GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
This is in pursuit of an issue with another CL, but large enough
to be worth submitting separately.
Bug: webrtc:13931
Change-Id: If470488f092f8640d3a773922f6f0d22765b9e97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261728
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36833}
The killswitch is no longer needed, because the googScreencastMinBitrate
has been successfully removed from the web platform.
The native RTCConfiguration::screencast_min_bitrate is still available
though because there are other downstream users than Chrome.
Bug: chromium:1315155
Change-Id: I2145f9014dbe57bb50e61f1faeacd533d76acb29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261725
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36831}
Instead offer accessors for the specific config values from the struct
that are needed at different times. The remote_ssrc and rtx_ssrc
properties maybe accessed from any thread, other properties have
stricter requiremets.
Bug: webrtc:11993
Change-Id: I3ff8527b13452c773fae1b2574f1e3fd2583b481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261319
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36823}
This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.
Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}
`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.
Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.
Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
This reduces the surface of externally accessible state that belongs
to the class, which makes it easier to control what state belongs to
what thread. In this CL enforcing remote_ssrc() to be conceptually const
and sync_group to conceptually belong to the packet delivery thread.
Bug: webrtc:11993
Change-Id: I7de9366dc0c2bf451b5c58595c2d073b4016f2e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261450
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36813}
* Paramaterize VideoReceiveStream2Test to have variations that run with
and without a metronome.
* Migrate over tests to ensure frame timing is used.
Bug: webrtc:14003
Change-Id: Icccc2f0d548aaa64c50e010056e1e651174e02fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260942
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36812}
This crash happened when:
* The cursor was located in the corner
* The screen was resized so that the cursor position is outside of the frame.
This caused us to add out-of-bound coordinates to the frame's updated_region, which caused crashes further down the pipeline.
Bug: chromium:1323241
Test: new unittest
Test: manually reproduced crash
Change-Id: Ie71db58c8a347f00af8a3803fcd55cdcad6eafac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261263
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jeroen Dhollander <jeroendh@google.com>
Cr-Commit-Position: refs/heads/main@{#36809}
Several tests starting failing when run on trybots on Win10. This CL
fixes several issues that were uncovered.
Issue 1:
Capture failed to start because `get_Size` returned {0, 0}. This is a
known issue in the WGC API that occurs when there are multiple user
sessions on the same machine.
Solution:
Add a `GetSize` method to the `WgcCaptureSource` interface so we can
fallback to other methods if `get_Size` fails.
Issue 2:
The screen capture tests assume there will be displays attached and
fail if there aren't.
Solution:
Always run `IsWgcSupported` for the appropriate capture type.
Issue 3:
ASAN container-overflow in `GetTestWindowIdFromSourceList`
Solution:
Check the validity of the iterator before dereferencing.
Issue 4:
Occasionally, the call to `GetMessage` in the `CloseWindowMidCapture`
test would hang because there were no messages in the queue.
Solution:
Use `PeekMessage` instead which will return if there are no messages.
Bug: webrtc:14002
Change-Id: I69b2f765db87d34a41d6a1796cd5a81f4029be33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260202
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#36802}
Also remove a bunch of functions in ChannelManager that were just
forwarding to MediaEngineInterface.
Bug: webrtc:13931
Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36801}
The state machine for handling resets couldn't handle resets
happening from both sides at the same time.
Bug: webrtc:13994
Change-Id: I2c268e54f4c5c9858913faef91ff00f6af956e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261305
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36799}
When an outgoing stream reset is sent, with sequence number A, and the
receiver can't perform it immediately, it will return IN_PROGRESS with
response sequence number A.
This is then retried with sequence number A+1, and the peer would then
possibly respond PERFORMED with response sequence number A+1.
Before this CL, whenever a request was sent that didn't immediately
succeed, it wouldn't increment its expected response sequence number.
So in the retry above, the socket would still expect the response
sequence number to stay at A, not at A+1.
Bug: webrtc:13994
Change-Id: I6f36d45229a7fb312e97ad15826e0377f4efb64f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261310
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36797}
This reverts commit 45361f78ed18c350b3edcaef19ae4c7cf167e95b.
Reason for revert: Perf alerts galore.
Original change's description:
> Calculate video stream max bitrate using expression.
>
> This replaces the ealier table-based caps.
> Apart from the VGA cap (now 1600kbps instead of 1700kbps), or if using
> "in between" resolutions, the caps are unchanged - but now cover high
> resolutions better.
>
> Bug: webrtc:14017
> Change-Id: I8649b528495d6c917e38ea8cb1a272df6c464c03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260940
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36776}
Bug: webrtc:14017
Change-Id: I18ebc81c6054713c58d49bd227e37090686958c9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261309
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36794}
To quote rfc6464:
The audio level for digital silence -- for a muted audio source, for
example -- MUST be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
The behavior in webrtc is correct that digital silence is represented
with 127, but it is also possible to get a value of 127 for not quite
digitally silent audio buffer (as in, not strictly 0s).
Bug: webrtc:14029
Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36793}
