Subsequently also tighten IceCandidateType error checking.
The Candidate type in `cricket` should be using something similar
(currently using a string for the type), so I'm making sure that
types that we have already, align with where we'd like to be overall.
Possibly we can move IceCandidateType to where Candidate is defined.
Bug: none
Change-Id: Iffeba7268f2a393e18a5f33249efae46e6e08252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335980
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41640}
This allows to share an instance of VideoCaptureModulePipeWire which is
what browsers usually do when the same camera is being shared with more
than one consumer. This matches V4L2 implementation.
Bug: webrtc:15211
Change-Id: I2ae466739c2649029e76a29e6f16aad1014e9d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306964
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#41639}
H.265 bitstream parser currently always assume pps id to be 0 when
calculating the last slice QP. This assumption is incorrect.
Bug: webrtc:13485
Change-Id: I06918df035e8e4a8e68eb3002a49b824ffd5f516
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337080
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41636}
And RTC_CHECK(NOTREACHED) Socket::RecvFrom(void* pv..)
This cl also change usage of PhysicalSocket to use PhysicalSocket::RecvFrom(ReceivedBuffer&) in Nat and tests.
Note that Socket::RecvFrom(ReceiveBuffer& buffer) is already used in AsyncUdpSocket.( https://webrtc-review.googlesource.com/c/src/+/332200)
AsyncTCPSocket uses Socket::Recv(). Therefore, there should be no production usage left of Socket::RecvFrom(void* pv..) in open source webrtc.
Follow up cls should remove usage of Socket::RecvFrom(void* pv..) in implementations of rtc:AsyncSocketAdapter such as FirewallSocketAdapter.
Change-Id: I597dc32b14be98e954a3dc419723f043e8a7e19e
Bug: webrtc:15368
Change-Id: I597dc32b14be98e954a3dc419723f043e8a7e19e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332341
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41635}
This does not remove all traces of SDES - we still need to delete
the cricket::CryptoParams struct and all code that uses it.
Bug: webrtc:11066, chromium:804275
Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41634}
The division by 2 has been accidentally removed in https://webrtc-review.googlesource.com/c/src/+/76921
The code and comment are out of sync now.
Bug: None
Change-Id: If43a40461878ffe58dd9ed0ab8a6244ad79c4f6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336283
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41627}
If allow_bandwidht_estimation_probe_without_media is true and a writable
video rtp stream with RTX exist, a probe can be sent immediately without
waiting for a large media packet.
Bug: webrtc:14928
Change-Id: Ie2204734f9fe3e6bff9aed4a1f7f8995956d35cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336000
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41626}
This is a follow up to a previous CL that removed direct dependency on
the `cricket::` string globals.
Bug: none
Change-Id: I4d839a36739fc4694ce81b72ee036e83dae580df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41623}
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/
Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
Rename RtpVideoSender::SetActiveModules to SetSending to better match
what it does. When a RtpVideoSender::SetSending(true) RTP packets can be
sent on all associcated RTP streams (simulcast streams).
Move registration of RtpRtcpModules to RtpTransportControllerSend to
allow RtpTransportControllerSend to know when there are sending RTP
streams. Purpose is to in later CLs allow RtpTransportControllerSend to
trigger BWE probes.
Bug: webrtc:14928
Change-Id: Ibf6c040b86713cdc4763c4691b7fd794b251eb49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335961
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41620}
Instead, always use VideoSendStream::Start.
VideoSendStream::StartPerRtpStream was used for controlling if
individual rtp stream for a RtpEncodingParameter should be able to send RTP packets. It was not used for controlling the actual encoder layers.
With this change RtpEncodingParameter.active still controls actual encoder layers but it does not control if RTP packets can be sent or not.
The cleanup is done to simplify code and in the future allow sending
probe packet on a RtpTransceiver that allows sending, regardless of the
RtpEncodingParameter.active flag.
Bug: webrtc:14928
Change-Id: I896c055ed4de76db58d76f452147c29783f77ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335042
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41619}
This CL does:
- Run IWYU on the relevant elements
- Make connection depend on port_interface, not port
- Make port_allocator depend only on port
- Move some constants from port.h into p2p_constants
This allows a dependency graph without ugly groups.
Bug: webrtc:15796
Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41618}
With this CL the migration to absl::optional<T> is complete.
Bug: webrtc:15164
Change-Id: I978d86833fbd39154ed9026206a02ea7496f8c4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41616}
This is a reland of commit 33c7edd58ad0edc71939b9372fff3ab563c1f4a7
taking into account GFD which can be enabled by field trials and somewhat conflicts with DD
Original change's description:
> Enable DD and VLA header extensions by default for Simulcast/SVC
>
> When Simulcast (more than one encoding) or SVC (a scalability mode
> other than the default L1T1) is used, enable the AV1 Dependency
> Descriptor and the video-layer-allocations RTP header extensions by
> default.
>
> The RTP header extensions API can be used to disable them if needed.
>
> BUG=webrtc:15378
>
> Change-Id: I587ac32c9d681461496a136f6950b007e72da86d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326100
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41332}
Bug: webrtc:15378
Change-Id: I190edc9435083c0a0a65a6959363f3c41e4a3d1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330563
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41615}
With this CL, the only usage of RTCStatsMember within WebRTC is the
actual type alias declaration. It's not referenced anywhere anymore.
This allows us to deleting the type alias, but let's do that in a
standalone CL in case it gets reverted.
Bug: webrtc:15164
Change-Id: I766d07abb62b5ddd524859b8ed749394fc439e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335621
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41612}
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.
One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.
Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
Low-Coverage-Reason: EXPERIMENTAL_CODE Code is behind field trial that will only be used for testing.
Bug: webrtc:13322
Change-Id: Ie306be808381b3a20b4e0d58349927bf3524018a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335840
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41608}
This is a reland of commit 79ac694d9b70fa9cd7b6a0f00bbee5d7fbbe64de
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
# Only unrelated bot failures
NOTRY=True
Bug: webrtc:15164
Change-Id: I0f1991409326679a260cb24907808eaa28385350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41603}
This isn't a problem right now but will be when we move away from
using a free string as the storage type.
Bug: none
Change-Id: I12c2de416f9fce13a8284e7d36fa3c2081453299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335880
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41602}
These are no longer needed since the RTP transport runs on the worker
thread now.
Some tests that were too strict on ordering needed change.
Bug: none
Change-Id: I4265cb1a4fd3355208f19aefdbb7abeb45b6cadf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335700
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41599}
This reverts commit 79ac694d9b70fa9cd7b6a0f00bbee5d7fbbe64de.
Reason for revert: Breaks downstream (found another use of
ValueOrDefault instead of value_or)...
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
Bug: webrtc:15164
Change-Id: Ice3f44057b82a7ba9be000d9a0b714152fd07d2f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335701
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41595}
The moment we've all been waiting for.
Step 1: Add type alias (this CL).
Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
Step 3: Delete type alias.
Bug: webrtc:15164
Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41593}
Instead embed functionality of the rtc::TaskQueue into destructors and describe the potential race.
Bug: webrtc:14169
Change-Id: I01b570b530986a0d07798893057201493a8bef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335141
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41592}
There can only be one!
Bug: webrtc:15164
Change-Id: Ib7265bf2103f24a6dab07737b2caed7f39ba75c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334643
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41591}
