By producing new message only when new max is 10% larger than the previous max.
Bug: b/305042040
Change-Id: Id85784939f944de8115b881471b02214c34b3043
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323841
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40950}
The purpose is to ensure send socket buffers are not overfilled at high
pacing rates.
Bug: chromium:1354491
Change-Id: Ic6f473080292f84a2a099b85fb5817f7e14e7355
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323000
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40911}
since it might have been deregistered previously.
BUG=chromium:1454860,chromium:1459124
Change-Id: I70ba43265361d040e568f83b6400ff8f3c2a8e98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40431}
This CL removes a PostTask in response to packet receipt reception.
This is made possible due to PacketRouter lock removal in
https://webrtc-review.googlesource.com/c/src/+/300964.
Depending on how transport code is organized, this may lead to
possibility of packet receipts arriving in
RtpTransportControllerSend which may re-enter the PacingController's
ProcessPackets method, leading to out-of-order packet sends. Fix
this by detecting re-entry and avoiding a second ProcessPackets call
in the TaskQueuePacedSender.
Bug: chromium:1373439
Change-Id: I24928f2d28a240d0860fe7e4a114cedf1f13d2bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40017}
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.
PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.
When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.
Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
This CL prepares for send packet batching support in later CLs.
Bug: chromium:1439830
Change-Id: I0bbecfa895aa6d4317ef8049b3789272a440d032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304282
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40009}
This helper class is no longer used.
Bug: webrtc:14502
Change-Id: I7940de762ebb9a7c6d04927603f249f5b0061051
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39880}
Since PacketRouter now is only used on the worker thread, there is no need for a lock.
Bug: webrtc:14502
Change-Id: I65778f68b7e211d7bc7388a4615888a49ceb5f59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300964
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39848}
This remove use of MaybeWorkerThread from TaskQueuePacedSender. Instead,
the TaskQueue used when creating the TaskQueuePacedSender is used for
pacing. That is, the "worker thread".
Bug: webrtc:14502
Change-Id: I504f8e634653af6493e609db6e42b07d488fd699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300820
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39813}
If during shutdown the pacer has packets enqueued during pause these
packets will be posted to the pacer after the worker thread is free -
after the ssrcs should have been cleared. This fixes flakes in
picutre_id_tests.
Bug: webrtc:14985
Change-Id: Ib5547a501670fc145543df32fdc43bbc6596375f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297401
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39640}
Ensure a second probe can be sent, after the first probe has been sent, even though no large
media packets have been sent.
This fixes a bug in https://webrtc-review.googlesource.com/c/src/+/294521
This cl also refactor and simplify a bit. Remove the unecessary state kSuspended.
Bug: webrtc:14928
Change-Id: Ia561441ea3d8b648b025eedd0618c82cca03b418
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296882
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39547}
This CL migrates the task queue paced sender unit test
to the new TaskQueueBase interface.
Bug: chromium:1416199
Change-Id: Id0568bb9a08bf43b92e33fdf45fe75a57e5a7a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39436}
This CL adds functionality to remove packets matching a given SSRC from
the pacer queue, and calls that with any SSRCs used by an RTP module
when that module is removed.
Bug: chromium:1395081
Change-Id: I13c0285ddca600e784ad04a806727a508ede6dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287124
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38880}
These histograms have expired and have no owner.
Remove to clean up the code and save memory.
Fixed: chromium:1117100
Change-Id: I24a009d8e432109c1d62c4a3a16eff5cd21c8541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286660
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38850}
If the real-time clock makes a sudden and large shift forward while
there is at least one packet in the queue, the pacer would previously
try to "replay" the behavior it would have done while it was asleep.
This can lead to grand bursts of padding packets, which is
undesireable.
This CL mitigates this problem by forwarding the internal state clock
to (now - 50ms) if there is nothing to do but generate padding.
Bug: webrtc:11340, b/258509536
Change-Id: I5b8a130d938dd2566f72c1946c139f50e099e5a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285380
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38752}
Created probes are currently timed out after 5s. But to be safe, also limit the number of pending probes to 5.
Bug: webrtc:14392, b/259541308
Change-Id: Ibf630704ffe14cb165ab849b881cf75857376f82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38697}
We have seen a few instances in a down-stream project where the circuit
breaker is still triggering and causing issues.
This CL makes the threshold configurable and adds more debug logging to
try and get to the bottom of this rarely occuring bug.
Bug: webrtc:11340, b/258509536
Change-Id: I92674d446b926ad66538ff9c8be2a32a3d95b057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283762
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38664}
Address perkj's comments left in
https://webrtc-review.googlesource.com/c/src/+/283420. I was a bit
trigger-happy with the submit button.
Bug: chromium:1354491
Change-Id: Ifd052f75af3763b0b52807c31ea790e3efee921d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38638}
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.
When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.
NOTRY=True
Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
This metric was always supposed to be the spec's answer to
googBucketDelay, and is defined as "The total number of seconds that
packets have spent buffered locally before being transmitted onto the
network." But our implementation measured the time between capture and
send, including encode time. This is incorrect and yields a much larger
value than expected.
This CL updated the metric to do what the spec says. Implementation-wise
we measure the time between pushing and popping each packet from the
queue (in modules/pacing/prioritized_packet_queue.cc).
The spec says to increment the delay counter at the same time as we
increment the packet counter in order for the app to be able to do
"delta totalPacketSendDelay / delta packetSent". For this reason,
`total_packet_delay` is added to RtpPacketCounter. (Previously, the
two counters were incremented on different threads and observers.)
Running Google Meet on a good network, I could observe a 2-3 ms average
send delay per packet with this implementation compared to 20-30 ms
with the old implementation. See b/137014977#comment170 for comparison
with googBucketDelay which is a little bit different by design -
totalPacketSendDelay is clearly better than googBucketDelay.
Since none of this depend on the media kind, we can wire up this metric
for audio as well in a follow-up:
https://webrtc-review.googlesource.com/c/src/+/280523
Bug: webrtc:14593
Change-Id: If8fcd82fee74030d0923ee5df2c2aea2264600d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280443
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38480}
The pacer can thus run on the Worker thread or an owned TQ depending on field trial string "WebRTC-SendPacketsOnWorkerThread"
Bug: webrtc:14502
Change-Id: Ic74b92b21371cc62c7b2f62f039bc800dcceef8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277622
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38301}
The value is today set to 200 which is too low for an audio packet to trigger sending probes.
For the initial probing, it would be good if audio packets, that may arrive before the first video frame can trigger sending a probe.
Also fix field trial parsing of required number of probes.
Bug: webrc:14392
Change-Id: I1f3cebcda38b71446e3602eef9cfa76de61a1ccf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38089}
This CL adds the ability to forward aborted retransmission notifications
to specified RTP modules, as well as a way to find the RTX ssrc
associated with a media SSRC.
These will both be used by upcoming logic that can selectively flush
given streams from the pacer queue.
Bug: webrtc:11340
Change-Id: Ief3be47e4fd7dc5a1499bc21890e8979400ecb44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274706
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38050}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
This adds a (default off) flag which makes retransmissions be processed
immediately, just like audio packets normally are.
This might increase send rates and thus losses in some cases, but will
also reduce retranmission delays especially when timer slack or bursting
is used. Usefuleness TBD via experiment.
Bug: chromium:1354491
Change-Id: Icaa83125bfb30826ce72e6e786963d411e05ea57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272483
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37926}
And add a unit test that verifies that next probe time is set at is
the expected if the recommended probe size is used.
Bug: webrtc:14392
Change-Id: I239bb3a1c8eefc85509aacc82037c64e3ce49ed7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272648
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37901}
This removes the field trial WebRTC-Pacer-UsePrioritizedPacketQueue.
Bug: webrtc:11340
Change-Id: I9a7ee64ff5ae3ad1fee6ed5d552ec681e3b4b534
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272240
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37852}
This experiment will tell if we still see the performance gains that we
saw with the "bursty slacked pacer" even if we don't apply slack (since
the "slack without burst" showed little impact at Stable).
The hope is that without slack all quality regressions will go away but
that bursting will still provide the desired performance benefits.
Bug: chromium:1354491
Change-Id: I95f05d040713addaaa1856c8e374a01c27311612
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272366
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37845}
follow-up from https://webrtc-review.googlesource.com/c/src/+/262810
* replace Time::Millis(0) and TimeDelta::Millis(0) with ::Zero()
* drop unnecessary webrtc namespace from some TimeDeltas
* make TimeDelta do the unit conversion for stats
BUG=webrtc:13756
Change-Id: Ic60625ae0fc7959a47a6be9f5051851feaf76373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37664}
This should avoid any potential race in static initialization for the
tracing logic.
Bug: webrtc:12715
Change-Id: Ic91d8e5fbd9b45a91e7e7a9e76226fc558e00c4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270381
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37662}
Keeping the field trial around as a kill-switch for another milestone.
Bug: webrtc:11340
Change-Id: I3285baefab176f541cbb5ed3bacbc669d3e8836f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269384
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37611}
