Update iOS deployment target to 11.0 #10

jim-signal · 2020-06-22T23:36:33Z

No description provided.

…or vp9 1) Fix several typos and small mistakes which could lead to crashes 2) Adjust bitrates if leading layers are disabled 3) Wire up webrtc quality scaler (cherry picked from commit 7a82467) Bug: webrtc:11319 Change-Id: I16e52bdb1c315d64906288e4f2be55fe698d5ceb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177525 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#31546} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177660 Cr-Commit-Position: refs/branch-heads/4147@{#10} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}

This CL corrects the temporary buffers size in the pre-processing of the capture audio before encoding. As part of this it removes the ACM-specific hardcoding of the size and instead ensures that the size of the temporary buffer matches that of the AudioFrame. (cherry picked from commit d82a02c) No-Try: True TBR: kwiberg@webrtc.org Bug: webrtc:11242, chromium:1060647 Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30775} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/branch-heads/4044@{#10} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}

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}

The new version of MSan (rolled by [1]) detects the following: ``` ==39908==WARNING: MemorySanitizer: use-of-uninitialized-value #0 0x5591400a52ef in GetPlayoutDelayMs ./../../modules/audio_coding/neteq/decision_logic.cc:466:35 #1 0x5591400a52ef in webrtc::DecisionLogic::ExpectedPacketAvailable(webrtc::NetEqController::NetEqStatus) ./../../modules/audio_coding/neteq/decision_logic.cc:311:36 #2 0x5591400a39e9 in webrtc::DecisionLogic::GetDecision(webrtc::NetEqController::NetEqStatus const&, bool*) ./../../modules/audio_coding/neteq/decision_logic.cc:0:0 #3 0x55913cf590c9 in webrtc::DecisionLogicTest_PreemptiveExpand_Test::TestBody() ./../../modules/audio_coding/neteq/decision_logic_unittest.cc:139:3 #4 0x55913ef28283 in HandleExceptionsInMethodIfSupported<testing::Test, void> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:3 #5 0x55913ef28283 in testing::Test::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2710:5 #6 0x55913ef2ab46 in testing::TestInfo::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2856:11 #7 0x55913ef2da34 in testing::TestSuite::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:3034:30 #8 0x55913ef621e8 in testing::internal::UnitTestImpl::RunAllTests() ./../../third_party/googletest/src/googletest/src/gtest.cc:5964:44 #9 0x55913ef60f54 in HandleExceptionsInMethodIfSupported<testing::internal::UnitTestImpl, bool> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:0 #10 0x55913ef60f54 in testing::UnitTest::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:5543:10 #11 0x55913ee1a944 in RUN_ALL_TESTS ./../../third_party/googletest/src/googletest/include/gtest/gtest.h:2334:73 #12 0x55913ee1a944 in webrtc::(anonymous namespace)::TestMainImpl::Run(int, char**) ./../../test/test_main_lib.cc:203:21 #13 0x55913cbd36b8 in main ./../../test/test_main.cc:72:16 #14 0x7fdb18c73082 in __libc_start_main /build/glibc-LcI20x/glibc-2.31/csu/../csu/libc-start.c:308:16 #15 0x55913cb3e1a9 in _start ??:0:0 ``` [1] - https://webrtc-review.googlesource.com/c/src/+/353620 Bug: b/344970813 Change-Id: I9b5d7791e68b4c494168ba9f007a3099ae21fed4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353581 Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42433}

Update iOS deployment target to 11.0

43a43b7

jim-signal requested a review from peter-signal June 22, 2020 23:36

peter-signal approved these changes Jun 22, 2020

View reviewed changes

jim-signal merged commit a8c6be0 into master Jun 22, 2020

jim-signal deleted the jim/update-ios-target branch June 22, 2020 23:39

Provide feedback

Saved searches

Use saved searches to filter your results more quickly

Update iOS deployment target to 11.0 #10

Update iOS deployment target to 11.0 #10

jim-signal commented Jun 22, 2020

Update iOS deployment target to 11.0 #10

Update iOS deployment target to 11.0 #10

Conversation

jim-signal commented Jun 22, 2020