Before:
No attempt was made to figure out of the cursor was embedded into the
captured video frame when using DXGI on Windows as screen capturer.
Instead the cursor is superimposed on the frame by an external mouse
and cursor composer.
After:
We now check if the display adapter supports embedding the mouse
cursor and if so use it as is and thereby avoid adding it independently.
Bug: chromium:1421656
Change-Id: Ie07fe13e1c8f9583769961328bb41fbc689cd8e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299241
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39742}
If configured, the packet parser will allow packets with
a set checksum of zero. In that case, the correct checksum
will not even be calculated, avoiding a CPU intensive
calculation.
Also, if specified when building a packet, the checksum can
be opted to be not calculated and written to the packet.
This is to be used when draft-tuexen-tsvwg-sctp-zero-checksum
has been negotiated, except for some packets during association
establishment.
This is mainly a preparation CL and follow-up CL will enable
this feature.
Low-Coverage-Reason: Affects debug logging code not run in tests
Bug: webrtc:14997
Change-Id: I3207ac3a626df039ee2990403c2edd6429f17617
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39737}
(using no-try due to bot infra issue)
No-try: true
Bug: b/276434297
Change-Id: I33f796b501f96731c4ca76cb62c2331f10c795f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299708
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39734}
This is needed in order to be able to update the legacy stats
collector to fetch data channel stats from the network thread, which
is part of an upcoming change to data channels.
Bug: webrtc:11547
Change-Id: Ic205b0314b9f11a024d36d714c223cbddd0f3df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299462
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39732}
The goal is to reduce the amount of time stretching done in response
to network jitter. Specifically, we should be able to “ride” over delay
spikes if the current delay is sufficient, without decelerating
playout. We should also avoid accelerating immediately after a buffer
underrun, until we are reasonably sure that the jitter has passed.
This is achieved by increasing the deadband where we choose to do
normal playout, based on the maximum delay in the short term packet
arrival history.
The buffer level filter is still used to report the average delay for
A/V sync purposes.
The new behavior is behind a flag and will be experimented with before
it is made default.
Bug: webrtc:13322
Change-Id: I5fba0c9d46d835dbe5401669598fa031512ccced
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299500
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39730}
With this cl, packets that are discarded in RtpTransport now notifies Call, so that
they can be part of BWE even if they are dropped.
These packets have been recevied on the transport, and has bin decrypted
and parsed and thus can be accounted for.
The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded.
Bug: webrtc:14928
Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39727}
in order to not regress existing use-cases while following rules
described by the specification. This change now makes the existing
regression test pass after the spec-compliant modifications.
BUG=chromium:1051821
Change-Id: Ia384adf9a172ed88b5ec6a3cc5c478764a686cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39726}
The tests now have two ways of addressing the channel under test:
* channel_: All SctpDataChannelTest methods. This way calls get
executed as specified by the proxy and more accurately reflects
production.
* inner_channel: For SctpDataChannel methods that are not a part of
the SctpDataChannelTest interface. Use this to invoke event handlers
etc.
Upcoming changes will include threading changes and changes to the
proxy, so it's important to cover both.
Bug: webrtc:11547
Change-Id: I26c284ece82b9a58e2b5dc4468d124d54012d959
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299264
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39725}
* Rename id_ -> id_s_, add id_n_ and thread guards.
* Same for getters, sid() -> sid_s(), add sid_n()
As more things migrate over to the network thread, we'll only need the
_n variant.
Bug: webrtc:11547
Change-Id: Ic998330f4c81b0f6833967631ac70edc2ca2301c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299141
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39724}
It is a part of "encoding statistics" feature [1] available in Android SDK 33. Local testing revealed that for HW VP8/9 encoders we get QP in range [0,64] which is not what WebRTC quality scaler expects. Exclude VP8/9 encoders for now.
[1] https://developer.android.com/reference/android/media/MediaFormat#VIDEO_ENCODING_STATISTICS_LEVEL_1
Bug: webrtc:15015
Change-Id: I8af2fd96afb34e18cb3e2cc3562b10149324c16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298306
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39722}
The reason is that:
* SDP/PeerConnection code should use HasUsedDataChannels() instead.
* HasDataChannels() touches the list of data channels on the signaling
thread, which will be a problem when it's moved to the network
thread.
* It's only needed by tests.
Bug: webrtc:11547
Change-Id: Idd47365c429e5f1d6e3812cf558c4e6fefbf733c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299481
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39721}
...when checking if negotiation is needed.
I noticed this call site after we recently updated PeerConnection here:
https://webrtc-review.googlesource.com/c/src/+/297860
Bug: chromium:1423562
Change-Id: Id37c938d731eadfccff44c95ef757a3cabd64936
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299480
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39720}
* Change data channel creation code to return RTCError for more
detailed/accurate errors.
* Move DataChannelController::sid_allocator_ to the network thread.
* Add a temporary duplicate vector of channels on the network thread.
This will eventually be the main vector.
* Delete one test that turns out to be racy (as long as we're using
both the signaling and network threads).
Bug: webrtc:11547, webrtc:12796
Change-Id: I93ab721a09872d075046a907df60e8aee4263371
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298624
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39719}
This is to avoid counting concealed samples after comfort noise as
speech.
Bug: webrtc:13322
Change-Id: I12cf18d720c697d81376c6f6cdc02d7c6bfa49a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299300
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39717}
MediaType::Any will be used by packets that can not be demuxed by
RtpTransport.
Bug: webrtc:14928
Change-Id: Ib759e65c7eede29defdad8073fd1ed6be814ab81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299280
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39710}
Pass FieldTrialsView by const& to note it can't be null and doesn't need to outlive the constructor
In unittests use AimdRateControl object directly instead of through helpers
Use unit types (TimeDelta, DataRate) directly, reducing their conversion to plain numbers
Replace SimulatedClock with a single Timestamp now variable or constant
Bug: None
Change-Id: I147f85e629b4d8923aa19896ea211a6f9dca1e68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299260
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39707}
UpdateLastDecodedPacket is anyway only called when a new packet is
decoded.
Bug: webrtc:10178
Change-Id: I8cfcc5791e71079034a2d0806c44b3b071ac2ffb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299180
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39703}
This will be included in INIT/INIT_ACK if the socket has
configured the "accept zero checksum" parameter, that will
be added in follow-up CLs.
Bug: webrtc:14997
Change-Id: I1a2823fbc77cfea8fe746b07c1c77593bc15efe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39702}
This flag isn't needed for sctp data channels.
Bug: none
Change-Id: I07b8ba2c5186729b8a5edb4d2bba7b800335ab5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299074
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39701}
which describe the existing behavior that necessitated the revert
b396e2b159b060791954495d68278a55e8f72092
Also change the fake media engine audio clockrate to 8000 instead
of 0 and the fake media engine video payload type to something but
0 as this value seems to be treated specially by the video engine
and is a payload type reserved for PCMU.
BUG=chromium:1051821
Change-Id: Ib0a345d59baba50a565f01685d240e41584367e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299000
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39699}
