With this cl, sending can be forced with field trial "WebRTC-RFC8888CongestionControlFeedback/force_send:true/"
In the future, ReceiveSideCongestionController::EnablSendCongestionControlFeedbackAccordingToRfc8888 if RFC 8888 has been negotiated.
Bug: webrtc:42225697
Change-Id: Ib09066aa89ca7b3fffc551da541090c69ab8d75f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352720
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42413}
Without this, packets may be sorted in the wrong order.
Bug: webrtc:42225697
Change-Id: Ib9a72cdc7cb8f7ef6ca1571d095a6474215a83f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352821
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42411}
Because it is flaky !?
Bug: webrtc:42225697, b/343600373
Change-Id: I74415a9b97e90c25807b55053fd549f335b863ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352820
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42408}
CongestionControlFeedbackGenerator collect receive time information about received
packets and sends feedback according to RFC8888
Bug: webrtc:42225697
Change-Id: I70b7f7322fd262f99f45fd56b6eb8630a11b30c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351543
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42404}
BWE logging has as far as I know know been used for a long time. RTC event logs are the prefered method of logging.
Removed since it causes some BUILD pain.
For debugging the metrics API https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/metrics/ can be used instead.
Bug: webrtc:343347276
Change-Id: I046b58d880faabfadbc22269b0392fdd644155fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352602
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42402}
This is done to better reflect the responsibility of the class.
The implementation implement a new interface FeedbackGeneratorInterface. The purpose of the interface is to allow a new implementation that supports RFC 8888.
Bug: webrtc:42225697
Change-Id: Id087dd7422abbcd6016693c076a65f4c4efd5712
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351280
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42366}
This makes Environment and thus field trials a required parameter, thus
RemoteBitrateEstimators member no longer need to fallback to the global field trial string.
Bug: webrtc:42220378
Change-Id: Ieb6ff442d5fde5fa9715573c758a7e078f0ceea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349922
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42314}
This ensure upper link capacity estimate upper limit an increase in
delay based estimate, but the delay based estimate is not decreased if
link capacity estimate decrease.
Bug: webrtc:10498, b/300868877
Change-Id: I87e76e2a869e6f721cc8fe9d422e0194371d4e45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327801
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41196}
Traditionally, we'd back off to 85% of the measured throughput in response to an overuse. However, this backoff doesn't appear to be sufficient to drain the queues in some low-bitrate scenarios, and the problem has gotten a bit worse with the RobustThroughputEstimator. (The new estimate looks more stable. The old estimator had more variation, the lowest points were lower, causing backoffs to lower rates.)
With this change, we back off to 0.85*thoughput-5kbps. The difference is negligible except at low bitrates.
Bug: webrtc:13402,b/298636540
Change-Id: I53328953c056b8ad77f6c7561d6017f171b2dfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321701
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40827}
BitrateTracker uses same implementation as RateStatistics, but provides api using Timestamp and DataRate types instead of plain numbers
Bug: webrtc:13756
Change-Id: Ie37fa58ede7590f870ec4376a64e7cf2c94431d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318841
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40697}
Start using RobustThoughputEstimator in DelayBasedBwe test in preparation for making it default.
Experiments has not showed significant metric changes. However, simulations has showed that RobustThroughputEstimator better follow the actually receive rate better. Especially during bursts of sent packets. Code is also simpler.
Bug: webrtc:13402 chromium:1411666
Change-Id: I83cfa1fc15486982b18cc22fbd0752ff59c1c1b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40644}
This is to ensure that a bad NetworkState estimate can not decrease BWE
unless an delay BW overuse has been detected.
Bug: webrtc:10489
Change-Id: Ic3a516345999eeba814200c2e295a19b347b2eb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317800
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@google.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40628}
Store Detectos in a map by value instead of by unccessary pointer
Bug: None
Change-Id: Iab9904aafca02d9f9ae6633c87de860a5bd62ac7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313621
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40499}
Allow absolute send time to go back in time as long as there has not been a large gap in arival time. Use the first packets arival time as time base.
Bug: b/282153758, webrtc:15230
Change-Id: I8663079ab9c202079bf8db303353918d46ba1d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308142
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40251}
Pass FieldTrialsView by const& to note it can't be null and doesn't need to outlive the constructor
In unittests use AimdRateControl object directly instead of through helpers
Use unit types (TimeDelta, DataRate) directly, reducing their conversion to plain numbers
Replace SimulatedClock with a single Timestamp now variable or constant
Bug: None
Change-Id: I147f85e629b4d8923aa19896ea211a6f9dca1e68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299260
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39707}
when there are lot of missing packets to report before next received packet
Bug: chromium:1426582
Change-Id: I6746294152d13e18120cdb173b11b245e5c803f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39698}
Values of these parameters are always the same and thus can be hardcoded
Bug: None
Change-Id: Ie19a1c6305d503ad2c92af503006a72b7981e178
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298622
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39637}
They are called only by ReceivedSideCongestionController that already
ensures all access is synchronized.
Bug: None
Change-Id: I0f87e24e3fbb0bd8f6ff679fb949d2373c554fba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269300
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39622}
relax DCHECK and explain when it previous version could be hit.
Use concise versions of the GetExtension functions.
Reduce scope of the `lock_`
Bug: None
Change-Id: Iafc570ffe7e5b2dcbdfe166b26b140f7959c28c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291711
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39246}
Today, behaviour is decided based on if transport sequence number v2 is
in the SDP answer. But it might be better to decide based on received
packets since it is valid to negotiate both extensions.
Another bonus With this solution is that Call does not need to know
about receive header exensions.
This is an alternative to https://webrtc-review.googlesource.com/c/src/+/291337
Bug: webrtc:7135
Change-Id: Ib75474127d6e2e2029557b8bb2528eaac66979f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291525
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39226}
PacketArrivalMap explicitly doesn't promise packet at the beginning
of it is received. Ensuring that property is wasteful
Bug: chromium:1382563
Change-Id: Ifc898b7ec2bc7a302af8dcfd233e0c598f62db95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290501
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39083}
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.
This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset
It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.
Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
That member is always created during construction and thus doesn't need
to be wrapped into unique_ptr and rechecked for beeing nullptr
Bug: None
Change-Id: I5d608c9b7bdfd8ec6e3296245359610ec1cf176c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285660
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38785}
There was a DCHECK in PacketArrivalMap::EraseTo() that the
seqeuence number that is used as argument has been received.
However, this is not necessarily the case since it's cleared upon
a request of a feedback report from the sender.
Bug: webrtc:14679
Change-Id: I908b4bf1f2a4355593f0a361e1733fc91527366d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283741
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38755}
Removed old disabled tests
enable test on android
Bug: webrtc:4711
Change-Id: Ic9adbdadc9e847bdf31b8be4ce116a3695499944
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284922
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38736}
When packet arrives with large gap majority of the time could be spend
in finding next received packet. Embedding such search into PacketArrivalMap
makes it faster
Bug: chromium:1373414
Change-Id: I2e0be0f2fc4ea96af081531d575a17c70b72b25b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279881
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38459}
replace std::deque implementation with a manually controlled circular buffer.
replace Timestamp validity check from 'IsInfinite()' accesser to cheaper comparison to zero.
These greatly increase PacketArrivalTimeMap::AddPacket perfomance when packet arrive with large sequence number gaps.
Bug: chromium:1349880
Change-Id: I6f4e814b1086ca9d0b48608531e3a387d9e542dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270564
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37722}
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}
Replace helper functions with the constant
Remove option to set min bitrate in RemoteBitrateEstimator as unused:
ReceivedSideCongestionController is the only user of the
RemoteBitrateEstimator interface, and it always sets the same value
right after construction that RemoteBitreateEstimators already use.
Bug: None
Change-Id: If179fdd72b1ded6ad1fd0a6dfffc97b302153322
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269383
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37613}
Inserting packet with zero arrival time may trigger inconsistent state in the internal map where packet sometimes treated as received, but sometimes treated as not received.
Bug: chromium:1346959
Change-Id: I0809e41a873103dcd62528358e64794c1d3cb28f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37609}
Return the bitrate estimate as DataRate type
Remove list of affected ssrcs as unused
Bug: None
Change-Id: Ie31dce591d861624736d834194f90eb6c93f70f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37397}
