On NetEq level latency corresponds to delay and two terms can be used interchangeably here.
In order to implement latency constraint we need to provide a range of possible values which should be constant. See getCapabilities() here:
https://www.w3.org/TR/mediacapture-streams/#dfn-applyconstraints-algorithm
Lowest possible value accepted value is constant and equals 0. But because |packet_len_ms_| and |maximum_delay_ms_| may be updated during live of DelayManager upper bound is not constant. Moreover, due to change in |packet_len_ms_| the |minimum_delay_ms_| which was valid when its was set may be considered invalid later on.
To circumvent that and provide constant range for capabilities we separate base minimum delay and minimum delay. ApplyConstraints algorithm will set base minimum delay. Base minimum delay will act as recommendation for lower bound of minimum delay and will be used to limit target delay. If user sets base minimum delay through ApplyConstraints which is bigger than currently
possible maximum (e.g. bigger than NetEq maximum buffer size in milliseconds) then base minimum delay will be clamped to currently possible maximum to match user's intentions as best as possible.
Note, that we keep original behavior when minimum_delay_ms_ (effective_minimum_delay_ms after this CL) in LimitTargetLevel method may be above upper bound due to changing packet audio length.
Bug: webrtc:10287
Change-Id: I06b8f5cd3fd1bc36800af0447f91f7c4dc21a766
Reviewed-on: https://webrtc-review.googlesource.com/c/121700
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26666}
This quality boost means that we sometimes drop a _lot_ of frames in the
base layer. It also interacts poorly with the bitrate adjuster since
even if frames are dropped they are often over-sized.
The setting still leaves the current behavior as default, but can be
changed using the WebRTC-VideoRateControl field trial.
Bug: webrtc:10155
Change-Id: I1a92ec69bab61b5148fe9d8bc391ac5ee1019367
Reviewed-on: https://webrtc-review.googlesource.com/c/122840
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26659}
Set padding bit if the TransportFeedback packet contains zero padding.
Also write number of padding elements at the last position of the packet.
Bug: webrtc:10263
Change-Id: I8d17bc0e889f658ac383dec64ddcb95d438c9702
Reviewed-on: https://webrtc-review.googlesource.com/c/122240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26646}
The lock is unnecessary and potentially unsafe:
1) All gain_control accesses in AudioProcessingImpl happen - and are intended to happen - while holding the crit_capture_ lock, and all external API calls take the same lock once inside GainControlImpl.
2) If ProcessCaptureStreamLocked (locked by crit_capture) calls a gain_control function that takes crit_render, the mandated locking order (render before capture) is violated and we might get a deadlock with the render thread.
Bug: b/123456404
Change-Id: Id7a888827e347e5e1d50e2f87d90e8b68f52b7b8
Reviewed-on: https://webrtc-review.googlesource.com/c/122087
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26637}
rtx packet may have addition extension (mid) and may use different
header size for extension (e.g. if repaired rtp stream id registered
to larger id than rtp stream id)
As a result rtx packet size calculation as orginial size + 2 bytes in
some scenarious may be incorrect. This chenage avoids crash in that cases.
Bug: None
Change-Id: I620d95e0592d6bdac0d3623b2675a49fc2177580
Reviewed-on: https://webrtc-review.googlesource.com/c/122180
Reviewed-by: Erik Varga <erikvarga@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26635}
We are about to split RtpGenericFrameDescriptorExtension
into v00 and v01. Allow downstream projects to refer to
RtpGenericFrameDescriptorExtension00 now, so that we may later
delete references to RtpGenericFrameDescriptorExtension
without breaking their build.
Bug: webrtc:10214
Change-Id: I45528699bf7d8cc6c22c22a601f248cca2ba6c90
Reviewed-on: https://webrtc-review.googlesource.com/c/121769
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26588}
Also refactor most of the RtpSender tests to not use the frame-level
method RTPSender::SendOutgoingData.
Bug: webrtc:7135
Change-Id: I9b0af6aa45e9b908d8197e48b143fede7e2804c7
Reviewed-on: https://webrtc-review.googlesource.com/c/121461
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26577}
Currently, if the last packet was reordered (e.g. due to retransmission) then the next packet's inter-arrival time will be estimated incorrectly due to the jump in sequence numbers. This change prevents that by not resetting the stopwatch on reordered packets.
This will also better estimate inter-arrival times when we have multiple reordered packets in a burst. Currently we would only measure the iat of the first reordered packet correctly and not the ones coming after it.
There is a slight risk introducing this: If we would receive an out of order packet far into the future (in sequence numbers) and then continue getting packets in the normal order, then we would not update the current sequence number for these and incorrectly estimate their inter-arrival times since they would all be considered reordered.
Change-Id: Ic938a37cbddf1cb9c30b610218f56794568d3d01
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/119949
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26572}
Configurability via SetSelectiveRetransmissions was deleted in
https://webrtc-review.googlesource.com/c/119920.
Delete constants kRetransmitFECPackets and kRetransmitAllPackets,
which are never enabled in production code. Also move the declaration
of RetransmissionMode from rtp_rtcp_defines.h to rtp_sender_video.h,
to reduce visibility to applications.
Bug: None
Change-Id: I70dcf7532aa3415a2449d8d807c500c1f149bf6d
Reviewed-on: https://webrtc-review.googlesource.com/c/120053
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26570}
This CL is a manual spin-off of [1], which tried to apply clang-tidy's
performance-move-const-arg [1] to the WebRTC codebase.
Since there were some wrong fixes to correct, this CL lands a few
different fixes, like adding a constructor overload to take an rvalue
reference or remove 'const' to make std::move effective.
[1] - https://webrtc-review.googlesource.com/c/src/+/120350
[2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html
Bug: webrtc:10252
Change-Id: I42a777247fee2cb788efcd7c2035148330056b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/120928
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26553}
With the current implementation, whenever we are toggling between
sending/not sending retransmissions, the BitrateAllocator will
toggle the total requested max bitrate that is signalled to the
probing mechanism. The result is that spurious probes are sent
mid-call, at |max_bitrate| and |2*max_bitrate|. This behaviour
is undesirable, and thus removed in this CL. Instead, whenever
the allocation limits actually change, we produce a single
set of probes at |max_bitrate| and |2*max_bitrate|.
This CL does not change how the BitrateAllocator hysteresis is
accounting for protection, since it does not relate to the
spurious probes.
Bug: webrtc:10275
TBR: sprang@webrtc.org
Change-Id: Iab3a690a500372c74772a8ad6217fb838af15ade
Reviewed-on: https://webrtc-review.googlesource.com/c/120808
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26544}
GetStreamCaps can return S_FALSE, which is not caught by FAILED().
Instead it is necessary to compare for equality with S_OK.
Bug: webrtc:10267
Change-Id: I4113ef5544f830f5cb5f09f143dff43129993d3d
Reviewed-on: https://webrtc-review.googlesource.com/c/120644
Commit-Queue: Dan Minor <dminor@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26539}
Digging in the git history, I see one reference to this table, deleted
in 2011. And reference to the header file disappeared in the cleanup cl
https://webrtc-review.googlesource.com/c/src/+/106280
Bug: None
Change-Id: Iab8cf407a5606e7c28f798f933ff57da0de8d1cc
Reviewed-on: https://webrtc-review.googlesource.com/c/120962
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26537}
This is first step to allow to set latency
from client code in Chromium.
Existing minimum latency hasn't been used because it can clash
with video syncronization code.
Bug: webrtc:10287
Change-Id: Ia38906506069a1abfa01698dc62df283fc15cfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/121423
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26536}
This CL continues the work began by CL #119958, extending it
to ScreenshareLayers.
Bug: webrtc:10249
Change-Id: I59d0c062a93b288007977e00aa3a2e0929509e0c
Reviewed-on: https://webrtc-review.googlesource.com/c/120042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26526}
Update RtpSenderAudioTest to call methods on RTPSenderAudio rather
than RTPSender, when possible. In particular, avoid
RTPSender::SendOutgoingData. Drop parameterization on the
WebRTC-SendSideBwe-WithOverhead field trial, since that appears
unrelated to these tests.
Also delete some unused parts of the RtpSender test.
Bug: webrtc:7135
Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0
Reviewed-on: https://webrtc-review.googlesource.com/c/120920
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26516}
Prior to this CL, when software VP8 encoding was done with one temporal
layer, instead of only predicting from the latest frame, the code
allowed the encoder to reference the latest key frame as well.
This improves quality for the few frames immediately after
the key frame, but is not useful for later frames, which diverge
significantly from the key frame. However, the cost of producing
the prediction from more than one reference is incurred by all frames.
My measurements of the effect of this show an improvement
in CPU utilization of 5%-13% when this is not done.
foreman_352x288, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(566.187, 570.012, 575.665) = 570.621
send_avg_qp: 45.36
send_avg_psnr: 37.13
Post-change:
send_enc_speed_fps: avg(633.188, 604.694, 623.232) = 620.371
send_avg_qp: 45.88
send_avg_psnr: 37.0749
Improvement in send_enc_speed_fps: 8.71%
foreman_480x272, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(481.244, 486.971, 487.322) = 485.179
send_avg_qp: 48.9
send_avg_psnr: 37.6217
Post-change:
send_enc_speed_fps: avg(521.651, 499.416, 511.551) = 510.872
send_avg_qp: 48.88
send_avg_psnr: 37.6094
Improvement in send_enc_speed_fps: 5.29%
news_352x288, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(699.407, 697.837, 699.49) = 698.9113333
send_avg_qp: 24.15
send_avg_psnr: 40.9551
Post-change:
send_enc_speed_fps: avg(758.526, 768.104, 757.232) = 761.2873333
send_avg_qp: 23.9833
send_avg_psnr: 40.9697
Improvement in send_enc_speed_fps: 8.92%
Bridge_180x320_15 (video of brandtr@ from Google), 15fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(454.757, 450.399, 446.812) = 450.656
send_avg_qp: 17.6771
send_avg_psnr: 39.9267
Post-change:
send_enc_speed_fps: avg(500.014, 513.316, 513.613) = 508.981
send_avg_qp: 17.6837
send_avg_psnr: 39.9137
Improvement in send_enc_speed_fps: 12.94%
Bug: webrtc:10281
Change-Id: If02736e1535c5f46689fd42b657e35a1e1f64d6d
Reviewed-on: https://webrtc-review.googlesource.com/c/120904
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26511}
With a recent change, the webrtc::DesktopFrame is created and
initialized with 0. So it's not necessary to call memset() to
the frame again any more.
See https://webrtc-review.googlesource.com/c/src/+/97184/
Bug: webrtc:9703
Change-Id: I27d096058ead075765f4c49bf60cb8d725da1afd
Reviewed-on: https://webrtc-review.googlesource.com/c/120700
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26504}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
Calculate packet exit times "just in time" rather than at send time.
This allows changing bandwidth with packets in the queue being reflected
correctly.
Bug: webrtc:10265
Change-Id: I5a38663def4d2bfee64164f9ae62bc61277064bb
Reviewed-on: https://webrtc-review.googlesource.com/c/120403
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26487}
Moved from RtpSender to RtpSenderVideo, since currently the
PlayoutDelay extension is used for video only, and configured via
RTPVideoHeader.
Bug: webrtc:7135
Change-Id: Idfcc90cefea83e40a4e79164d7914cdcd50e41fe
Reviewed-on: https://webrtc-review.googlesource.com/c/120357
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26484}
Will focus on delivering model based controller instead.
Bug: webrtc:9718
Change-Id: I5df82424469c577f3c170758e0db64e3e1aa7705
Reviewed-on: https://webrtc-review.googlesource.com/c/120607
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26478}
There's now one const method PlayoutDelayToSend to produce the delay
values to insert into outgoing packets, and two update methods,
OnSentPacket, and OnReceivedAck, to observe outgoing packets and acks,
respectively.
Bug: webrtc:7135
Change-Id: I07498c30f0de87ae0113f7e2eb6357a091a1f0af
Reviewed-on: https://webrtc-review.googlesource.com/c/120603
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26474}
