With this CL, the decision if an RTP packet should be sent as ect(1) is made in RtpControllerSend depending on if RFC 8888 has been negotiated and if CCFB is received with ECN enabled.
Since webrtc does not yet adapt to ECN feedback, packets are sent as ECT(1) until the first feedback is received.
Change-Id: Iddf63849328afbe54a7c8f921f2e8db134aeff6a
Bug: webrtc:42225697
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367388
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43609}
Fix todo to ensure TransportSequence numbers are generated if CCFB according to RFC 8888 is used. Transport sequence numbers are used in BWE algorithms regardless of feedback format.
Bug: webrtc:42225697
Change-Id: I6eab95c0241d590f6e7a90d19c82d13ab8692f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370341
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43515}
Increased the number of errors the automation is fixing to 150 from
75 in this commit.
Bug: webrtc:370878648
Change-Id: If6e6a5f40db7eb54c27c1a85fb7031838e478c70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43337}
To support libraries and dependencies compatible with absl way of debug printing custom types.
In particular gtest can use AbslStringify to produce nice output when unit types are compared with EXPECT macros.
Bug: None
Change-Id: Ie78293a225f61977f256f0234e07d166b1977e2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364162
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43164}
Repeated initial probes are sent every second until
ProbeController::OnMaxAllocatedBitrate is invoked (Media is beeing sent) or 5s has passed.
Each probe has a duration of 100ms, sent in packet bursts every 20ms.
ProbeController::waiting_for_initial_probe_result_ is no longer needed
and is removed.
Setting field trial for duration between probe packets bursts are moved
from BitrateProber to ProbeController.
Bug: webrtc:14928
Change-Id: I3170533f679fc2cc2aa5402e248fa1f6996d3ccd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350640
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42323}
With this cl
RtpTransportControllerSend::OnAddPacket is instead directly invoked from PacketRouter::SendPacket instead of going via RTP module.
Transport sequence numbers are instead of directly written to header
extension, added to RtpPacketToSendMetaData and written to the extenion
by RTP module.
This is to allow transport sequence numbers without actually sending
them in an extension.
Bug: webrtc:15368
Change-Id: Idd03e02a4257dfc4d0f1898b2803345975d7dad2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344720
Reviewed-by: Erik Språng <sprang@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41974}
And move writing of the header extension from PacketRouter to
RtpSenderEgress::SendPacket.
Bug: webrtc:15368
Change-Id: Ieb18af4bc20115bf02d37e1f9a815a5c120975a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343786
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41949}
If allow_bandwidht_estimation_probe_without_media is true and a writable
video rtp stream with RTX exist, a probe can be sent immediately without
waiting for a large media packet.
Bug: webrtc:14928
Change-Id: Ie2204734f9fe3e6bff9aed4a1f7f8995956d35cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336000
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41626}
Ensure top_active_prio_level_ is set to -1 in MaybeUpdateTopPrioLevel if
last packet is purged.
Bug: webrtc:15740
Change-Id: I81df9ee084de89f79b8ab79db8ce52fe1e20738a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333883
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41498}
Adds separate priorities for audio and video retranmission.
Done by adding an original type to RtpPacketToSend.
Add possiblity to set TTL for audio nack, video nack and video packet separately.
Oldest packet for these types are dropped when a new packet of that type is pushed to the pacer, or when the pacer switch current priority type to that priority.
Effect is that:
-pacer queue does not grow unlimited for these types if a TTL has been set.
-an old packet is not sent.
Bug: webrtc:15740
Change-Id: I38718bc570aebca54eacbded69824905f3694f41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331823
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41414}
PacingController per default use a burst interval of 40ms. The behaviour can still be overriden by using the method SetSendBurstInterval.
Bug: chromium:1354491
Change-Id: Ie3513109e88e9832dff47380c482ed6d943a2f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41254}
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}
