Setting the transport cc flag was only possible post creation for
audio receive streams, while video receive streams need to be recreated.
This CL moves the setter for transport_cc() to where the getter is and
adds boiler plate implementations for the video streams. For audio
streams this splits "SetUseTransportCcAndNackHistory" into two methods,
SetTransportCc and SetNackHistory.
Bug: none
Change-Id: Idbec8217aef10ee77907cebaecdc27b4b0fb18e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264443
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37038}
This changes FlexfecReceiveStreamImpl so that instead of holding on to
the entire config structure, the state is broken down into member
variables whose constness and thread access can be individually set.
Bug: none
Change-Id: I497b5816d40678774dee76d8a97012e8539629b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263723
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37027}
To make it usable in tests without depending on all of CallTest.
Bug: None
Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36932}
This will be used as a fall-back when the encoder adapter doesn't
provide any dependency structure. This ensures we can always generate a
dependency descriptor RTP header extension for VP8.
Before, when switching between encoder adapters where the old one
generated a dependency structure but the new one didn't we had to make
sure the structure was cleared so that packets weren't sent with the
dependency structure from the previous adapter. This will not be a
problem anymore since the new adapter will use the simulated dependency
structure.
Bug: b/227749056
Change-Id: I8463c48a9dcde4b8d32c519819dd8a92acd8e43b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262765
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36930}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
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@{#36922}
Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.
To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>
Deprecates field trial "WebRTC-Bwe-ProbingConfiguration"
Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36911}
This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042
This is a reland of commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca
with an additional fix in Patchset 2. Another problem turned out to be
in RTCPReceiver, which is fixed in:
https://webrtc-review.googlesource.com/c/src/+/262663
Bug: webrtc:11993
Change-Id: I63c7cf62a6dd50f88b491fea3ba866697552ef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262665
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36907}
This reverts commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca.
Reason for revert: Checking if this is blocking the Chromium autoroller.
Original change's description:
> Update local_ssrc without needing to recreate video streams.
>
> This is comparable to this change done previously for for audio streams:
> https://webrtc-review.googlesource.com/c/src/+/222042
>
> Bug: webrtc:11993
> Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36876}
Bug: webrtc:11993
Change-Id: I3a8d2f6a7e89b6784754d8e891a4e01479807c2d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262422
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36892}
The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36890}
GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
Instead offer accessors for the specific config values from the struct
that are needed at different times. The remote_ssrc and rtx_ssrc
properties maybe accessed from any thread, other properties have
stricter requiremets.
Bug: webrtc:11993
Change-Id: I3ff8527b13452c773fae1b2574f1e3fd2583b481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261319
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36823}
This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.
Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}
`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.
Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}
