For now use doubles as units in api/units have insufficient precision for jitter estimation.
Bug: webrtc:14381
Change-Id: I5a691b6a404b734a5bef11d677b72040bc02ff0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272367
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37841}
Now Configure(), Decode() and Release() calls to the decoders should
all happen on the decoder thread. Added thread checkers to verify.
Bug: None
Change-Id: I2a1cf2cf7f3c3c7c50e382d82a3638e916ed9c34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272368
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37840}
which has been superseeded by the equivalent nonstandard sdp fmtp
sps-pps-idr-in-keyframe
parameter.
Bug: webrtc:11769
Change-Id: I02667a165dd3f86b4685530c43f19531ec654737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271121
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37839}
This is a reland of commit 02b5f3c9c12cddf3fc6e9125238b77ddb44f3b53
without making SetRemoteFingerprint private (but adding a deprecation warning)
Original change's description:
> dtls: allow dtls role to change during DTLS restart
>
> which is characterized by a change in remote fingerprint and
> causes a new DTLS handshake. This allows renegotiating the
> client/server role as well.
> Spec guidance is provided by
> https://www.rfc-editor.org/rfc/rfc5763#section-6.6
>
> BUG=webrtc:5768
>
> Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#37821}
Bug: webrtc:5768
Change-Id: I8dd674db8b683160013e1b4aa7776775d130978f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272221
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37838}
Re-enabled multithreaded encoding using OpenH264, as the issue described in crbug.com/583348 no longer applies.
Bug: webrtc:14368
Change-Id: I5ae768a6edf3b40d99c13fb4ee4662626c993a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271820
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37837}
This way we're sure instantiation, configuration and decode calls all
happen on the decoder queue - making thread checking easier in the
actual decoder classes.
Bug: None
Change-Id: Ia98f47009f26b34eb8dad2ee0b4ddcde082d1994
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272022
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37825}
Encapsulate the results in a struct and move the post back to the packet
sequence in the original method. This allows the calls to
ReceiveStatisticsProxy::OnPreDecode to be moved to the main thread
always.
Bug: webrtc:11993
Change-Id: I37516444efeabdb330868733bccb1e842ea4d54f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271285
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37823}
This was only set to nullptr in non-test environments and was thusly
unused. With this change, the stats callbacks are gaurenteed to only
come from the VideoStreamBufferController and so the thread checks can
be removed.
Bug: webrtc:14003
Change-Id: Iaf0e77aa7c45a317e38ae27739edcefd3123d832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37816}
I've added the proper headers to the only file in Chromium which includes screen_capture_frame_queue.h (see https://chromium-review.googlesource.com/c/chromium/src/+/3836317).
I've also built the remoting host and Chrome on Windows and Linux with this change and did not see any build errors.
The only build error I encountered was in shared_screencast_stream when building webrtc so I added the required header there.
Bug: webrtc:14378
Change-Id: Ie88e606dfa52f18514a87b87e5904424543d7df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271922
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37811}
While transitioning to TimeDelta, WebRTC and Chromium has a
different idea about what type rtc::Event::kForever is. Code
can't assume rtc::Event::kForever is the same type as timed
wait arguments.
Bug: webrtc:14366
Change-Id: I4783c7de6d567c70b211de9aa9889417f6fafba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37810}
Use TimeDelta type instead of raw ints
Remove usage of rtc::MessageHandler and RTC_FROM_HERE
Reduce usage of sigslots
Simplify storage for Channels and Permissions.
Classes that were previously self deleted are now deleted by their owner
Bug: webrtc:9702, webrtc:11318, webrtc:11943
Change-Id: I32c63c11c16c42ae58a65020b1c20a03b21a5abb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271298
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37808}
Moves FrameBuffer2 to its own GN target to reduce the binary size of the
video target.
Bug: None
Change-Id: I40e86a1eabc0c9e8e6fada3dcdb4e3a043c61c6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271286
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37803}
It no longer has to interact with the decode queue, that will only
happen in VideoReceieveStream2. This moves some members in
VideoReceieveStream2 to the packet sequence which removes a few
post-tasks.
Bug: webrtc:14003, webrtc:11993
Change-Id: I4641b593b1a2f68e017c384b73ee4e75d06cf559
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271700
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37802}
- Move existing check on `max_frame_size` to the top.
By doing this early, the filter will not end up in an
inconsistent state (predicted but not updated) when
called with a tiny `max_frame_size`.
- Add sanity check on noise variance.
This will avoid sqrt of a negative number.
Bug: webrtc:14151
Change-Id: I2507a9114655d3c3ae35bf5ed83f3f1154c42ad3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271281
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37798}
In migrating rtc::Event to use TimeDelta instead of int,
rtc::Event::kForever will have to become something else.
This change removes dependencies on that kForever is int.
Bug: webrtc:14366
Change-Id: Ic36057dda95513349e7ae60204e7271ff1f58825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271288
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37795}
