This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
1. GetTransportFeedbackVector will now return a vector which also explicitly states lost packets.
2. The returned vector is unsorted (uses default order - by sequence number). It's up to the users to sort otherwise, if they need a different order.
BUG=None
Review-Url: https://codereview.webrtc.org/2707383006
Cr-Commit-Position: refs/heads/master@{#17114}
In this CL:
- Add message BweProbeCluster and BweProbeResult to rtc_event_log.proto.
- Add corresponding log functions to RtcEventLog.
- Add optional field |probe_cluster_id| to RtpPacket message and added
an overload function to log with this information.
- Propagate the probe_cluster_id to where RTP packets are logged.
BUG=webrtc:6984
Review-Url: https://codereview.webrtc.org/2666533002
Cr-Commit-Position: refs/heads/master@{#16857}
In order to not make this CL too large I have broken it down into at least two
steps. Previous CL: https://codereview.chromium.org/2628563003/
webrtc::PacedSender::Process <--- previous CL start here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- previous CL end here, this Cl start here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- this CL end here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2708873003
Cr-Commit-Position: refs/heads/master@{#16796}
Rename loss based and delay based bwe updates in proto (and correspondingly in the C++ code).
BUG=webrtc:6423
Review-Url: https://codereview.webrtc.org/2705613002
Cr-Commit-Position: refs/heads/master@{#16719}
In order to not make this CL too large I have broken it down into at least two steps. In this CL we only propagate the pacing information part of the way:
webrtc::PacedSender::Process <--- propagate from here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- to here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- goal is to propagte it here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2628563003
Cr-Commit-Position: refs/heads/master@{#16664}
The main difference to the old computation is that the expected number of packets during an interval is now computed as the change in highest sequence number encountered, rather than the sequence number difference between the first and last packet in the interval.
BUG=webrtc:7046
Review-Url: https://codereview.webrtc.org/2656333002
Cr-Commit-Position: refs/heads/master@{#16361}
Reason for revert:
Downstream project relied on changed struct.
Transition made possible by https://codereview.webrtc.org/2655243006/.
Original issue's description:
> Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ )
>
> Reason for revert:
> Breaks internal downstream project.
>
> Original issue's description:
> > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
> >
> > Prior to this CL, received RTX (associated) payload types were only configured
> > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> > SSRC was set up.
> >
> > After this CL, the RTX (associated) payload types are set in
> > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> > that is the code path that sets other SSRCs.
> >
> > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> > We remove the possibility for each video payload type to have an associated
> > specific RTX SSRC. Although the config previously allowed for this, all payload
> > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> > did not support multiple SSRCs. This change to the config struct should thus not
> > have any functional impact. The change does however affect the RtcEventLog, since
> > that is used for storing the VideoReceiveStream::Configs. For simplicity,
> > this CL does not change the event log proto definitions, instead duplicating
> > the serialized RTX SSRCs such that they fit in the existing proto definition.
> >
> > BUG=webrtc:7011
> >
> > Review-Url: https://codereview.webrtc.org/2646073004
> > Cr-Commit-Position: refs/heads/master@{#16302}
> > Committed: fe2bef39cd
>
> TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2649323010
> Cr-Commit-Position: refs/heads/master@{#16307}
> Committed: e4974953ceTBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
# NOTREECHECKS=true
# NOTRY=true
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2654163006
Cr-Commit-Position: refs/heads/master@{#16322}
Reason for revert:
Breaks internal downstream project.
Original issue's description:
> Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
>
> Prior to this CL, received RTX (associated) payload types were only configured
> when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> SSRC was set up.
>
> After this CL, the RTX (associated) payload types are set in
> WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> that is the code path that sets other SSRCs.
>
> As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> We remove the possibility for each video payload type to have an associated
> specific RTX SSRC. Although the config previously allowed for this, all payload
> types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> did not support multiple SSRCs. This change to the config struct should thus not
> have any functional impact. The change does however affect the RtcEventLog, since
> that is used for storing the VideoReceiveStream::Configs. For simplicity,
> this CL does not change the event log proto definitions, instead duplicating
> the serialized RTX SSRCs such that they fit in the existing proto definition.
>
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2646073004
> Cr-Commit-Position: refs/heads/master@{#16302}
> Committed: fe2bef39cdTBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2649323010
Cr-Commit-Position: refs/heads/master@{#16307}
Prior to this CL, received RTX (associated) payload types were only configured
when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
SSRC was set up.
After this CL, the RTX (associated) payload types are set in
WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
that is the code path that sets other SSRCs.
As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
We remove the possibility for each video payload type to have an associated
specific RTX SSRC. Although the config previously allowed for this, all payload
types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
did not support multiple SSRCs. This change to the config struct should thus not
have any functional impact. The change does however affect the RtcEventLog, since
that is used for storing the VideoReceiveStream::Configs. For simplicity,
this CL does not change the event log proto definitions, instead duplicating
the serialized RTX SSRCs such that they fit in the existing proto definition.
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2646073004
Cr-Commit-Position: refs/heads/master@{#16302}
And delete the method CongestionController::packet_router.
BUG=None
Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
Reason for revert:
Unfortunately, this change breaks internal projects. Specifically the change to the CongestionController interface means anything implementing it will be forced to change in lock-step.
Original issue's description:
> Pass time constanct to bwe smoothing filter.
>
> BUG=webrtc:6443, webrtc:6303
>
> Committed: https://crrev.com/9abbf5ae4ec7d688a9b4aa03a405f3faadb74b90
> Cr-Commit-Position: refs/heads/master@{#15266}
TBR=minyue@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443, webrtc:6303
Review-Url: https://codereview.webrtc.org/2532993002
Cr-Commit-Position: refs/heads/master@{#15272}
This helps a lot to avoid reducing the bitrate too quickly when there's a short period of very few packets delivered, followed by the rate resuming at the regular rate. It specifically avoids the BWE going down to super low values as a response delay spikes.
BUG=webrtc:6566
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2422063002 .
Cr-Commit-Position: refs/heads/master@{#14802}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
unnecessary dependencies.
Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
BUG=webrtc:6427, webrtc:6422
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2378103005 .
Cr-Commit-Position: refs/heads/master@{#14452}
Reason for revert:
Caused issues with webrtc_perf_tests on build bots.
Original issue's description:
> Fix race / crash in OnNetworkRouteChanged().
>
> To achieve this some refactoring was done to make it possible to synchronize
> access to the DelayBasedBwe in TransportFeedbackAdapter:
> - The callback was removed from DelayBasedBwe, it now instead returns its
> result.
> - TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
> unnecessary dependencies.
>
> Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
>
> BUG=webrtc:6427, webrtc:6422
>
> Committed: https://crrev.com/fd0d42669204e6dd92a60736bca7ae0196663024
> Cr-Commit-Position: refs/heads/master@{#14430}
TBR=terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6427, webrtc:6422
Review-Url: https://codereview.webrtc.org/2377303002
Cr-Commit-Position: refs/heads/master@{#14433}
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
unnecessary dependencies.
Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
BUG=webrtc:6427, webrtc:6422
Review-Url: https://codereview.webrtc.org/2366333003
Cr-Commit-Position: refs/heads/master@{#14430}
This uses the audio level values that are stored in the RTP header extension.
BUG=webrtc:4741
Review-Url: https://codereview.webrtc.org/2346363003
Cr-Commit-Position: refs/heads/master@{#14352}
Uses generic functions to plot packet sizes, sequence number delta and bitrate per SSRC. Also detects and prints warnings if delay differences seem unrealistic.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2234883002
Cr-Commit-Position: refs/heads/master@{#13872}
