If a FORWARD-TSN contains an ordered skipped stream with a large TSN
but with a too small SSN, it can result in messages being assembled
that should've been skipped. Typically:
Receive DATA, ordered, complete, TSN=10, SID=1, SSN=0
- will be delivered.
Receive DATA, ordered, complete, TSN=43, SID=1, SSN=7
- will stay in queue, due to missing SSN=1,2,3,4,5,6.
Receive FORWARD-TSN, TSN=44, SSN=6
- is invalid, as the SSN should've been 7 or higher.
However, as the TSN isn't used for removing messages in ordered streams,
but just the SSN, the SSN=7 isn't removed but instead will be delivered
as it's the next following SSN after 6. This will trigger internal
consistency checks as a chunk with TSN=43 will be delivered when the
current cumulative TSN is set to 44, which is greater.
This was found when fuzzing, and can only be provoked by a client that
is intentionally misbehaving. Before this fix, there was no harm done,
but it failed consistency checks which fuzzers have enabled. When
bug 13799 was fixed (in a previous commit), this allowed the fuzzers to
find it faster.
Bug: webrtc:13799
Change-Id: I830ef189476e227e1dbe08157d34f96ad6453e30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36157}
When a FORWARD-TSN is received as the first chunk on an ordered stream,
it will fail to set the new "next expected SSN" that is present in the
FORWARD-TSN as that stream hasn't been allocated yet. It's allocated
when the first DATA is received on that stream.
This is a non-issue for ordinary data channels as the first message on
any stream will be the "Data Channel Establishment Protocol" messages,
which are always sent reliably. But if prenegotiated channels are used,
and the very first packet received on an ordered data channel is lost
_and_ signaled to the receiver as lost _before_ the receiver has
received any other fragments on that data channel, future messages will
not be delivered on that channel.
Bug: webrtc:13799
Change-Id: Ide5c656243b3a51a2ed9d76615cfc3631cfe900c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253902
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36155}
This adds a new paramater to WebRTC-Bwe-EstimateBoundedIncrease that ensure that even if the link capacity has decreased, the delay based estimate does not immediately decrease unless an overuse has been detected.
This is a follow up to https://webrtc-review.googlesource.com/c/src/+/252442/
Bug: none
Change-Id: I98d77ba1e3f7856b06f2691575f2d248a500e659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253901
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36154}
And remove srte since they are no longer active.
Bug: none
Change-Id: I259898db1223d43d13b918ece6555c5f687ce23f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254060
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36153}
When loading the library fails, the user will be faced with this error:
java.lang.UnsatisfiedLinkError: No implementation found for void org.webrtc.PeerConnectionFactory.nativeInitializeAndroidGlobals()
With no context, however.
Bug: webrtc:13619
Change-Id: I88565f085773ad1e8c2f5742d7fdba96fb6043d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253960
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36150}
Additionally,
* Moved to its own GN target.
* Added unittests.
* Removed unused variable `_zeroWallClock`.
* Renamed variables to match style guide.
* Moved fields _dTS and _wrapArounds to variables.
Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36147}
As stashed frames are retried their `tl0_pic_idx` are again unwrapped which can lead to the `tl0_unwrapper_` to unwrap the `tl0_pic_idx` of newer frames backwards. Instead unwrap the `tl0_pid_idx` only once and save it with the frame if necessary.
In this CL
- Only unwrap the TL0 once in ManageFrame.
- Split ManageFrameInternal into ManageFrameFlexible and ManageFrameGof.
- Save the unwrapped TL0 with the stashed frame.
Bug: none
Change-Id: I56e6b071c0082682e010c049c537d66060635567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253844
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36146}
* Uses DataSize to represent incoming and outgoing bytes.
* Puts units into doubles as they enter the Kalman filter
* Moved to its own GN target.
Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36143}
Following https://abseil.io/tips/122 to make tests easier to understand
and adds a bit of flexibility to create sockets with custom parameters.
This also simplifies handover tests.
Additionally, AdvanceTime will now also run timers, as that was easily
forgotten previously.
Bug: None
Change-Id: Ieb5eece7aca51c98a7634ed1c61646383ad1712d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253782
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36141}
WebRTC’s minSdk is 21, so all those checks are dead code.
Change-Id: I26497fd92259b66d9e5ac6afbb393adf4d904c77
Bug: webrtc:13780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253124
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36140}
* Moved into its own GN target
* Switched the internal buffer types to absl::InlinedVector as arrays
are tricky to use with types that do not have default constructors.
* Update fields arnd variables to use style guide.
* Use constexpr for formerly const fields.
* Adds unit tests.
Change-Id: I476ae8491f0f9878c176e7b87a5133942c3d79f7
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36133}
this has been enable by default since M96
BUG=webrtc:11640
Change-Id: I5d310d3929882007211eae12bc3ac1366107ca87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36123}
This change adds a cache for networks in the SimpleNetworkCallback that
is already registered, allowing the cache to be used preferentially as
opposed to the deprecated getAllNetworks call.
This is a fork of https://webrtc-review.googlesource.com/c/src/+/251401
- adds field trials for new behavior
- removes test that did not work
- add (poor) test of field trials
- remove the "network_monitor_java" build target (that I could
not find any reference to...)
Bug: webrtc:13741
Change-Id: I2829c2f1940d4b42455d8e1a2217cf15c133e22b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252284
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36121}
history no longer used for storing unsent packets and for legacy pacer.
Bug: None
Change-Id: I639c37de66857a64c620e80df6288fa6ce8326d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253260
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36120}
- missing negation causes the opposite behavior when
`analog_agc_disable_digital_adaptive` is used
- flag replaced with `analog_agc_use_digital_adaptive_controller`
which is less error-prone
Bug: webrtc:7494
Change-Id: If9e0ba4fc9e539c73269faf9096ca782620dac6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251322
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36113}
This cl/ extends the RTCIceCandidateStats object with
network_adapter_type and vpn, so that it maps the underlying
WebRTC objects completly.
Bug: webrtc:13773
Change-Id: I5cf79972c60ca6bf2a127dc96fa90811263ba6fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36110}
This flag was used as a kill switch in case turning off payload type
demuxing in some Unified Plan cases (https://crbug.com/webrtc/12814)
would cause any issues. That landed way back in M93 and no issues were
ever reported, so time to clean up the flag.
Bug: webrtc:12814
Change-Id: I1970936131384dc0be1cd118e6b0ac877b8c289c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253240
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36109}
