Change the way the forget factor converge to the steady state so that we don't overemphasize the first packets received.
The logic is controlled by the delay histogram field trial which has an added parameter to control if emphasis should be even (c=1, default) or put on later packets (c>1) until we reach our steady state forget factor.
Bug: webrtc:10411
Change-Id: Ia5d46c22d1a4a66994652f71c8cde664362bfacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137050
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28039}
WebRTC combines VP9 SVC spatial layer frames into superframe and passes
it to a decoder. The chromium HW VP9 decoder (wrapper) needs to know
location of each spatial layer frame in the frame buffer. To provide
decoder with such information this CL:
- Adds Set/SpatialLayerFrameSize methods to EncodedImage.
- Sets size of each spatial layer frame on superframe at assembly stage.
Bug: webrtc:10495
Change-Id: I68c3c0d668c67dfa1740e004059d860dd98f67f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28032}
This reverts commit a8ae407a480a2a9982eecf9e3a9b10da5373cd9a.
Reason for revert: This CL incorrectly affects non-experiment branch. A new CL affecting only the experiment will be uploaded.
Original change's description:
> Add ability to cap the video jitter estimate to a max value.
>
> Bug: webrtc:10572
> Change-Id: I21112824dc02afa71db61bb8c2f02723e8b325b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133963
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27744}
TBR=stefan@webrtc.org,mhoro@webrtc.org
Bug: webrtc:10572
Change-Id: I4af334168ca70ecfae7fd18fc7c852819a98d866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138063
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28025}
RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.
Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28020}
Simplifying the code to better fit with how it is used.
Bug: webrtc:9883
Change-Id: I2bd52f26b829413e516dee4f551cf36574275019
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136681
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27994}
A later change will allow them to differ.
Bug: webrtc:10631
Change-Id: I4e13f41980261990b3bbbc6897cd754369265ca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137046
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27991}
This is a standardized metric. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
It is meant to replace the legacy googBucketDelay. The average
packet delay over any interval can be calculated as the delta
totalPacketSendDelay divided by the delta packetsSent between two
calls to getStats().
Bug: webrtc:10506
Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27979}
This change allows NetEq to reach preferred jitter buffer size much faster
for high target delays because it uses absolute units instead of relative ones
during computation of lower_limit.
More details can be found here:
https://docs.google.com/document/d/12qPMJYFhGXrA_o_nvz9VshpzAJX6aULxFig1fTzBzDI/edit
Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5
Bug: webrtc:10619
Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135745
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27970}
Ensure that frame_buffer_controller_ does not get assigned null
by the factory.
Bug: None
Change-Id: I84e141ae0390cd024863f88cdcdc79b8b13e7c64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137043
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27962}
The ReportBlockData contains information about a ReportBlock and
additional data such as RTT. This will be used for the calculation of
RTCRemoteInboundRtpStreamStats, see full picture here:
https://webrtc-review.googlesource.com/c/src/+/134107
ReportBlockData is a class version of the previously internal struct
RTCPReceiver::ReportBlockWithRtt.
- The new name makes sense even if we add more info to it, which will
be needed for future metrics.
- The new location is modules/rtp_rtcp/include/report_block_data.h.
The RTCPReceiver allows obtaining the ReportBlockData in two ways:
1. Using a ReportBlockDataObserver that is notified on receiving a
report block.
2. Using the GetLatestReportBlockData().
Both codepaths will be needed; video stats uses observers and audio
stats uses polling.
Further plumbing will be done in follow-up CLs.
Bug: webrtc:10455, webrtc:10456
Change-Id: Ic9e5b4f451b5f4b203efcd6fa3bbf9736487e1f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136584
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27961}
This is a reland of e779847fb6499ac2dc4757de8c625ac377e9d0d4
Original change's description:
> Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
>
> Also add explicit includes of rtc_base/string_utils.h in files depending on it.
>
> Bug: webrtc:6424
> Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27903}
Tbr: kwiberg@webrtc.org
Bug: webrtc:6424
Change-Id: Ic08d5d7fbc25ff89e4182d7c9cb3b0e8e356339a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135946
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27957}
webrtc::RealTimeClock::TimeInMilliseconds() and
rtc::TimeMillis() have for some time been backed by the same clock,
no need for adjustment.
Bug: None
Change-Id: I5962153d9f5aa5e58ccde26393c322972cb51d43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136808
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27939}
Reland with fixes.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27930}
This CL introduces three-value enum, in order to be able to distinguish
between send success, send failure, and invalid states such as missing
packet or invalid ssrc.
The behavior is unchanged in this CL, a follow-up will change the pacer
to not consume media budget on invalid states.
Bug: webrtc:8052,webrtc:8975
Change-Id: I1c9e2226f995356daa538d3d3cf44945f35e0133
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135165
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27923}
Add a PlayoutVolumeChange RuntimeSetting. Trigger an echo path change when the playout volume is changed.
Bug: webrtc:10608
Change-Id: I1e736b93c1865d08c7d2582f6fe00216c1e1f72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135746
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Fredrik Hernqvist <fhernqvist@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27913}
When backing down, the delay based estimator can still use acked bitrate instead of the last set estimate.
Original code was reviewed in:
https://webrtc-review.googlesource.com/c/src/+/113880
BUG=webrtc:10144
Change-Id: Ia6e2d6d7d05f88f7e51d61b6e37c61a89adccf8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135950
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27912}
Use the C++-style stdlib headers, add `std::` prefix, in order to avoid implicit casts to double.
Bug: None
Change-Id: I78d9caaee715be341d2480c6d5e769068966d577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133625
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27905}
Also add explicit includes of rtc_base/string_utils.h in files depending on it.
Bug: webrtc:6424
Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27903}
This reverts commit bd20c3f5ae1908a6680945c6f849373d8a268e32.
Reason for revert: chromium:961253
This CL is not the cause of the regression, but reverting it will make the reverting of the actual cause easier.
Original change's description:
> Rename configurations_ to vpx_configs_ in LibvpxVp8Encoder
>
> Bug: None
> Change-Id: I548a724f0fb81f46785517c90e527edc075e1476
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135040
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27836}
TBR=brandtr@webrtc.org,eladalon@webrtc.org
Bug: chromium:961253
Change-Id: I707337e0ce50f29f9cda7cf45500c11debace1a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135750
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27898}
This reverts commit 4fb12b0caec9faa57cfbceb0f86b0e10c32a0cc2.
Reason for revert: Breaks some asan chromium bots
Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
>
> Reland with fixes.
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
>
> Bug: webrtc:10460
> Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27828}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10460
Change-Id: I9c87a43a716622b389974cb8377f973573fc29a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135747
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27895}
The main purpose right now of this CL is to avoid the situation
where multiple retransmissions are queued for sending (normally after
network glitch with increased pacer queue length), and some of those
fail sending because the can't be retrieved from the packet history
due to too short time since last sent.
Bug: webrtc:8975, webrtc:10607
Change-Id: I9f6369d83f0b8208e5f57b2dc2fd3f2db7c6fea1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135164
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27884}
