RTPFragmentationHeader is already ignored by H264 packetizer
and thus doesn't need to be provided and calculated.
Bug: webrtc:6471
Change-Id: I45bc22827f0dc811457e3ebe477a16293501c2fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179843
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31791}
It is a follow up CL to
https://webrtc-review.googlesource.com/c/src/+/179368.
Now when network stats became more complex structure it's better to hide
its implementation details and provide an interface for read-only
access.
Bug: webrtc:11756
Change-Id: I1980ef938f8de0c6aa90092d1dc90a14a82e0ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179840
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31784}
The time precision of delayed tasks is one millisecond, so the
TaskQueuePacedSender makes sure that is the minimum sleep time, and
then allows sending prior data as if it was on time.
Furthermore, if there already exists a pending task within 1ms of a
new desired process time - we don't schedule a new one with the same
motivation as above.
These two facts clashes somewhat with how BitrateProber works, and
especially if they coincide it can result in scheduled ProcessPackets()
that is 2ms late. The default timeout set in BitrateProber is 3ms, so
there is a higher risk of probes timing out.
This CL changes the TaskQueuePacedSender to allow scheduling a
ProcesPackets() call as soon as possible if we are probing - even if
that means executing up to 1ms earlier than expected (the BitrateProber
will compensate for that). The PacingController is updated in order to
allow early execution in this one case.
Bug: webrtc:10809
Change-Id: Ia5097ddc39aa80c05ebfe56369310c94ef0e0baf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178901
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31778}
The app showcased the ability to send real-time voice data between two endpoints using the VoIP API.
Users can also configure session parameters such as the endpoint information and codec used.
Bug: webrtc:11723
Change-Id: I682f4aa743b707759536bce59e598789a77b7ec6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178467
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31775}
local_addresses is a list of IPs that were used to send data, which was
used during stats calculation.
Bug: webrtc:11756
Change-Id: Ie6307eaa69c73ebe9f69e44503752151be9e9ef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179841
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31774}
It is not longer needed by the rtp_rtcp module.
Bug: webrtc:6471
Change-Id: I89a4374a50c54a02e9f20a5ce789eac308aaffeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179523
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31773}
i.e. when chain are used,
require each decode target to be protected by some chain.
where previously it was allowed to mark decode target as unprotected.
See https://github.com/AOMediaCodec/av1-rtp-spec/pull/125
Bug: webrtc:10342
Change-Id: Ia2800036e890db44bb1162abfa1a497ff68f3b24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178807
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31772}
No need to keep error_resilience 1 for layers in AV1
Bug: None
Change-Id: I6570d653a34ed2187307154ccdfd9e941ed8f917
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179742
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31769}
A packet's capture time may be -1 to indicate an unset value. We need to
check that this is the case before adjusting it when generating padding.
Otherwise, invalid values will result.
Bug: webrtc:11615
Change-Id: Ibbeb959f1d4d37dd4d65702494b97246642b57d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176281
Commit-Queue: Dan Minor <dminor@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31766}
This reverts commit c6801d4522ab94f965e258e68259fde312023654.
Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.
One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.
Original change's description:
> sdp: parse and serialize b=TIAS
>
> BUG=webrtc:5788
>
> Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31729}
TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:5788
Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31762}
It was only used by RtcpDemuxer that is now deleted
Bug: None
Change-Id: Ief2c285e293cde3ed7576b194d2df137d6cbad38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178902
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31760}
This avoids a difference in behaviour between mobile and
desktop platforms since the bitrate is now too low for
CELT mode.
BUG=webrtc:11643
Change-Id: I9ac1439bea0ccbbfee7388516932e30d6cb06bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179522
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31757}
This CL adds a parameter to the BirateProber field trial config, which
allows the prober to actually discard probe cluster is pacer scheduling
is too delayed. Today it just keeps going at a too low rate.
Some refactoring was needed anyway, so also switch to using unit types
in more places.
Initially keeps legacy behavior default, to verify no perf regressions.
Bug: webrtc:11780
Change-Id: I9edd114773b10a8d86b54a1a0398a4052aab9dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179090
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31756}
This name change communicates that the recursive critical section
should not be used for new code.
The relevant files are renamed rtc_base/critical_section* ->
rtc_base/deprecated/recursive_critical_section*
Bug: webrtc:11567
Change-Id: I73483a1c5e59c389407a981efbfc2cfe76ccdb43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179483
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31754}
This change implements the GetSourceList and SelectSource APIs from the
DesktopCapturer interface for WindowCapturerWinWgc. No functional
changes were made as the WGC capturer is not in use yet.
I refactored the source enumeration functionality out of the GDI
capturer and into the utils file, so both of the capturers can share
the implementation.
This change also renames the window capturers to include Win in the
name, and updates some of the out dated code style.
I've tested these changes by running the related unit tests and
applying them to a Chromium enlistment and testing on
https://webrtc.github.io/samples/src/content/getusermedia/getdisplaymedia/
Bug: webrtc:9273
Change-Id: If0ca023cb13900ab2b897aec0f38333f75a1b748
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178960
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31748}
instead reparse nalu boundaries from the bitstream.
H264 is the last use of the RTPFragmentationHeader and this would allow
to avoid passing and precalculating this legacy structure.
Bug: webrtc:6471
Change-Id: Ia6e8bf0836fd5c022423d836894cde81f136d1f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178911
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31746}
