video_stream_decoder2 is the only one used.
Change-Id: Iabee3521b2946f097296cf2b02025aa6e41e87a4
Bug: webrtc:11489
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260282
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36692}
This is achieved by wrapping a fake decoder inside the mock decoder, in
a sort of spy pattern.
This is preperation for moving the FrameBufferProxy tests into the main
VideoReceiveStream2 suite.
Bug: webrtc:14003
Change-Id: I7b9691cc5a1a8a3fadfb7aa6981752b647d5c73f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260113
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36691}
...and add to TCPConnection as it's still needed there.
Bug: webrtc:11943
Change-Id: Id66b890677d7e0b03a4a700414f574abd6e3af58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260320
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36689}
This is mainly an issue when sending items with partial reliability,
with high bandwidth on a link with packet loss.
In SCTP, when a fragment isn't included in the SACK a number of times,
it's scheduled to be retransmitted or abandoned, if it has been
retransmitted too many times already (depending configuration). Before
this CL, if a fragment was NACKed and scheduled for retransmission, but
couldn't be retransmitted immediately (due to congestion window not
allowing it), future received SACKs - that would still indicate that the
fragment hasn't been received yet - would still increment the
retransmission counter. Which wasn't fair, because this fragment hasn't
had a chance to be retransmitted yet.
With this CL, the fragment will only have its retransmission counter
increased when it is not already scheduled to be retransmitted, but
actually sent on the wire and considered in-flight again.
Bug: webrtc:12943
Change-Id: I2af08255650221c044cc14591a5835c885e94c58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259825
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36683}
Remove the `started_` member variable and some other minor updates to
follow conventions elsewhere in the code.
Bug: none
Change-Id: I4cbb914b39cb2e2787719b906ca937931dc3dad7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258360
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36677}
This changes most deletion paths of Connection objects to go through
the owner class of the Connection instances, Port.
In situations where Connection objects still need to be deleted
asynchronously, `async = true` can be passed to
`Port::DestroyConnection` and get the same behavior as
`Connection::Destroy` formerly gave.
The `Destroy()` method still exists for downstream compatibility, but
instead of deleting connection objects asynchronously, the deletion
now happens synchronously via the Port class.
Bug: webrtc:13892, webrtc:13865
Change-Id: I07edb7bb5e5d93b33542581b4b09def548de9e12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259826
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36676}
This is an implementation API, user classes should in principle
only use the channel_interface.h
Bug: webrtc:13931
Change-Id: I85c285217858dc087c90a50792e980f731f4439f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260185
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36674}
This will make it easier to extend testing, implement new features (e.g.
packet culling) and experiment with new variants.
Bug: webrtc:11340
Change-Id: I747f5f6cff61e11a420e43b06ffe0c4aba438c7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260116
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36670}
Prior to this CL, calling RtpTransceiver::SetChannel() with null
arguments would cause the receiver's track to end. This is wrong,
because the channel can be nulled for other reasons than the transceiver
being stopped/removed - such as when the transceiver is rolled back but
still in use. Also, stopping a transceiver will end the track, so we
should simply ensure to always stop the transceiver when that is needed.
This CL makes sure that the transceiver is stopped or stopping in all
appropriate places, allowing us to remove the ability to end the source
for any other reason. A side-effect of this is that:
- The track never ends prematurely, fixing https://crbug.com/1315611.
- Removed transceivers are always stopped, fixing
https://crbug.com/webrtc/14005.
This CL fixes the issue of track being ended in the ontrack event when
running https://jsfiddle.net/henbos/nxebusjm/.
- We don't have WPT test coverage for this, so I'll add that separately.
With SetSourceEnded() removed, some stopping/stop in response to
rejecting locally SDP munged content had to be added in order not to
regress the existing test coverage for this:
*PeerConnectionInterfaceTest.RejectMediaContent/1
Bug: chromium:1315611, webrtc:14005.
Change-Id: I21f30a1259e51324066dc84f72a72485b9e0fadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36669}
There is no active use of it, and the fields are enabled by default in
the uses of it.
Change-Id: Ibfdb3f1befca886cb4b2f4b2ae4d6235aafafe3d
Fixed: webrtc:13998
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256262
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36655}
Also remove tests in //tools_webrtc/mb/mb_unittest.py that are testing exclusively code in //tools/mb/mb.py.
Bug: webrtc:13662
Change-Id: Ifdfbe26c11f7c315e307856b1d3ab06483d57641
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260041
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36651}
In the referenced ticket, we changed the allowed_lag from 40h to 90h (for low volume commit periods) to survive weekends without succeeding builders.
This rolls the change into WebRTC, and removes a duplicated builder entry.
Bug: chromium:1317669
Change-Id: Ife9dd6ff119948ffbd0fa92c7d59466d31ebb215
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260040
Auto-Submit: Alex Schulze <alexschulze@google.com>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36650}
This is the first step in the next refactoring:
1. Make general peer params immutable and accessible only via const ref.
2. Extract params which can be changed during the call
3. Expose mutators for mutable params protecting them with proper
locking.
Bug: b/213863770
Change-Id: Ie3ac17918f1fed3b8ec84992f8b0afc402bce7f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260003
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36647}
Types with ToLogString implemented were not being recognized correctly.
Now types like TimeDelta and Timestamp can be logged as normal.
Change-Id: Ia15f90bdd1d63a39f7452f9b4bba178d01b74352
Bug: webrtc:13995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259863
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36646}
This will simplify upcoming changes to the queue.
Bug: webrtc:11340
Change-Id: Id023618fdef8a8bc9fb50e477cc87e6ba20c5421
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36644}
