Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.
Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
and move usages to webrtc::RefCountInterface
This CL also moves more stuff to webrtc:: and adds backwards
compatible aliases for them.
Bug: webrtc:42225969
Change-Id: Iefb8542cff793bd8aa46bef8f2f3c66a1e979d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353720
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42446}
This removes several references across the code base that depended on
the global string constants.
Bug: none
Change-Id: I007bd4b195c35261039f655f1a8f52e632c3691f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335320
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41578}
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
Track stats are roughly equal in size as the RTP stream stats which
are the largest objects making up the majority of the RTCStatsReport
size and scales with meeting size. Deleting track/stream reduces the
size in approximately half which should reduce performance overhead
and unblock code simplifications.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4517530
# Relevant bots already passed
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: Ib7bdb84c10459b42b829228d11876498e5227312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40129}
This reverts commit 7f16fcda0fd5bb625584b71311dd37b54c096136.
Reason for reland: Re-landing after addressing issues in downstream
code and hardening the ObserverAdapter from situations where attempted
usage of data channel proxies could occur after shutting down the
peer connection and terminating the network thread.
Original change's description:
> Revert "[DataChannel] Send and receive packets on the network thread."
>
> This reverts commit fe53fec24e02d2d644220f913c3f9ae596bbb2d9.
>
> Reason for revert: Speculative revert, may be breaking downstream project
>
> Original change's description:
> > [DataChannel] Send and receive packets on the network thread.
> >
> > This updates sctp channels, including work that happens between the
> > data channel controller and the transport, to run on the network
> > thread. Previously all network traffic related to data channels was
> > routed through the signaling thread before going to either the network
> > thread or the caller's thread (e.g. js thread in chrome). Now the
> > calls can go straight from the network thread to the JS thread with
> > enabling a special flag on the observer (see below) and similarly
> > calls to send data, involve 2 threads instead of 3.
> >
> > * Custom data channel observer adapter implementation that
> > maintains compatibility with existing observer implementations in
> > that notifications are delivered on the signaling thread.
> > The adapter can be explicitly disabled for implementations that
> > want to optimize the callback path and promise to not block the
> > network thread.
> > * Remove the signaling thread copy of data channels in the controller.
> > * Remove several PostTask operations that were needed to keep things
> > in sync (but the need has gone away).
> > * Update tests for the controller to consistently call
> > TeardownDataChannelTransport_n to match with production.
> > * Update stats collectors (current and legacy) to fetch the data
> > channel stats on the network thread where they're maintained.
> > * Remove the AsyncChannelCloseTeardown test since the async teardown
> > step has gone away.
> > * Remove `sid_s` in the channel code since we only need the network
> > state now.
> > * For the custom observer support (with and without data adapter) and
> > maintain compatibility with existing implementations, added a new
> > proxy macro that allows an implementation to selectively provide
> > its own implementation without being proxied. This is used for
> > registering/unregistering a data channel observer.
> > * Update the data channel proxy to map most methods to the network
> > thread, avoiding the interim jump to the signaling thread.
> > * Update a plethora of thread checkers from signaling to network.
> >
> > Bug: webrtc:11547
> > Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39760}
>
> Bug: webrtc:11547
> Change-Id: Id0d65594bf727ccea5c49093c942b09714d101ad
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300341
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Owners-Override: Andrey Logvin <landrey@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39764}
Bug: webrtc:11547
Change-Id: I47dfa7e7168be0cd2faab4f8f3ebf110c3728af5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300360
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39786}
This reverts commit fe53fec24e02d2d644220f913c3f9ae596bbb2d9.
Reason for revert: Speculative revert, may be breaking downstream project
Original change's description:
> [DataChannel] Send and receive packets on the network thread.
>
> This updates sctp channels, including work that happens between the
> data channel controller and the transport, to run on the network
> thread. Previously all network traffic related to data channels was
> routed through the signaling thread before going to either the network
> thread or the caller's thread (e.g. js thread in chrome). Now the
> calls can go straight from the network thread to the JS thread with
> enabling a special flag on the observer (see below) and similarly
> calls to send data, involve 2 threads instead of 3.
>
> * Custom data channel observer adapter implementation that
> maintains compatibility with existing observer implementations in
> that notifications are delivered on the signaling thread.
> The adapter can be explicitly disabled for implementations that
> want to optimize the callback path and promise to not block the
> network thread.
> * Remove the signaling thread copy of data channels in the controller.
> * Remove several PostTask operations that were needed to keep things
> in sync (but the need has gone away).
> * Update tests for the controller to consistently call
> TeardownDataChannelTransport_n to match with production.
> * Update stats collectors (current and legacy) to fetch the data
> channel stats on the network thread where they're maintained.
> * Remove the AsyncChannelCloseTeardown test since the async teardown
> step has gone away.
> * Remove `sid_s` in the channel code since we only need the network
> state now.
> * For the custom observer support (with and without data adapter) and
> maintain compatibility with existing implementations, added a new
> proxy macro that allows an implementation to selectively provide
> its own implementation without being proxied. This is used for
> registering/unregistering a data channel observer.
> * Update the data channel proxy to map most methods to the network
> thread, avoiding the interim jump to the signaling thread.
> * Update a plethora of thread checkers from signaling to network.
>
> Bug: webrtc:11547
> Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39760}
Bug: webrtc:11547
Change-Id: Id0d65594bf727ccea5c49093c942b09714d101ad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300341
Auto-Submit: Andrey Logvin <landrey@webrtc.org>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39764}
This updates sctp channels, including work that happens between the
data channel controller and the transport, to run on the network
thread. Previously all network traffic related to data channels was
routed through the signaling thread before going to either the network
thread or the caller's thread (e.g. js thread in chrome). Now the
calls can go straight from the network thread to the JS thread with
enabling a special flag on the observer (see below) and similarly
calls to send data, involve 2 threads instead of 3.
* Custom data channel observer adapter implementation that
maintains compatibility with existing observer implementations in
that notifications are delivered on the signaling thread.
The adapter can be explicitly disabled for implementations that
want to optimize the callback path and promise to not block the
network thread.
* Remove the signaling thread copy of data channels in the controller.
* Remove several PostTask operations that were needed to keep things
in sync (but the need has gone away).
* Update tests for the controller to consistently call
TeardownDataChannelTransport_n to match with production.
* Update stats collectors (current and legacy) to fetch the data
channel stats on the network thread where they're maintained.
* Remove the AsyncChannelCloseTeardown test since the async teardown
step has gone away.
* Remove `sid_s` in the channel code since we only need the network
state now.
* For the custom observer support (with and without data adapter) and
maintain compatibility with existing implementations, added a new
proxy macro that allows an implementation to selectively provide
its own implementation without being proxied. This is used for
registering/unregistering a data channel observer.
* Update the data channel proxy to map most methods to the network
thread, avoiding the interim jump to the signaling thread.
* Update a plethora of thread checkers from signaling to network.
Bug: webrtc:11547
Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39760}
* The pointer isn't needed for this notification. Arguably using
the internal id is more consistent with the stats code.
* Using the int makes it safer down the line to post the operation
from the network thread to the signaling thread rather than post
an object reference.
Bug: none
Change-Id: I1e9eb31d8386dca3feaa90ee3267ea98eb3e81ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299144
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39696}
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.
The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.
BUG=webrtc:14973
Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39497}
RTCStatsCollector internally keeps track of open data channels but
does not need (or want) to interact directly with those channels,
hence uintptr_t was used instead of pointers to the channel objects.
This changes that to use void* to avoid having to do the cast.
This is a follow-up action item to
https://webrtc-review.googlesource.com/c/src/+/295781
This CL also changes the container type:
std::set -> webrtc::flat_set
Bug: webrtc:12689
Change-Id: I13d3f4a41ef83dab38411193187e872b9d6d3cff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295871
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39468}
It turns out that there were several sigslot instances across data
channel, pc and stats classes that in practice only served as means
to update two counters in RTCStatsCollector. There's already a
notification path that's suitable.
This also fixes a case where the PC instance sat in the middle
of notifications from datachannels to the datachannel controller.
Bug: webrtc:11943
Change-Id: Ic60b76021584019f82085f6651230fe2fe82d465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39456}
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.
Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
This is a reland of commit 81aab488781c1a736c9d85ff1532631be2989523
See diff between Patch Set 1 and latest Patch Set.
The original CL broke this WPT[1] because getStats() with the receiver
as the selector stopped working in the event of unsignalled SSRCs due
to the receiver not knowing what the SSRC was.
This fix is to query media_channel_ for the unsignalled SSRC in the
event that the receiver does not know the SSRC.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html
Original change's description:
> Remove 'trackId' dependency in stats selector algorithm.
>
> In preparation for the deletion of deprecated 'track' stats, the
> stats selector algorithm needs to be rewritten not to use 'trackId'.
>
> This is achieved by finding RTP stats by their SSRC, as obtained via
> getParameters(). This unfortunately adds a block-invoke (in the sender
> case the block-invoke happens inside GetParametersInternal and in the
> receiver case the block-invoke is explicit at the calling place), but
> it can't be helped and it's just once per getStats() call and only if
> the selector argument is used.
>
> Bug: webrtc:14175
> Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38981}
Bug: webrtc:14175, webrtc:14811
Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39010}
making it clear what unit is being used.
BUG=webrtc:13756
Change-Id: I6354d35a8e02bb93a905ccf32cb0b294b4813e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289460
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39008}
This reverts commit 81aab488781c1a736c9d85ff1532631be2989523.
Reason for revert: external/wpt/webrtc/simulcast/setParameters-active.https.html is failing with this change
Original change's description:
> Remove 'trackId' dependency in stats selector algorithm.
>
> In preparation for the deletion of deprecated 'track' stats, the
> stats selector algorithm needs to be rewritten not to use 'trackId'.
>
> This is achieved by finding RTP stats by their SSRC, as obtained via
> getParameters(). This unfortunately adds a block-invoke (in the sender
> case the block-invoke happens inside GetParametersInternal and in the
> receiver case the block-invoke is explicit at the calling place), but
> it can't be helped and it's just once per getStats() call and only if
> the selector argument is used.
>
> Bug: webrtc:14175
> Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38981}
Bug: webrtc:14175
Change-Id: Id1cbe892250fe88bd6db0b47269bcefa346709b4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290502
Commit-Queue: Christoffer Jansson <jansson@google.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#38993}
In preparation for the deletion of deprecated 'track' stats, the
stats selector algorithm needs to be rewritten not to use 'trackId'.
This is achieved by finding RTP stats by their SSRC, as obtained via
getParameters(). This unfortunately adds a block-invoke (in the sender
case the block-invoke happens inside GetParametersInternal and in the
receiver case the block-invoke is explicit at the calling place), but
it can't be helped and it's just once per getStats() call and only if
the selector argument is used.
Bug: webrtc:14175
Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38981}
`cached_certificates_by_transport_` is used on the network thread, but
can be cleared from the signaling thread. To fix the race where clear
happens at the same time as stats collecting, a mutex is added.
This mutex should very rarely be contended in practise since
ClearCachedStatsReport() typically only happen during renegotiation
(e.g. when someone joins/leaves) and getStats only happens once per
second or less (typically).
NOTRY=Everything passes except unrelated purple bot
Bug: webrtc:14510
Change-Id: Iaf539a5cc8c87184fa0a87b9c889a13b941a9ad1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38262}
According to a pprof, creating RTCCodecStats is one of the places where
we spend the most CPU time in the event of creating hundreds of them:
https://screenshot.googleplex.com/B6QNDvvoX8dK5vk
The lifetime was recently updated so that we no longer have to risk
creating hundreds of them, here is the relevant section:
https://w3c.github.io/webrtc-stats/#codec-dict*
This allows code simplifications and the deletion of
ProduceCodecStats_n since we can now do a lazy instantiation of codec
stats at the point of being referenced.
Bug: webrtc:14444
Change-Id: I342c5bfebe6a4be0359da3ea106692c7a217779e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275763
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38209}
Unlike the cache of the entire stats report which is time limited, this
certificate cache is valid for an unlimited amount of time, but is
cleared at ClearCachedStatsReport() which is already called on each
SLD/SRD call. Since certificates can only change by negotiation, this
cache is ensured to always be invalidated when certificates change.
Since ClearCachedStatsReport() can happen for other reasons than
certificates changing we may clear the cache more often then is
necessary, but arguably this is seldom enough that we don't have to
create a separate "ClearCertificateStats()" method. Keep it simple?
The cache specifically avoids rtc::SSLCertChain::GetStats which
trigger rtc::SSLCertificate::GetStats and rtc::Base64::EncodeFromArray.
Bug: webrtc:14458
Change-Id: I5f95a4a5eb51cc4462147270fdae7bb9fb7bc822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276602
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38205}
Optional better describes "optionality" so let's do it for the sake of
style. But a side-effect of switching to optional may be better memory
locality than std::unique_ptr<>. (Anecdotally I saw a pprof suggesting a
significant amount of time being spent allocating/reading these maps.
This CL is unlikely to make the difference but it can't hurt.)
Bug: webrtc:14289
Change-Id: I7dcea9625b95c2f1a23e7d9595d27b58883570e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269404
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37624}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
This reverts commit 8efc914cf353cea138a453c45e970e589bec0834.
Reason for revert: Breaks downstream project.
Original change's description:
> Replace use of sigslot with CallbackList in data_channel_controller
>
> This is a straightforward replacement; simplifications due to the ability
> to inline functions in the lambdas are for a later CL.
>
> Bug: webrtc:11943
> Change-Id: I7274cedde507b954f1d8aa8bc560861102eeb264
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250540
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35936}
TBR=nisse@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I8fd0f32ceec866bfd9a08cac1108b559bf03caac
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251280
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35941}
This is a straightforward replacement; simplifications due to the ability
to inline functions in the lambdas are for a later CL.
Bug: webrtc:11943
Change-Id: I7274cedde507b954f1d8aa8bc560861102eeb264
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35936}
This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0.
Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}
TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
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}
pc_->GetCallStats() does a blocking-invoke if not already on the worker
thread. By moving this call into one of the lambdas that is already
executing on the worker thread, we can "piggy-back" on it and reduce
the number of blocking-invokes by one.
No change in behavior is intended with this CL, other than performance
improvements.
Bug: webrtc:11767
Change-Id: I04eaf990be946720353adca82e87b739ec6614f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193060
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32602}
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 ensures with DCHECK-crashes that we don't accidentally do more
blocking invokes than we think.
Remaining blocking invokes FYI:
- PrepareTransceiverStatsInfos_s_w() does 1 blocking invoke (regardless
of the number of transceivers or channels) to the worker thread. This
is because VoiceMediaChannel, VideoMediaChannel and GetParameters()
execute on the worker thread, and the result of these operations are
needed on the signalling thread.
- pc_->GetCallStats() does 1 blocking invoke to the worker thread.
These two blocking invokes can be merged, reducing the total number of
blocking invokes from 2 to 1, but this CL does not attempt to do that.
I filed https://crbug.com/webrtc/11767 for that.
Bug: webrtc:11716
Change-Id: Iebc2ab350d253fd037211cdd283825b4e5b2d446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178867
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31670}
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
This implements the essentials of RTCRemoteInboundRtpStreamStats. This
includes:
- ssrc
- transportId
- codecId
- packetsLost
- jitter
- localId
- roundTripTime
https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
The following members are not implemented because they require more
work...
- From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded,
packetsRepaired, burstPacketsLost, burstPacketsDiscarded,
burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate,
gapLossRate and gapDiscardRate.
- From RTCRemoteInboundRtpStreamStats: fractionLost.
Bug: webrtc:10455, webrtc:10456
Change-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28073}
This implements RTCAudioSourceStats and RTCVideoSourceStats, both
inheriting from abstract dictionary RTCMediaSourceStats:
https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
All members are implemented except for the total "frames" counter:
- trackIdentifier
- kind
- width
- height
- framesPerSecond
This means to make googFrameWidthInput, googFrameHeightInput and
googFrameRateInput obsolete.
Implemented using the same code path as the goog stats, there are
some minor bugs that should be fixed in the future, but not this CL:
1. We create media-source objects on a per-track attachment basis.
If the same track is attached multiple times this results in
multiple media-source objects, but the spec says it should be on a
per-source basis.
2. framesPerSecond is only calculated after connecting (when we have a
sender with SSRC), but if collected on a per-source basis the source
should be able to tell us the FPS whether or not we are sending it.
Bug: webrtc:10453
Change-Id: I23705a79f15075dca2536275934af1904a7f0d39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137804
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28028}
This is a reland of 05d43c6f7fe497fed0f2c8714e2042dd07a86df2
The original CL got reverted because Chrome did not support IsQuitting() which
triggered a NOTREACHED() inside of a DCHECK. With
https://chromium-review.googlesource.com/c/chromium/src/+/1491620
it is safe to reland this CL.
The only changes between this and the original patch set is that this is now
rebased on top of https://webrtc-review.googlesource.com/c/src/+/124701, i.e.
rtc::PostMessageWithFunctor() has been replaced by rtc::Thread::PostTask().
Original change's description:
> Fix getStats() freeze bug affecting Chromium but not WebRTC standalone.
>
> PeerConnection::Close() is, per-spec, a blocking operation.
> Unfortunately, PeerConnection is implemented to own resources used by
> the network thread, and Close() - on the signaling thread - destroys
> these resources. As such, tasks run in parallel like getStats() get into
> race conditions with Close() unless synchronized. The mechanism in-place
> is RTCStatsCollector::WaitForPendingRequest(), it waits until the
> network thread is done with the in-parallel stats request.
>
> Prior to this CL, this was implemented by performing
> rtc::Thread::ProcessMessages() in a loop until the network thread had
> posted a task on the signaling thread to say that it was done which
> would then get processed by ProcessMessages(). In WebRTC this works, and
> the test is RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection.
>
> But because Chromium's thread wrapper does no support
> ProcessMessages(), calling getStats() followed by close() in Chrome
> resulted in waiting forever (https://crbug.com/850907).
>
> In this CL, the process messages loop is removed. Instead, the shared
> resources are guarded by an rtc::Event. WaitForPendingRequest() still
> blocks the signaling thread, but only while shared resources are in use
> by the network thread. After this CL, calling WaitForPendingRequest() no
> longer has any unexpected side-effects since it no longer processes
> other messages that might have been posted on the thread.
>
> The resource ownership and threading model of WebRTC deserves to be
> revisited, but this fixes a common Chromium crash without redesigning
> PeerConnection, in a way that does not cause more blocking than what
> the other PeerConnection methods are already doing.
>
> Note: An alternative to using rtc::Event is to use resource locks and
> to not perform the stats collection on the network thread if the
> request was cancelled before the start of processing, but this has very
> little benefit in terms of performance: once the network thread starts
> collecting the stats, it would use the lock until collection is
> completed, blocking the signaling thread trying to acquire that lock
> anyway. This defeats the purpose and is a riskier change, since
> cancelling partial collection in this inherently racy edge-case would
> have observable differences from the returned stats, which may cause
> more regressions.
>
> Bug: chromium:850907
> Change-Id: Idceeee0bddc0c9d5518b58a2b263abb2bbf47cff
> Reviewed-on: https://webrtc-review.googlesource.com/c/121567
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26707}
TBR=steveanton@webrtc.org
Bug: chromium:850907
Change-Id: I5be7f69f0de65ff1120e4926fbf904def97ea9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/124781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26896}
This reverts commit 05d43c6f7fe497fed0f2c8714e2042dd07a86df2.
Reason for revert: It breaks some Chromium trybots:
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_asan_rel_ng/206387https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_tsan_rel_ng/207737https://ci.chromium.org/p/chromium/builders/luci.chromium.try/win10_chromium_x64_rel_ng/202283
Original change's description:
> Fix getStats() freeze bug affecting Chromium but not WebRTC standalone.
>
> PeerConnection::Close() is, per-spec, a blocking operation.
> Unfortunately, PeerConnection is implemented to own resources used by
> the network thread, and Close() - on the signaling thread - destroys
> these resources. As such, tasks run in parallel like getStats() get into
> race conditions with Close() unless synchronized. The mechanism in-place
> is RTCStatsCollector::WaitForPendingRequest(), it waits until the
> network thread is done with the in-parallel stats request.
>
> Prior to this CL, this was implemented by performing
> rtc::Thread::ProcessMessages() in a loop until the network thread had
> posted a task on the signaling thread to say that it was done which
> would then get processed by ProcessMessages(). In WebRTC this works, and
> the test is RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection.
>
> But because Chromium's thread wrapper does no support
> ProcessMessages(), calling getStats() followed by close() in Chrome
> resulted in waiting forever (https://crbug.com/850907).
>
> In this CL, the process messages loop is removed. Instead, the shared
> resources are guarded by an rtc::Event. WaitForPendingRequest() still
> blocks the signaling thread, but only while shared resources are in use
> by the network thread. After this CL, calling WaitForPendingRequest() no
> longer has any unexpected side-effects since it no longer processes
> other messages that might have been posted on the thread.
>
> The resource ownership and threading model of WebRTC deserves to be
> revisited, but this fixes a common Chromium crash without redesigning
> PeerConnection, in a way that does not cause more blocking than what
> the other PeerConnection methods are already doing.
>
> Note: An alternative to using rtc::Event is to use resource locks and
> to not perform the stats collection on the network thread if the
> request was cancelled before the start of processing, but this has very
> little benefit in terms of performance: once the network thread starts
> collecting the stats, it would use the lock until collection is
> completed, blocking the signaling thread trying to acquire that lock
> anyway. This defeats the purpose and is a riskier change, since
> cancelling partial collection in this inherently racy edge-case would
> have observable differences from the returned stats, which may cause
> more regressions.
>
> Bug: chromium:850907
> Change-Id: Idceeee0bddc0c9d5518b58a2b263abb2bbf47cff
> Reviewed-on: https://webrtc-review.googlesource.com/c/121567
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26707}
TBR=steveanton@webrtc.org,hbos@webrtc.org
Change-Id: Icd82cdd5bd086a90999f7fd5f8616e1f2d2153bf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:850907
Reviewed-on: https://webrtc-review.googlesource.com/c/123225
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26721}
PeerConnection::Close() is, per-spec, a blocking operation.
Unfortunately, PeerConnection is implemented to own resources used by
the network thread, and Close() - on the signaling thread - destroys
these resources. As such, tasks run in parallel like getStats() get into
race conditions with Close() unless synchronized. The mechanism in-place
is RTCStatsCollector::WaitForPendingRequest(), it waits until the
network thread is done with the in-parallel stats request.
Prior to this CL, this was implemented by performing
rtc::Thread::ProcessMessages() in a loop until the network thread had
posted a task on the signaling thread to say that it was done which
would then get processed by ProcessMessages(). In WebRTC this works, and
the test is RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection.
But because Chromium's thread wrapper does no support
ProcessMessages(), calling getStats() followed by close() in Chrome
resulted in waiting forever (https://crbug.com/850907).
In this CL, the process messages loop is removed. Instead, the shared
resources are guarded by an rtc::Event. WaitForPendingRequest() still
blocks the signaling thread, but only while shared resources are in use
by the network thread. After this CL, calling WaitForPendingRequest() no
longer has any unexpected side-effects since it no longer processes
other messages that might have been posted on the thread.
The resource ownership and threading model of WebRTC deserves to be
revisited, but this fixes a common Chromium crash without redesigning
PeerConnection, in a way that does not cause more blocking than what
the other PeerConnection methods are already doing.
Note: An alternative to using rtc::Event is to use resource locks and
to not perform the stats collection on the network thread if the
request was cancelled before the start of processing, but this has very
little benefit in terms of performance: once the network thread starts
collecting the stats, it would use the lock until collection is
completed, blocking the signaling thread trying to acquire that lock
anyway. This defeats the purpose and is a riskier change, since
cancelling partial collection in this inherently racy edge-case would
have observable differences from the returned stats, which may cause
more regressions.
Bug: chromium:850907
Change-Id: Idceeee0bddc0c9d5518b58a2b263abb2bbf47cff
Reviewed-on: https://webrtc-review.googlesource.com/c/121567
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26707}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
