't have a TSN that is beyond the stream reset TSN for that stream. This is just one example - combining stream resetting and partial reliability, together with a lossy network, and different variants of this can occur, which results in the receiver possibly not delivering packets because it expects a different SSN than the one the sender is later using. So this CL adds "breakpoints" to how far a FORWARD-TSN can stretch. It will simply not cross any Stream Reset last assigned TSNs, and only when a receiver has acked that all TSNs up till the Stream Reset last assigned TSN has been received, it will proceed expiring chunks after that. Bug: webrtc:14600 Change-Id: Ibae8c9308f5dfe8d734377d42cce653e69e95731 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321600 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40829}
santhoshvai
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Nov 20, 2024
* Audio Device Optimization allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io> * fix compilation errors --------- Co-authored-by: CloudWebRTC <duanweiwei1982@gmail.com> Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io>
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Patch origin:
https://github.com/webrtc-sdk/webrtc/pull/48.patch