std::not2 is deprecated in C++17, and that starts failing on C++17 mode
of ios_simulator build. This CL replaces it with a lambda to avoid the
warning.
Bug: chromium:752720
Change-Id: Id7ef847df0fbe0c44583ef3320e06f44644de929
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128620
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taiju Tsuiki <tzik@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27198}
This change fixes some inefficiencies and quirks in the code that
originates in RtpTransport leading up to the demux.
This work is in preparation for more refactoring of the Demux stage
onwards.
Bug: webrtc:10297
Change-Id: I7b8f00134657d62c722939618a55a91a2b6040bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128220
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27185}
Main change is deleting support for @userinfo in turn urls. This was
specified in early internet drafts, but never made it into RFC 7065.
Bug: webrtc:6663, webrtc:10422
Change-Id: Idd315a9e6001326f3104be62be3bd0991adc7db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128423
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27171}
(so far SetBitrate did not do anything for media transport)
Bug: webrtc:9719
Change-Id: I48e669341ffe6c9e4697ff9146c314be7796a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27169}
MediaChannel accepted the RtpPacket buffers through non-const pointer.
This is both unclear and introduces questions regarding if the buffer is
actually copied or not.
This change modifies the method to accept by value to reduce ambiguity.
Usage of the non-const data() method which could potentially copy the
buffer contents is also reduced in favor of cdata() which never copies.
Bug: None
Change-Id: I3b2daef0d31cb6aacceb46c86da3a40ce836242b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27090}
This is not used in practice as there's functionality on
other levels that serves the same purpose.
Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
Call DataChannelObserver::OnBufferedAmountChange on each successful send.
Previously, the observer would get notified of buffered amount changes only when
queued send data is consumed. Data gets queued only if it cannot be sent right
away. According to the WebRTC standard[1], bufferedamount should be increased
before each sent and decreased after each successful sent. Update implementation
to be standard compliant.
Design doc: http://doc/1lorHBn-GMn5U0T0RQANxrsW0pXhw8XGZM-xZyVUOW90
[1] https://w3c.github.io/webrtc-pc/#dom-datachannel-bufferedamount
Bug: chromium:878682
Change-Id: Ife009d30c4a18dced9a54cf600a445bb1f02561d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123237
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27057}
The previous URI was a placeholder and is not valid. The URI
https://webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02/
must be used instead.
Bug: webrtc:10264
Change-Id: Ibabde599b5bbd116c1c5e86ba0c9c64019bf7026
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126360
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27051}
Metrics are added to measure:
1. The number of send encodings in calls to AddTransceiver.
2. The number of times that simulcast is disabled because there is no
support from remote peer.
3. The number of times simulcast is indicated in ApplyLocal and
ApplyRemote and with which API surface (no simulcast, legacy munging,
spec-compliant).
Bug: webrtc:10372
Change-Id: I84717a1911efdf8aaf43cd6c04c7f09fcf2c58f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125482
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26979}
Currently, tests that verify metrics use a combination of
metrics::NumSamples and metrics::NumEvents to assert which samples
were recorded and how many times they were recorded. This means
that a comprehensive tests has n + 1 assertions for n distinct
samples.
The new metrics::Samples function returns a map of sample --> num
events which can be asserted against using gmock matchers,
achieving better coverage and better test failure messages in just
one line.
Bug: None
Change-Id: I07d4a766654cfc04e414b77b6de02927683a361f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125486
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26974}
The old iceConnectionState only becomes completed on the controlling side, and only if we're not configured to continually gather ICE candidates. This change makes the new ICE connection state do the same thing.
Bug: webrtc:10356
Change-Id: I82ca854a638a52674e4ca43364cf454dacb0cf1e
Reviewed-on: https://webrtc-review.googlesource.com/c/124360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26950}
Simulcast is disabled if the RIDs are not negotiated.
This change addresses the scenario in which RIDs are negotiated but
support for the RID extension is not negotiated.
In such cases, the RID extension cannot be used, so support for
simulcast should be turned off, as if RIDs were not negotiated.
A similar case can be made for MIDs, however MIDs are not explicitly
specified in simulcast. RIDs are only guaranteed to be unique within
a media section so it would seem that MIDs should be required.
However, applications supply RID values and can guarantee their
uniqueness, so unlike RIDs, the use of MIDs is not enforced as mandatory.
Bug: webrtc:10075
Change-Id: Ic1b27878ea152eaee43a38bbfda11144307766fe
Reviewed-on: https://webrtc-review.googlesource.com/c/125176
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26934}
We already support decoding of the x-mt line. This change adds the
a=x-mt line to the SDP offer. This is not a backward compatible change
for media transport (because of the changes in pre-shared key handling)
1) if media transport is enabled, and SDES is enabled, generate the
media transport offer.
2) if media transport generated the offer, add that offer to the x-mt
line.
3) in order to create media transport, require an x-mt line (backward incompatible).
The way it works is that
1) PeerConnection, on the offerer, asks jsep transport for the
configuration of the media transport.
2) Tentative media transport is created in JsepTransportController when
that happens.
3) SessionDescription will include configuration from this tentative
media transport.
4) When the LocalDescription is set on the offerer, the tentative media
transport is promoted to the real media transport.
Caveats:
- now we really only support MaxBundle. In the previous implementations,
two media transports were briefly created in some tests, and the second
one was destroyed shortly after instantiation.
- we, for now, enforce SDES. In the future, whether SDES is used will be
refactored out of the peer connection.
In the future (on the callee) we should ignore 'is_media_transport' setting. If
Offer contains x-mt, media transport should be used (if the factory is
present). However, we need to decide how to negotiate media transport
for data channels vs data transport for media (x-mt line at this point
doesn't differentiate the two, so we still need to use app setting).
This change also removes the negotation of pre-shared key from the
a=crypto line. Instead, media transport will have its own, 256bit key.
Such key should be transported in the x-mt line. This makes the code
much simpler, and simplifies the dependency / a=crypto lines parsing.
Also, adds a proper test for the connection re-offer (on both sides: callee and caller).
Before, it was possible that media transport could get recreated, based on the offer.
The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test.
This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even
when there is a re-offer.
Bug: webrtc:9719
Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01
Reviewed-on: https://webrtc-review.googlesource.com/c/125040
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26933}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
This CL applies clang-tidy's bugprone-argument-comment [1] to the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/bugprone-argument-comment.html
Bug: webrtc:10252
Change-Id: I77fec17509311275f18e730e482fb9f3fb2998ad
Reviewed-on: https://webrtc-review.googlesource.com/c/124989
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26900}
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 involves inserting an extra layer between jsep_transport_controller
and the cricket::SctpTransportInternal layer. The objects at this layer
are reference counted.
Bug: chromium:818643
Change-Id: Ibed57c4a538de981cee63e0f7f1f319f029cab39
Reviewed-on: https://webrtc-review.googlesource.com/c/123884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26889}
If x-mt line is present (one or more), and the first line is dedicated
for the media transport that we support, pass the config down to this
media transport.
In the future we will do 3 changes:
1) Add MediaTransportFactory::IsSupported(config) to let the
implementation decide whether the current factory can support a given
setting
2) Add support for multiple x-mt lines. Right now the support is
minimal: we only look at the first line (because we only allow single
media transport factory). In the future, when RtpMediaTransport is
introduced, this may and will change.
3) Allow multiple MediaTransportFactories and add fallback to RTP if
media transport is not supported.
Current solution provides backward compatibility for the 2 above
extensions.
Bug: webrtc:9719
Change-Id: I82a469fecda57effc95d7d8191f4a9e4a01d199c
Reviewed-on: https://webrtc-review.googlesource.com/c/124800
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26882}
This change adds a parser for the x-mt line to the WebRTC SDP, and adds a collection in the SessionDescription for media transport.
x-mt line contains information about media transport settings.
This is an experimental line.
This change will be followed up by two other changes:
* Setting the offer with x-mt
* Using the x-mt line to pass it to the media transport
Bug: webrtc:9719
Change-Id: I46568610d4092a69ada7b7c1d3987d698ddba0be
Reviewed-on: https://webrtc-review.googlesource.com/c/124601
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26881}
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
RtpSender uses a transactional model when getting and setting RtpParameters.
One must call GetParameters() and then can use the returned object in a
subsequent call to SetParameters().
PeerConnection was calling GetParameters() and SetParameters() during
negotiation in SetLocalDescription and SetRemoteDescription effectively
invalidating any parameters that the client previously held.
This change introduces an internal way for the platform to modify
parameters without invalidating the transactional model, provided that
the modification is not severe.
Ex. removing simulcast layers is a severe modification and will
invalidate any outstanding parameters.
Bug: webrtc:10339
Change-Id: I362e8ca4d9556e04a1aa7a3e74e2c275f8d16fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/124504
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26864}
Bug: webrtc:9719
Change-Id: I4aef407c4770fc98abcbc114b87e73bbf13d8f56
Reviewed-on: https://webrtc-review.googlesource.com/c/124021
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26860}
This informs the media transport that PeerConnection wants to use a data channel
and gives it a chance to set up before the data channel sends the first message.
Bug: webrtc:9719
Change-Id: I6ea905a74b29b8735e77ac68bc8606e7bca77f18
Reviewed-on: https://webrtc-review.googlesource.com/c/124020
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26823}
TransportSequenceNumberV2 is an experimental feature that should
not be part of the default offer. However, if we receive an offer
with this extension we should respond that we support it.
Bug: webrtc:10264
Change-Id: Id2424d421361e5d71f3a608cb8f74b63645c264a
Reviewed-on: https://webrtc-review.googlesource.com/c/123783
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26817}
As a unintended consequence of the changed iceConnectionState implementation we won't ever transition to the "failed" iceConnection state unless a connection has first been established. This CL fixes that and adds a integration test for this scenario.
Bug: chromium:933786
Change-Id: I45effd7411959ac0e5b16a13d7568756dbeff4d1
Reviewed-on: https://webrtc-review.googlesource.com/c/123785
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26811}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: I4be508284af573d93657c933a64e9f970b7e3adf
Reviewed-on: https://webrtc-review.googlesource.com/c/123190
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26805}
Since the ice_transport element of DtlsTransport is never changed
over the lifetime of the object, it's safe to return it on any thread.
Added const qualifier to make this obvious.
Bug: chromium:907849
Change-Id: Ib180682689a57df5106717b125c626d7ac9e7561
Reviewed-on: https://webrtc-review.googlesource.com/c/123781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26800}
It was previously possible to escape the sandbox by calling
rtc::Thread::SetAllowBlockingCalls(true).
This CL only removes the loophole on non-Android builds, because we
still have old Android code that relies on it. We expect that code to
go away soon-ish, though.
Bug: webrtc:9987
Change-Id: Ida96400d0abe430af4c2046284795d37d64f6613
Reviewed-on: https://webrtc-review.googlesource.com/c/123523
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26792}
It is possible to get event logs from both RTP and MediaTransport when
media transport is only used for data channel. This is undesirable. We
would rather not get any logs from media transport when it's not used
for media.
This change disables rtc event log when media transport is not used for
media.
Bug: webrtc:9719
Change-Id: Ibc660b37c5d98001144e5f68b32f0608fd6ede33
Reviewed-on: https://webrtc-review.googlesource.com/c/123260
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26776}
Most of the implementation in rtp_sender.cc is a copy paste for both
Audio & Video RTP senders. This change moves all the common behavior
into the base RtpSenderInternal class.
Template method pattern is used to accomodate for the very slight differences
between audio and video senders.
Bug: None
Change-Id: I6d4e93cd32fbb0fb361fd0e1883791019bde9a92
Reviewed-on: https://webrtc-review.googlesource.com/c/123411
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26758}
Added a layer in RtpSender that bridges the gap between the layers
that the user sees and the layer that the media engine sees.
Media engine still maintains the invariant that the number of layers
cannot be changed, while RtpSender adds and removes layers between
the user GetParameters and SetParameters calls and the media engine.
Bug: webrtc:10251
Change-Id: I33839c1f9a9052cb6130253e5a582606f2cbe54a
Reviewed-on: https://webrtc-review.googlesource.com/c/122641
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26756}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: If7bbd5a468a8c50ac3cfe03cd2ed4f5b5f461195
Reviewed-on: https://webrtc-review.googlesource.com/c/123047
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26727}
Latency corresponds to base minimum delay on NetEq.
Bug: webrtc:10287
Change-Id: I538d202e3e4fe07b779c46bf560e2fde38e0468e
Reviewed-on: https://webrtc-review.googlesource.com/c/121704
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26724}
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}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: I68a51bfa3342c6d66d67276d5979144af34692c9
Reviewed-on: https://webrtc-review.googlesource.com/c/123046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26716}
Plus all the annotations that are necessary to make things compile
again.
(The latter could only be made const, since it is accessed from both
the signal thread and the worker thread.)
Bug: webrtc:9987
Change-Id: I49f1568d26a327f48cd94c2c70bc5939e9fbd59a
Reviewed-on: https://webrtc-review.googlesource.com/c/122842
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26715}
