This CL allows the users to now enable/disable WGC capturing support
for Window and Screen sharing independently.
Bug: chromium:1314868
Change-Id: Ieeb15539434dac2caf29c515aa7c5dbb7abcc5df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309560
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40330}
Media*Channel objects used to subclass webrtc::Transport.
This was not an optimal design. This CL makes the transport
a member variable of MediaChannelUtil.
Bug: None
Change-Id: I85d33cc1b32b931e563b7bb2d277f1c512600831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309800
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40328}
This feature has had positive impact in downstream experiments, so we should enable it by default. It will be kept around as a kill switch for a while though.
Bug: webrtc:15260
Change-Id: Ibfd25f5be124f65cd4360ae76f7022bb46f65301
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309781
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40327}
specified in https://github.com/w3c/webrtc-stats/pull/762
and take FlexFEC into account for receive statistics.
BUG=webrtc:15250
Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40325}
Scoped enums do not get automatically promoted to their underlying type,
so these uses have undefined behavior and Clang recently started warning
about it.
Bug: chromium:1456289
Change-Id: I9cf4e5a68378930a3bf7d8ac7b0a21eaf0d12670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309520
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Hans Wennborg <hans@chromium.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40321}
As before, events are processed primarily in timestamp order.
This CL adds a heuristic to break ties for events with the same timestamp.
- Roughly speaking, configs and connectivity events are processed first, followed by incoming packets, then BWE updates, then other (general) events and finally outgoing packets and ALR events.
- Among RTP packets, transport sequence number is used to break ties.
- The insertion order (into the EventProcessor) is used as a last resort.
Bug: b/282153758
Change-Id: I914e4500ca63e1a8754766d1833a7b32f6a38176
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308140
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40318}
[RTCIceCandidate initWithNativeCandidate:] does not take ownership on
candidate, so it must be released by caller.
Bug: None
Change-Id: I516e740e81a7aec04556f5fa71cbbecf3be6deb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308500
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40314}
Referencing this type directly is not allowed when building
with the macOS 14.0 SDK.
Other usages in webrtc follow this inline pattern too so
going with this instead of "auto" which also works.
Change-Id: I84a0ba9c78e83843bc73c71c642caca75750f127
Bug: chromium:1454356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40313}
This groups functions for WebRtcVideoSendChannel and
WebRtcVideoReceiveChannel together, rather than interspersing them.
Bug: webrtc:13931
Change-Id: Iecb5bac18e1d370331e9eb546c6b2fde4d92963f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309460
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40312}
Introduces a field trial
WebRTC-IncreaseIceCandidatePriorityHostSrflx
that adjusts the priority of non-relay candidates such that the STUN priority attribute calculated as
(prflx-type-preference << 24) | (priority & 0x00FFFFFF)
as described in
https://www.rfc-editor.org/rfc/rfc5245#section-7.1.2.1
will satisfy the condition that the STUN priority of server-reflexive candidates will always be higher than the STUN priority of relay candidates.
Previously this was not the case because the TURN relay preference was added to the local_preference for relay candidates, making it higher than the local_preference of srflx candidates gathered from the same interface.
This led to cases where the resulting candidate pair priority of a srflx-relay pair was higher than the candidate pair priority of a srflx-srflx pair, i.e. using a TURN server in cases that work using a direct P2P connection.
Whether the field trial is active can be observed by checking that
priority-of-srflx-candidate&0x00FFFFFF
is greater than
priority-of-relay-candidate&0x00FFFFFF
BUG=webrtc:13195,webrtc:5813,webrtc:15020
Change-Id: Ib91708fbe7310b6454f93158a45c9d77da009091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292700
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40311}
Make outgoing encoded audio frames inherit from the same Audio interface
that incoming frames inherit from, to align them and make it possible to
eg clone frames regardless of their direction.
Also begin removing GetHeader() from the Audio interface, replacing it
with getters for the specific values we actually need to propagate in
the API: sequence number and CSRCs. This makes it much easier to treat
incoming and outgoing frames the same, even if they don't have full
RtpHeaders prepared at the point of the transform.
Bug: chromium:1453226
Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40309}
This change fixes a minor issue where we previosuly assumed that the
following was true:
RTC_DCHECK_EQ(map_info.RowPitch, current_frame->stride())
It turns out that this is not always the case when sharing a window
where the stride can sometimes be a few bytes smaller than the
rowpitch.
The code is behind a command-line flag and no tests are affected.
Given limited review resources I therefore plan to bypass the CQ.
I know that it is not recommended but the change has been tested
locally on two different Windows platforms and it does avoid an
existing crash.
Code-Review: alcooper@chromium.org
No-Try: true
Bug: chromium:1421242
Change-Id: I01e7105a6f9fca7ce1349a57635dd373c28d160b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309342
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40308}
This is no longer needed after downstream redefinitions are deleted.
Bug: webrtc:15241
Change-Id: Iea6839bff781fe7d0c56b4739f3d43398c70f2b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40306}
This is a reland of commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199
Downstream project has been updated.
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: Ibc7aeb518ed0bd7f1d725f140132c99e5a89bcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308880
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40305}
Release + dereference operator does not magically move buffer from
heap to stack, so there was a leak.
Bug: None
Change-Id: I9f760b6719ca1fc03aa3efcfda0c0ff9d87efda8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308581
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40303}
Added as stopped by default as it should be requested by the application,
but it should be listed as available.
Bug: webrtc:14631
Change-Id: I301cfd29c79083c97b4a43b8fdafee2dbe4887a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308824
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40300}
This CL adds a light-weight detection of "frame content has changed
since last frame" to an existing pass where bytes are copied from a
texture to a DesktopFrame. The resulting boolean can then later be
used to bypass a full detection of if the content is static or not.
As a result, we only check for static content for a small fraction of
all captured WGC frames and this reduces the total load when 0Hz
is enabled for WGC.
Both WGC and 0Hz support for WGC is still behind a flag.
Bug: chromium:1421242
Change-Id: If9e3721c60a244a3919758fe861d56d4b54cb039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308821
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40299}
This reverts commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199.
Reason for revert: Breaks downstream project
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: I05db80ba9f29460239de82cea9d95136e4c708e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308860
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40298}
This is a reland of commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I123d1134a212f65cfbc90ecec9013d0aafebd9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308721
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40294}
This is useful in environments where OpenSSL may not be available.
Bug: webrtc:15240
Change-Id: I7ba29e44bd1d25231df13ee79dacc74f260ded67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40293}
while not really covered by
https://www.rfc-editor.org/rfc/rfc5576.html#section-4.2
and using the same SSRC for RTX and primary payload may work
since payload type demuxing *could* be used is not a good idea.
This also applies to flexfec's FEC-FR.
For the nonstandard SIM ssrc-group duplicates make no sense.
This rejects duplicates for unknown ssrc-groups as well.
BUG=chromium:1454860
Change-Id: I3e86101dbd5d6c4099f2fdb7b4a52d5cd0809c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308820
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40292}
VideoCaptureV4L2 has some members that are set in StartCapture during
configuration of the device, but later accessed both on the capture
thread and on the api thread (same as StartCapture) in
CaptureSettings(), which can be called at any time. This is safe because
they are written only on the api thread when the capture thread is not
running. However, there is no helper class that separates the read and
write modes to allow annotations and static analysis of the thread
access of these members.
This commit allows annotations to be added by making VideoCaptureV4L2
use:
- VideoCaptureImpl::_requestedCapability for storing those members
requested through StartCapture(), to allow access on the api thread
through CaptureSettings().
- A new member configured_capability_ to replace those members mentioned
in the first paragraph above. Writes to it happen only in
StartCapture() and StopCapture(), while reads happen exclusively on
the capture thread in between.
Bug: webrtc:15181
Change-Id: I27e0f578e6ac2a2e6b0e34fbabfe4f743b296321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306122
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40290}
Also add a preprocessor definition to avoid redefinition in downstream projects.
Bug: webrtc:15241
Change-Id: Ic55d98c3d3a69b9b19195ee78f03af6e38fdd0e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40289}
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
with a single encoding or multiple encodings. As long as only one
encoding is active we get a single L1Tx ssrc, same as legacy API.
Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.
The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.
This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.
Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
CaptureSettings() can read started_ on the api thread at any time. But
it is written and read in pipewire callbacks, on other threads. A lock
seems suitable.
Bug: webrtc:15181
Change-Id: I3d26f02408a4e56921b955f059e504ffa9b8cae9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306121
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40286}
frameInfo_ is used on multiple threads. This commit splits it into:
- VideoCaptureImpl::_requestedCapability for writing what was requested
in StartCapture() and for reading in CaptureSettings(), on the api
thread only.
- A new member configured_capability_ (renamed from frameInfo_) for
accesses in callbacks, or on the api thread when no callbacks can
happen.
Bug: webrtc:15181
Change-Id: I105d8adfde52320e43ffe95fe23e11d028c80684
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306120
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40285}
