When the slacked pacer experiment is enabled the next pacing opportunity
may be a full tick (~16 ms) from now. Add a flag to allow experimenting
with a burst interval (= 16 ms?) such that we can send bursts in
MaybeProcessPackets.
A common use case would be that EnqueuePackets triggers
MaybeProcessPackets when we are off-tick but we'd still like to create
an immediate burst instead of waiting for the next tick or two for that
to happen.
Bug: webrtc:14152
Change-Id: Ib0ed8312cb7d53b80f3520fff3a6e3bbb5a93fd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264985
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37116}
This is a re-land of commit 3180a5ad0663900a39adf4b9974052c356c835fe.
This is an issue found in fuzzer, and doesn't really happen in WebRTC
as it never closes the connection and reconnects.
The issue is that the send queue outlives any connection since you're
allowed to send messages (well, enqueue them) before the association is
fully connected. So the send queue is always present but the TCB
(information about the connection) is torn down when the connection is
closed for example. And the TCB holds the Stream Reset handler, which is
responsible for e.g. keeping track of stream reset sequence numbers and
such - which is tied to the connection.
So to ensure that the Stream Reset Handler is in charge of deciding
if a stream reset is taking place, make sure that the send queue is in
a known good state when the Stream Reset handler is created.
Bug: webrtc:13994, chromium:1320194
Change-Id: Ib8254488523c7abb58057c602f76f411fce896fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265000
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37115}
APM has historically allowed sample rates not divisible by 100, but there is also code that explicitly states that such rates are not supported.
It is unclear how well rates like 22050 are handled in practice.
This CL adds support for fuzzing more sample rates, to help find issues.
We usually preserve fuzzer data reads to avoid invalidating unresolved fuzzer-found issues, but to make the code a little more readable this CL removes the discarded reads. This renders the only currently open bug non-reproducible, crbug.com/1299393.
Bug: webrtc:9413, chromium:1299393
Change-Id: I98ac1c653627c20adc73b8edede02f1526d80d9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264504
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37114}
This is verifying the theory that the fix on bug 12592 also fixed
bug 3608.
Bug: webrtc:3608, webrtc:12592
Change-Id: Ia1f5ba5ebdc9a839034092351c970c3b6a159fa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264829
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37113}
This changes renames the event to better suit its purpose. An alias
with the old name is added for compatibility pending internal cleanup.
Bug: webrtc:14125, webrtc:14131
Change-Id: I87026e19f2620eaa6a6770dcbedf1d0399c6c6b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264149
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#37111}
The current code assumed that chunks that were scheduled for fast
retransmission would never be abandoned, as chunks marked for fast
retransmission would be immediately sent after the SACK has been
processed, giving no time for them to be abandoned.
But fuzzers keep on fuzzing, and can craft a sequence of chunks that
result in a SACK that both marks the chunks for fast retransmission and
later (while processing the same SACK) abandons them.
Bug: chromium:1331087
Change-Id: Id218607e18a6f3a9d6d51044dccb920e1e77372a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264960
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37108}
This is a partial reland of commit 794e68cf3dbc3b16ee8b12b5615d8a1622154dbd
which uses the new enum in P2PTransportChannel and BasicIceController.
Original change's description:
> Refactor IceControllerEvent
>
> This change is the first step in decoupling IceControllerEvent from
> the ICE switch reason. Further cleanup is earmarked, and will be
> landed after some internal cleanup.
>
> This change
> - adds a new enum - IceSwitchReason
> - adds a member for the new enum in IceControllerEvent
> - uses the new enum within P2PTransportChannel
> - adds methods to IceControllerInterface accepting the new enum
> - deprecates usages of the old enum in IceControllerInterface
> - adds conversion between the old and new enums for compatibility
>
> Bug: webrtc:14125, webrtc:14131
> Change-Id: I5b7201c3f631eb40db334dfeec842841a7e58174
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264140
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Sameer Vijaykar <samvi@google.com>
> Cr-Commit-Position: refs/heads/main@{#37051}
Bug: webrtc:14125, webrtc:14131
Change-Id: I81b0338ae2f0560cd3df7ad9bd9af37e7c3499df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264554
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#37105}
Allows the PacerController to send packets in bursts. If there are enqued packets, or a packet is enqueued while the pacer have a small media debt, an enqued packet is allowed to be sent immediately as long as the debt is smaller than the set burst interval.
Bug: b/233850913
Change-Id: Ibb0fa63c97409ca23b9fa7148b5ff6ce8c4517e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264462
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37098}
Adds a function to PeerConnectionIntegrationBaseTest to stop and destroy
the caller and callee objects. This should take care of dangling pointers.
Before this change, the affected test would crash randomly - typically
detected within a few minutes of a gtest-repeat=-1 run.
After this change, it has not crashed in 15 minutes of running.
Bug: webrtc:12592
Change-Id: I9980f8974015bf2b2104fcb83c2ca0d677d03c3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264555
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37096}
This is a partial reland of commit 794e68cf3dbc3b16ee8b12b5615d8a1622154dbd
which
- adds a new enum - IceSwitchReason
- adds a member for the new enum in IceControllerEvent
- adds methods to IceControllerInterface accepting the new enum
- adds conversion between the old and new enums for compatibility
Original change's description:
> Refactor IceControllerEvent
>
> This change is the first step in decoupling IceControllerEvent from
> the ICE switch reason. Further cleanup is earmarked, and will be
> landed after some internal cleanup.
>
> This change
> - adds a new enum - IceSwitchReason
> - adds a member for the new enum in IceControllerEvent
> - uses the new enum within P2PTransportChannel
> - adds methods to IceControllerInterface accepting the new enum
> - deprecates usages of the old enum in IceControllerInterface
> - adds conversion between the old and new enums for compatibility
>
> Bug: webrtc:14125, webrtc:14131
> Change-Id: I5b7201c3f631eb40db334dfeec842841a7e58174
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264140
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Sameer Vijaykar <samvi@google.com>
> Cr-Commit-Position: refs/heads/main@{#37051}
Bug: webrtc:14125, webrtc:14131
Change-Id: I2ccdd53c80e38dc139669aa3f438864befed3dc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264506
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37094}
Move warning about missing receive_statistics to AddReceiver to avoid
producing it for rtp send only endpoints.
Remove warning about missing cname as unimportant.
Bug: webrtc:8239
Change-Id: I8a90aa4b378284b9c68f67678b2392b9461c95b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264825
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37093}
FrameBuffer3Proxy and sync decoding has been shown to work. First step of cleaning up is to remove the FrameBuffer2Proxy.
Change-Id: Ic96303c2d4f9111cfeed9927e8826ea7ffe7ee17
Bug: webrtc:14003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264126
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37086}
This is a reland of commit 17a02a31d7d2897b75ad69fdac5d10e7475a5865.
This is the first part of supporting stream priorities, and adds the API
and very basic support for setting and retrieving the stream priority.
This commit doesn't in any way change the actual packet sending - the
specified priority values are stored, but not acted on.
This is all that is client visible, so clients can start using the API
as written, and they would never notice that things are missing.
Bug: webrtc:5696
Change-Id: I04d64a63cbaec67568496ad99667e14eba85f2e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264424
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37081}
* Add ctors for providing the type and transaction id at construction.
* Update tests to use them instead of SetType+SetTransactionID
* Make sure stun message enum types are based on uint16_t
* Mark SetTransactionID as deprecated.
* Mark SetStunMagicCookie as deprecated (unused in webrtc).
* Add SetTransactionIdForTest for the one test that uses it (might not
actually need it)
* Make StunRequest::Construct() protected.
* Add a TODO to follow up on this since construction of StunRequest
goes through an unnecessarily complex 3-step process involving
other classes and a virtual method.
Bug: none
Change-Id: Ib013e58f28e7b2b4fcb3b3e1034da31dfc93e9d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264546
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37079}
BaseTime represents fixed point in time with unknown epoch and thus
make sense to convert to Timestamp type, however Timestamp should always
be positive. however legacy tests expect GetBaseTimeUs to return negative time sometimes.
Bug: webrtc:13757
Change-Id: I3f780a7775fdd1e271402c59384c1298db76f75a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264549
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37076}
