This is needed to get rid of a mock in Chrome.
Bug: webrtc:14594
Change-Id: I27df2a1466e6a2dea87a211f803b3f2c7aa57478
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280041
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38458}
This patch adds variant of PeerConnectionInterface::AddTrack
that takes an initial_send_encodings.
This allows for setting/modifying encoding parameters before sdp
negotiation is performed/complete (e.g requested_resolution).
This is already available if using RtpTransciverInit and AddTransceiver,
but was not added to AddTrack because of concerns that it complicated matching with existing transceivers. This CL sidesteps that by never matching to a preexisting transceiver if initial_send_encodings are specified.
Note:
1) The patch adds a new method rather than an extra (e.g optional)
argument to existing AddTrack. This is to avoid problems with downstream mocks.
2) chromium "problems" was fixed in https://chromium-review.googlesource.com/c/chromium/src/+/3952684 and https://chromium-review.googlesource.com/c/chromium/src/+/3956060
Bug: webrtc:14451
Change-Id: I19b5a03872730280fbf868ca5d3a2f46443359f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38437}
This reverts commit c1d5fda22c8ae456950c5549d22d099b478c67e2.
Reason for revert: This CL created thousands of metric alerts in the perf tests. It's possible that these are all expected, but since mbonadei@ is OOO right now, I think it's better to revert, and have him re-land when he is back.
Most alerts are here: https://bugs.chromium.org/p/webrtc/issues/detail?id=14549
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138
Change-Id: I5bc56c954bb12e7c27cb859e838f0b7a89e006f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279522
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38415}
This CL increases the test coverage for webrtc::SimualtedNetwork, adds
some more comments to the class and the interface it implements and
simplify the logic around capacity and delay management in the
simulated network.
More CLs will follow to continue the refactoring but this is the
ground work to make this more modular in the future.
Bug: webrtc:14525, b/243202138
Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38388}
Only use the network thread for sending and receiving packets.
The one and only network thread is used as a worker thread in all
PeerConnections. Pacing when sending packets is done on the worker thread.
Bug: webrtc:14502
Change-Id: Ib373315688ae4d810ae1e4421101a859fca93b31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278621
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38354}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
This updates VideoReceiveStream2::Stop() to symmetrically tear down
state that's built up in VideoReceiveStream2::Start().
Bug: webrtc:11993, webrtc:14486
Change-Id: I41f4feea5584e5baaeed2143432136f8b9761321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272537
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38244}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
Creates the EmulatedSFUConfig that will receive the parameters for
controlling the virtual SFU used in the call.
Its current only field is the previous target_spatial_index from
VideoSimulcastConfig.
This allow to filter out the bottom layers for SVC S mode tests
and enable them.
Bug: webrtc:11607
Change-Id: Id4f3a96b3a03b9be7155796c3bafefce01f32b7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274162
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38182}
Rename C++ API units to match new proto format units in metric.proto
Bug: b/246095034
Change-Id: Ice5d388a080c474f275ef597f98fac1bcb98cf49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276100
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38137}
Introduce main API for new test performance metrics logging system.
Bug: b/246095034
Change-Id: I9b33740bfe69158c2d7f3f73e18442d1683aa8d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274901
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38116}
This is unused at the moment and webrtc::SimulatedNetwork is going
through a refactoring, to keep things simple and well tested this CL
removes CoDel but nothing blocks us from re-implementing it when needed.
No-Try: True
Bug: webrtc:14426
Change-Id: Ie7d40d20a66d3939fc7d3251c47e4f13f3869a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274407
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38032}
This reverts commit 021512b76a872b04e803d61f46c740ed363d641b.
Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly.
Original change's description:
> rtpsender interface: make pure virtual again
>
> after providing default implementations in Chromium tests
>
> BUG=None
>
> Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37941}
Bug: None
Change-Id: I40f27c36819365fadae32032521f7e11184bee62
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484
Owners-Override: Andrey Logvin <landrey@google.com>
Commit-Queue: Andrey Logvin <landrey@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37947}
The input SocketAddress for STUN host lookup is constructed with just
the hostname, so the family is AF_UNSPEC. So added an overload with a
target family to distinguish this from the family of the input addr.
Bug: webrtc:14319, webrtc:14131
Change-Id: Ia5ac5aa2e894e0c4dfb4417e3e8a76a6cec3ea71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270624
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@google.com>
Cr-Commit-Position: refs/heads/main@{#37750}
This CL propose a new API for video dumps in PCLF also removing
differences between p2p and multipeer usage of API.
Bug: b/240540206
Change-Id: Id4d32cc98250500949b3f9e2cf2e86c4bdce7efb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37665}