With this change, all production callers of BaseChannel::transport_name()
will be making the call from the right thread and we can safely delegate
the call to the transport itself. Some tests still need to be updated.
This facilitates the main goal of not needing synchronization inside
of the channel classes, being able to apply thread checks and eventually
remove thread hops from the channel classes.
A downside of this particular change is that a blocking call to the
network thread from the signaling thread inside of RTCStatsCollector
needs to be done. This is done once though and fixes a race.
Bug: webrtc:12601, webrtc:11687, webrtc:12644
Change-Id: I85f34f3341a06da9a9efd936b1d36722b10ec487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33775}
StatsCollector::ExtractSessionInfo was run fully on the signaling thread
and several calls were being made to methods that need to run on the
network thread.
Additionally, BaseChannel::transport_name() was being read directly
on the signaling thread (needs to be read on the network thread).
So with shifting the work that needs to happen on the network thread
over to that thread, we now also grab the transport name there and
use the name with the work that still needs to happen on the signaling
thread.
These changes allow us to remove Invoke<>() calls to the network thread from
callback functions implemented in PeerConnection:
* GetPooledCandidateStats
* GetTransportNamesByMid
* GetTransportStatsByNames
* Also adding a correctness thread check to:
* GetLocalCertificate
* GetRemoteSSLCertChain
Because PeerConnection now has a way of knowing when things are
or have been uninitialized on the network thread, all of these
functions can exit early without doing throw away work.
Additionally removing thread hops that aren't needed anymore from
JsepTransportController.
Using the RTC_LOG_THREAD_BLOCK_COUNT() macro in GetStats, the number
of Invokes (when >1), goes down by 3. Typically from 8->5, 7->4, 6->3.
Bug: webrtc:11687
Change-Id: I06ab25eab301e192e99076d7891444bcb61b491f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214135
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33656}
This reduces the degree of interdependency among modules related
to the PeerConnection class, and makes it easier to isolate inappropriate
external dependencies.
Bug: webrtc:11967
Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32235}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
Done in preparation for some threading changes that would be quite
messy if implemented with the class as-is.
This results in some code duplication, but is preferable to
one class having two completely different modes of operation.
RTP data channels are in the process of being removed anyway,
so the duplicated code won't last forever.
Bug: webrtc:9883
Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31691}
This removes code from DataChannelController that exposes
an internal vector of data channels and puts the onus of
returning stats for a data channel, on the data channel
object itself. This will come in handy as we make threading
changes to the data channel object.
Change-Id: Ie164cc5823cd5f9782fc5c9a63aa4c76b8229639
Bug: webrtc:11547, webrtc:11687
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177244
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31533}
This patch adds new enum values for different types of cellular
connections.
The new costs are currently blocked when sending to remote,
(so that arbitrary network switches does not starts occurring).
The end-game for this series to be able to distinguish between
different type of cellular connections in the ice-layer (e.g when
selecting/switching connections).
BUG: webrtc:11473
Change-Id: I587ac8fdff4f6cdd0f8905f327232f58818db4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172582
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30970}
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.
The new stat will not be reported unless it is GT 0.
Reporting of decoding_codec_plc events
Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.
This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
the unsigneled RtpReceiver track ID for both Plan B and Unified
Plan.
2) Removes a couple methods on PeerConnection that were only used by
the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
the code easier to understand.
Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27324}
updates the timestamp on the local and remote candidate stats for consistency
with other places. This also makes the graphs on chrome://webrtc-internals
work (even though most values don't update so showing graphs is not meaningful)
BUG=chromium:937833
Change-Id: I3267dd7a5f5a887dcd0756137077b8f02c201905
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128765
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27248}
