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}
This is a reland of commit 905c3a6c73d293882ef11942066ccda52a9e14d1
Change from previous attempt is between ps#1 and ps#2: Use PeerConnectionFactoryInterface::Options to clear the `network_ignore_mask`.
Original change's description:
> Move injection of PacketSocketFactory from PC to PCF
>
> Injection via PeerConnectionDependecies was broken, in not accepting
> ownership of the injected object.
>
> Bug: webrtc:7447, webrtc:14204
> Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37270}
Bug: webrtc:7447, webrtc:14204
Change-Id: Ic78ebec2e88a8c44699015c8c7a44e137f44253a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265982
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37290}
This reverts commit 905c3a6c73d293882ef11942066ccda52a9e14d1.
Reason for revert: New test fails internal tests, with a similar problem as the failed android test: No networks are detected on the test bot.
Original change's description:
> Move injection of PacketSocketFactory from PC to PCF
>
> Injection via PeerConnectionDependecies was broken, in not accepting
> ownership of the injected object.
>
> Bug: webrtc:7447, webrtc:14204
> Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37270}
Bug: webrtc:7447, webrtc:14204
Change-Id: Ib412d09699a48d8f5db27e2960e365b536ab3aa8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266146
Owners-Override: Niels Moller <nisse@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37273}
Injection via PeerConnectionDependecies was broken, in not accepting
ownership of the injected object.
Bug: webrtc:7447, webrtc:14204
Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37270}
This cl adds a forwarding header, a build target, and migrates headers
in api/ to use it.
Moving actual implementation, will follow, in
https://webrtc-review.googlesource.com/c/src/+/265390.
Bug: webrtc:12701
Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37200}
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36890}
VideoSimulcastConfig::encoding_params and VideoConfig::min/max_encode_bitrate_bps
were replaced by VideoConfig::encoding_params.
All usage of the previous options has been updated to the new option.
Bug: webrtc:11607
Change-Id: I52cd9efa6e640929485da7aa1e61d15a1a693b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261949
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36865}
Similar mocks are used internally, there should only be one in WebRTC.
Bug: None
Change-Id: Ic5163ae0c554c602344a0d25d17c3f0d46fc2e95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261955
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36864}
With the encoding parameters in the SimulcastConfig objects, it wasn't
possible to configure explicit encoding parameters when using singlecast,
required for example to use the spec standard SVC API.
Bug: webrtc:11607
Change-Id: I92b1446e772e2ecec93379dc91a3da159b8bc209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260002
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36731}
Move VideoSubscription::Resolution to the fixture class and rename it to
the VideoResolution. It should be then integrated into other video
related classes.
Bug: b/213863770
Change-Id: Ifd391f840ef8de43bbac66d23df3ecf7258b3943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259523
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36594}
