Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
SystemTimeNanos() will soon be replaced with another implementation
when built with Chromium. This will break the assumption of
SystemTimeNanos() being relative to the NTP epoch. To avoid breaking
any UWP apps, call the TimeHelper::Ticks() function directly, which
is synchronized with the NTP epoch during initialization.
Bug: chromium:516700
Change-Id: I4e50cb3f88d06e1385e73b1a9ded52956501dc1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208520
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33324}
Currently the echo canceller reference signal is high-pass filtered to
avoid the need of modeling the capture-side high-pass filter as part of
the echo path.
This can lead to the lowest frequency bins of the linear filter
diverging as there is little low-frequency content available for
training. Over time the filter can output an increasing amount of
low-frequency power, which in turn affects the filter's ability to
adapt properly.
Disabling the high-pass filtering of the echo canceller reference solves
this issue, resulting in improved filter convergence.
Bug: webrtc:12265
Change-Id: Ic526a4b1b73e1808cfcd96a8cdee801b96a27671
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208288
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33322}
Use 4 threads for 360p and above.
Use tile rows for VGA and 4 threads.
Use speed 8 for 360p.
Change min max qp scaling threshold.
Bug: None
Change-Id: Ib7a5b7e539d26d9fa60aa2c4a75eb6f4b19f7dea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208340
Commit-Queue: Jerome Jiang <jianj@google.com>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33320}
This CL refactors RtpSender and extracts handling of sequence number
assignment and timestamping of padding packets in a separate helper
class.
This is in preparation for allowing deferred sequencing to after the
pacing stage.
Bug: webrtc:11340
Change-Id: I5f8c67f3bb90780b3bdd24afa6ae28dbe9d839a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208401
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33316}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
VideoBroadcaster is marked as thread-safe, but that is currently not the
case as OnDiscardedFrame() iterates through an std::vector of sinks in
VideoSourceBase that is not thread-safe and elements of that std::vector
are added/removed in AddOrUpdateSink()/RemoveSink() that could be called
on a different thread.
Bug: None
Change-Id: I5b61127f7ea6ce7f1322c5e770ab56643d7bd0d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208404
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33313}
Simulcast with one active stream:
Use pixels from single active stream if set (instead of input stream which could be larger) to avoid going below the min_pixel_per_frame limit when downgrading resolution.
Bug: none
Change-Id: I65acb12cc53e46f726ccb5bfab8ce08ff0c4cf78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208101
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33309}
Instead of tracking packets accepted by PacketBuffer, track all incoming
packets, including packets discarded before getting into PacketBuffer.
Bug: b/179759126
Change-Id: I4d270c61455608fb78b0df8f27760868f4c98205
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208289
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33305}
This is the first CL out of three to enable overriding
of the function SystemTimeNanos() in
rtc_base/system_time.cc
Bug: chromium:516700
Change-Id: I7c33b0d3463fd68c777ef0c6d268dbde45746c64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208225
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33303}
Before this CL, WebRTC created a decoder for each negotiated codec
profile. This quickly consumed all available HW decoder resources
on some platforms. This CL adds a field trial,
WebRTC-PreStreamDecoders, that makes it possible to set how many
decoders that should be created up front, from 0 to ALL. If the
field trial is set to 1, we only create a decoder for the
preferred codec. The other decoders are only created when they are
needed (i.e., if we receive the corresponding payload type).
Bug: webrtc:12462
Change-Id: I087571b540f6796d32d34923f9c7f8e89b0959c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208284
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33300}
This is a reland of 5cc99570620890edc3989b2cae1d1ee0669a021c
Original change's description:
> Batch assign RTP seq# for all packets of a frame.
>
> This avoids a potential race where other call sites could assign
> sequence numbers while the video frame is mid packetization - resulting
> in a non-contiguous video sequence.
>
> Avoiding the tight lock-unlock within the loop also couldn't hurt from
> a performance standpoint.
>
> Bug: webrtc:12448
> Change-Id: I6cc31c7743d2ca75caeaeffb98651a480dbe08e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207867
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33291}
Bug: webrtc:12448
Change-Id: I7c5a5e00a5e08330ff24b58af9f090c327eeeaa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208221
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33296}
This is for better compatibility with thread annotations and how the
histogram data is collected. In the dtor we can make assumptions about
the state of the object, but that context is lost in member methods
even though they're only called from the destructor (and therefore
thread annotations can't "know" that the object is being destructed
inside those calls).
Bug: webrtc:11993
Change-Id: I8b698cc3340fb0db49430da6f7a9b9a02cabf0c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208200
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33295}
This reverts commit 5cc99570620890edc3989b2cae1d1ee0669a021c.
Reason for revert: Seems this CL breaks the below test when being imported in google3
https://webrtc-review.googlesource.com/c/src/+/207867
Original change's description:
> Batch assign RTP seq# for all packets of a frame.
>
> This avoids a potential race where other call sites could assign
> sequence numbers while the video frame is mid packetization - resulting
> in a non-contiguous video sequence.
>
> Avoiding the tight lock-unlock within the loop also couldn't hurt from
> a performance standpoint.
>
> Bug: webrtc:12448
> Change-Id: I6cc31c7743d2ca75caeaeffb98651a480dbe08e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207867
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33291}
Bug: webrtc:12448
Change-Id: I2547f946a5ba75aa09cdbfd902157011425d1c30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208220
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/master@{#33294}
This avoids a potential race where other call sites could assign
sequence numbers while the video frame is mid packetization - resulting
in a non-contiguous video sequence.
Avoiding the tight lock-unlock within the loop also couldn't hurt from
a performance standpoint.
Bug: webrtc:12448
Change-Id: I6cc31c7743d2ca75caeaeffb98651a480dbe08e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207867
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33291}
- Reset functionality based on loss history
- BWE rampup/down moved to SendSideBandwidthEstimation::UpdateEstimate to align with other estimators.
Bug: None
Change-Id: Ic13795c7ed1852b38baf8359c5c9f4dae6e9ea04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207427
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33288}
- This contains a CallbackList disconnect and handled it
by taking the given subscription tag to subscribe and unsubscribe.
- Left the original sigslot variable until downstream is update after
this change.
Bug: webrtc:11943
No-Try: True
Change-Id: Ie96d74b9594eae11beaa552f61e40f451242bfab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203780
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33285}
This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d.
Reason for revert: Fixed the bugs
Original change's description:
> Revert "Split peer_connection_integrationtest.cc into pieces"
>
> This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe.
>
> Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc).
>
> Original change's description:
> > Split peer_connection_integrationtest.cc into pieces
> >
> > This creates two integration tests: One for datachannel, the other
> > for every test that is not datachannel.
> >
> > It separates out the common framework to a new file in pc/test.
> > Also applies some fixes to IWYU.
> >
> > Bug: None
> > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33244}
>
> TBR=hbos@webrtc.org,hta@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> No-Try: True
> Bug: None
> Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33255}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: None
Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33283}
