This is to address flakiness of "DoubleThread" tests for the media
channel class. More investigation is in order though, so I'm adding
a TODO. The bug appears to be in test code only though, so this is
just to deflake the bots.
Bug: webrtc:12783
Change-Id: Ib6cf78927f2a9be9d2c6aa7f6915b1131a206e7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34049}
Xcode 12.5 triggers some warnings for -Wdeprecated-copy, and I believe
it is better to fix this problem than to suppress this warning.
Bug: webrtc:12749
Change-Id: I5ca5fd8fdcae18fe7d3941f78b3366b5f03b8c00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33990}
cricket::SendDataParams is replaced by webrtc::SendDataParams.
cricket::DataMessageType is replaced by webrtc::DataMessageType.
The sid member from cricket::SendDataParams is now passed as an argument
to functions that used one when necessary.
Bug: webrtc:7484
Change-Id: Ia4a89c9651fb54ab9a084a6098d49130b6319e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217761
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33966}
Pending messages on network thread for MediaChannel, will be dropped
when the MediaChannel object is deleted (without blocking).
Remove MessageHandler inheritance from Channel since Post-ing to the
network thread has been removed from there.
Copy/pasted code for SendRtp/SendRtcp in WebRtcVideoChannel and
WebRtcVoiceMediaChannel consolidated in MediaChannel.
Bug: webrtc:11993
Change-Id: I05320eb7f86b98adba50ca5eb8b76b78f4111263
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217720
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33955}
It doesn't make sense to use negative values or 0 to disable the
feature, so we use an optional int value.
Values bigger than 65535 are clamped down.
Bug: webrtc:12730
Change-Id: I6bd9cd92f7d0a70a78cf5a7c91dca52c28d08ba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33954}
This CL mostly adds plumbing to get awareness of the network thread
to the media channel classes. Currently this pointer is only used
to DCHECK that `SetInterface` for the `NetworkInterface` pointer, is
called on the network thread. Follow up changes will establish that
most of the methods are called on the network thread and the mutex
in the MediaChannel base class, can be removed.
Most of the changes in the CL are in channel_unittest.cc. They're mostly
around updating the tests to incorporate the network thread in ways
that reflect how the classes are used in production. Another change is
to use accessor methods for the media channel instances instead of
caching potentially dangling pointers.
Bug: webrtc:11993
Change-Id: I8e2ed1bc23724e238554dbce386789d69660f7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217682
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33951}
SdpVideoFormat is used to configure video encoder and decoders.
This CL adds support for comparing two SdpVideoFormat objects
to determine if they specify the same video codec.
This functionality previously only existed in media/base/codec.h
which made the code sensitive to circular dependencies. Once
downstream projects stop using cricket::IsSameCodec, this code
can be removed.
Bug: chromium:1187565
Change-Id: I242069aa6af07917637384c80ee4820887defc7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213427
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33794}
This field was only used in RTP Data Channels and isn't needed anymore.
Bug: webrtc:6625
Change-Id: Ieaa7ae03ca3e90eb4ddec4d384f5a76cef1600cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215687
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33791}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
Since there is only a single type of DataChannel now, the enum was only used
when data channels were disabled at the PC API. That option has been
deprecated 4 years ago, it's now time to remove it.
Bug: webrtc:6625
Change-Id: I9e4ada1756da186e9639dd0fbf0249c55ea0b6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33778}
* Adds a OnPacketSent callback to MediaChannel, which matches with
MediaChannel::NetworkInterface::SendPacket.
* Moves the OnPacketSent handling to the media channel implementations
(video/voice) and removes the PeerConnection/SdpOfferAnswerHandler
layer from the call path.
* Call::OnSentPacket is called directly from the channels on the network
thread. This eliminates a PostTask to the worker thread for every
audio/video network packet.
* Remove sigslot dependency from MediaChannel (and derived).
Bug: webrtc:11993
Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33777}
BaseChannel adds and removes receive streams on the worker thread
(UpdateRemoteStreams_w) and then posts a task to the network thread to
update the demuxer criteria. Until this happens, OnRtpPacket() keeps
forwarding "recently removed" ssrc packets to the WebRtcVideoChannel.
Furthermore WebRtcVideoChannel::OnPacketReceived() posts task from the
network thread to the worker thread, so even if the demuxer criteria was
instantly updated we would still have an issue of in-flight packets for
old ssrcs arriving late on the worker thread inside WebRtcVideoChannel.
The wrong ssrc could also arrive when the demuxer goes from forwarding
all packets to a single m= section to forwarding to different m=
sections. In this case we get packets with an ssrc for a recently
created m= section and the ssrc was never intended for our channel.
This is a problem because when WebRtcVideoChannel sees an unknown ssrc
it treats it as an unsignalled stream, creating and destroying default
streams which can be very expensive and introduce large delays when lots
of packets are queued up.
This CL addresses the issue with callbacks for when a demuxer criteria
update is pending and when it has completed. During this window of time,
WebRtcVideoChannel will drop packets for unknown ssrcs.
This approach fixes the race without introducing any new locks and
packets belonging to ssrcs that were not removed continue to be
forwarded even if a demuxer criteria update is pending. This should make
a=inactive for 50p receive streams a glitch-free experience.
Bug: webrtc:12258, chromium:1069603
Change-Id: I30d85f53d84e7eddf7d21380fb608631863aad21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214964
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33757}
* A ChannelManager instance is now created via ChannelManager::Create()
* Initialization is performed inside Create(), RAII.
* All member variables in CM are now either const or RTC_GUARDED_BY
the worker thread.
* Removed dead code (initialization and capturing states are gone).
* ChannelManager now requires construction/destruction on worker thread.
- one fewer threads that its aware of.
* media_engine_ pointer removed from ConnectionContext.
* Thread policy changes moved from ChannelManager to ConnectionContext.
These changes will make a few other issues easier to fix, so tagging
those bugs with this CL.
Bug: webrtc:12601, webrtc:11988, webrtc:11992, webrtc:11994
Change-Id: I3284cf0a08c773e628af4124e8f52e9faddbe57a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212617
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33614}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
VideoBroadcaster is marked as thread-safe, but that is currently not the
case as OnDiscardedFrame() iterates through an std::vector of sinks in
VideoSourceBase that is not thread-safe and elements of that std::vector
are added/removed in AddOrUpdateSink()/RemoveSink() that could be called
on a different thread.
Bug: None
Change-Id: I5b61127f7ea6ce7f1322c5e770ab56643d7bd0d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208404
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33313}
Functionality wise, there should be no change with this CL, aside
from updating tests to anticipate OnPacketReceived to handle the packet
asynchronously (as already was the case via BaseChannel).
This only removes the network->worker hop out of the BaseChannel
class into the WebRTC MediaChannel implementations. However, it updates
the interface contract between BaseChannel and MediaChannel to align
with how we want things to work down the line, i.e. avoid hopping to
the worker thread for every rtp packet.
The following steps will be to update the video and voice channel
classes to call Call::DeliverPacket on the network thread and only
handle unsignalled SSRCs on the worker (exception case).
Bug: webrtc:11993
Change-Id: If0540874444565dc93773aee89d862f3bfc9c502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33040}
removes the unused
x-google-port
x-google-max-message-size
which were probably sctp-related and variants of max-message-size
and sctp-port parameters. Similarly for the
protocol
streams
parameters which seem to have been fmtp parameters.
BUG=chromium:943975
Change-Id: I21b543f717d6e12fd737f91c7e159362488cc2e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198122
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32866}
Two audio channels going into the AudioSource::Sink can either be
down-mixed to mono or encoded as stereo. This change enables WebRTC
users (such as Chromium) to query the number of audio channels actually
encoded. That information can in turn be used to tailor the audio
processing to the number of channels actually encoded.
This change fixes webrtc:8133 from a WebRTC perspective and will be
followed up with the necessary Chromium changes.
Bug: webrtc:8133
Change-Id: I8e8a08292002919784c05a5aacb21707918809c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197426
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32836}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
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}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.
Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
This will enable wiring up these signals to the platform specific capturers
Bug: chromium:1116430
Change-Id: I6cdab61eab202a24fa56167da57c389a5b1880c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182683
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32017}
This reverts commit 20b701f3d79c499b0981f03fbf3a9b0fe531ac5d.
Reason for reland: Reverting did not affect the test regression.
Original change's description:
> Revert "sdp: parse and serialize b=TIAS"
>
> This reverts commit c6801d4522ab94f965e258e68259fde312023654.
>
> Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.
>
> One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.
>
> Original change's description:
> > sdp: parse and serialize b=TIAS
> >
> > BUG=webrtc:5788
> >
> > Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31729}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:5788
> Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31762}
TBR=nisse@webrtc.org
Bug: webrtc:5788
Change-Id: I5c0ef29d275bb2264d9b706b085f7933d59e2801
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179760
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31816}
