Note: Does not include code for the actual late allocation
of PTs.
Bug: webrtc:360058654
Change-Id: Iaa6bd2db2f974aad84fe1ae9c1aca5aea5d1d25e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362320
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43014}
Disable the checks ensuring we reject mixed-codec simulcast
when the field trial is enabled.
The feature is however not yet implemented.
Bug: webrtc:362277533
Change-Id: Ib1601767c951d61aaa37a3d8767d0a81444d626c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361404
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42942}
This allows to utilize libvpx optimizations considerably improving performance.
The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
Bug: webrtc:347737882
Change-Id: I03bc27c920787a7305a9775e6341e26904592fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360280
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42931}
The fieldtrials can be used to override the static QP threshold
that is used in QualityConvergenceMonitor to determine if an
encoded video stream has reached its target quality.
The fieldtrials do not change the dynamic detection.
Bug: chromium:328598314
Change-Id: I5995860eff461f0c712293e34cf75834ce414bed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361201
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42928}
Based on the results of the experiment (b/335129329).
Bug: webrtc:15827, b/320629637, b/335129329, chromium:329396373
Change-Id: I1599f4c1be79ee3385aac1ff345168982c8278f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42895}
Before this cl, ReadyToSend signaled false if sending a packet failed and transport->GetError() returns ECONN.
ECONN may be reported by the TCP connection (TcpConnection) if the remote closed the connection. TcpConnection will attempt to reconnect and should change the writable state if it fail.
Changing the state in the context of sending packets may cause recursive
calls and seems to cause problems with incorrect states.
It is simpler if RtpTransport::SendPacket ignore these failures and
upper layers treat these lost packets similar to if the packets had been
lost via UDP.
For safety, this change can be reverted by field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/.
Bug: webrtc:361124449 b/359989715
Change-Id: I8e7016dfb4301862286215c4512aa8ac03a16685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42868}
Controlled via added field trial WebRTC-ElasticBitrateAllocation.
Bug: webrtc:350555527
Change-Id: If57552144bd4a50421d618fd8bdab31d7c4afc35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359506
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#42834}
this is flipping
WebRTC-PermuteTlsClientHello
to a killswitch in the SSLStreamAdapter used for DTLS.
BUG=webrtc:42225803
Change-Id: I942851c474ec5e723c5b6c9f6206e7eafbe80ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42676}
Current thresholds were tuned to guarantee no buffer overshoot in an extreme scenario (encoding a high complexity video in a low bitrate).
Bug: b/337757868, webrtc:351644568
Change-Id: I832b2564af6f18f06550338cc9b3618f8acdf831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356580
Reviewed-by: Dan Tan <dwtan@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42620}
Behind a flag, the new behavior changes how the "media rate" utilization
is calculated:
* Instead of per spatial & temporal layer, it's per spatial layer only.
* Overshoot is compared to real target vs adjusted target.
* Window takes quite periods/frame drops more into consideration.
This should lead to less push-back when not network constrained and
complex content is used causing bursty behavior.
Bug: b/349561566
Change-Id: I402e6531183493c963fec48ae363ce0b859b396a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356480
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42593}
Add default values and a static Create() function that determines
the parameters to use based on the specified codec and potential
field trial overrides.
Bug: chromium:328598314
Change-Id: I7a9331a1fd0ed4bd258788760592ea84e535e43b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355903
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42567}
This reverts commit 86ff48adae.
Reason for revert: Speculative revert due to failing downstream tests
Original change's description:
> Rewrite simulcast config to equivalent SVC for vp9 simulcast
>
> This allows to utilize libvpx optimizations considerably improving performance.
> The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
>
> This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
>
> Bug: webrtc:347737882
> Change-Id: Ic48316ad597700ed07e594d592413cf84b6b20d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355003
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42554}
Bug: webrtc:347737882
Change-Id: Ib84c9c0e20763348abfae838f2fb1aff31581a55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355943
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42564}
This allows to utilize libvpx optimizations considerably improving performance.
The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
Bug: webrtc:347737882
Change-Id: Ic48316ad597700ed07e594d592413cf84b6b20d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355003
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42554}
The experiment has been disabled for several years and the code
is not maintained.
Bug: webrtc:42221141
Change-Id: I631e4bd476ca01eb5312d4077c9467e77c42ff78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351143
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42364}
This was a kill-switch for frame dropping in AV1 encoder. The frame dropping was enabled in June 2023. Since we have not heard about about any issues related to the frame dropping, we can remove the field trial.
Bug: webrtc:42225542
Change-Id: I4b2f1d5ff61e4ae3a4a7fc6711bb83f7d522fc6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42241}
Support for parsing the packet is gated behind field trial
WebRTC-RFC8888CongestionControlFeedback/Enabled/.
Bug: webrtc:15368
Change-Id: Ib4478e821fe5a43510af5131543e7861cf54d901
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348664
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42215}
If pacing rate, (current loss based bwe * pacing factor) is larger than the current upper link capacity estimate, reduce pacing factor to max of current bwe and upper link capacity.
Bug: webrtc:42220543
Change-Id: I5246da1f38530f8d411e7314adaa8651fc848f48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349601
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42210}
Implements a two-pass approach to packetization which creates
packets of an even size similar to RtpPacketizer::SplitAboutEqually.
This improves the bandwidth estimation.
The algorithm does a first pass with the existing packetizer, then
iterates through the resulting packet sizes and sums up the bytes left unused in each packet.
It then calculates a new maximum packet length as
configured_max_packet_len - ((unused_bytes - packets + 1) / packets)
adjusts for the overhead and re-runs the packetization algorithm.
For example, a list of OBUs with sizes
{1206, 1476, 1431}
currently gets packetized greedily as payload sizes
{1200, 1200, 1200, 523}
With this change, it gets packetized as
{1032, 1032, 1032, 1028}
This change is guarded by the field trial
WebRTC-Video-AV1EvenPayloadSizes
which is acting as a rollout flag.
BUG=webrtc:15927
Co-authored-by: Shyam Sadhwani <shyamsadhwani@meta.com>
Change-Id: I4f0b3c27de6f06104908dd769c4dd1f34115712c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348100
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42203}
This migration was done semi-automatically. I didn't manage to find any
corresponding bug ID for chromium:413437 nor chromium:949536 in the new
issue tracker. Since these are policy-exempt anyway I opted for setting
the ID to NO_BUG and leaving a comment with the old ID.
Bug: None
Change-Id: If2d212ba554e40c42193b51f62a7da8a7f783d41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349267
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42190}
the rollout has happened a while ago with no issues requiring the use
of the killswitch
BUG=chromium:40066610
Change-Id: I2c8148976a1da219ebbfbe6908224b6384348194
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348823
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42164}
Instead, PeerConnectionFactoryDependencies.network_controller_factory is
used if it exists.
Bug: webrtc:8415
Change-Id: I37d5cc7325072bf1d87993e53949f1b97c277f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347860
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42120}
The field trial has been default on for ages. This CL removes it.
Bug: b/40200151
Change-Id: I171f663a3e725b856238b14b26d083f6684586e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347621
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42080}
This CL updates RtpVideoStreamReceiver2 to use H26xPacketBuffer for
H.264 and H.265 packets. H.264 specific fixes are moved to
H26xPacketBuffer as well.
H26xPacketBuffer is behind field trial WebRTC-Video-H26xPacketBuffer.
Bug: webrtc:13485
Change-Id: I1874c5a624b94c2d75ce607cf10c939619d7b5b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42062}
