Reland of commit a743303211b89bbcf4cea438ee797bbbc7b59e80
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
- Mark FindHeaderExtensionByUri without filter argument as deprecated
Bug: webrtc:11713
Change-Id: I52a5ade1b94bc01d1c2a35cb56023684fcaf9982
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34129}
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}
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}
The channel classes have stored the negotiated headers but a more
natural place to store them is in the RtpTransceiver class where
RtpHeaderExtension state is managed as well as the implementation of
the only method that depends on the stored state,
HeaderExtensionsNegotiated().
Also adding a TODO for further improvements where we're unnecessarily
storing state in the channel classes for the purposes of the transports.
Bug: webrtc:12726
Change-Id: If36668e3e49782ddeada23ebed126ee2c4935b8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216691
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33917}
This reverts commit 48a4d33719390b7bcaf8445a1581a00825f67bfb.
Reason for reland:
Relanding the original change but without the modification for
VideoSendStream::GetStats. Essentially there's a TODO there to fix
the downstream issue, which seems to be benign.
Original change's description:
> Revert "Remove Invoke from VideoChannel::FillBitrateInfo."
>
> This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.
>
> Reason for revert: Speculative revert (breaks downstream project).
>
> Original change's description:
> > Remove Invoke from VideoChannel::FillBitrateInfo.
> >
> > The method is relied upon by StatsCollector where it was called from the
> > signaling thread in a loop. Now there's at most one invoke (not N).
> >
> > Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> > VideoSendStream. Updating all related tests that fetched stats from
> > the wrong context.
> >
> > Bug: webrtc:12726
> > Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33894}
>
> TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12726
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33898}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12726
Change-Id: I41cce3b11a29905cde982c22e82b9b1f5a98e654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217222
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33902}
The Invoke() isn't necessary as the method is always called on the
correct thread (now enforced with a DCHECK).
Bug: webrtc:12726
Change-Id: I53bbdabc5d6de1316e7cf478d8912e19dd0e45e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216690
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33899}
This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.
Reason for revert: Speculative revert (breaks downstream project).
Original change's description:
> Remove Invoke from VideoChannel::FillBitrateInfo.
>
> The method is relied upon by StatsCollector where it was called from the
> signaling thread in a loop. Now there's at most one invoke (not N).
>
> Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> VideoSendStream. Updating all related tests that fetched stats from
> the wrong context.
>
> Bug: webrtc:12726
> Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33894}
TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33898}
The method is relied upon by StatsCollector where it was called from the
signaling thread in a loop. Now there's at most one invoke (not N).
Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
VideoSendStream. Updating all related tests that fetched stats from
the wrong context.
Bug: webrtc:12726
Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33894}
This changes the notification to a single std::function pointer
instead of being a sigslot::signal1<> collection.
Summary:
* Remove SignalFirstPacketReceived_, the last sigslot member variable.
(still inherits from sigslot::has_slots<>)
* BaseChannel doesn't post to the signaling thread anymore. The only
reason that remains for the signaling_thread_ variable, is for
thread checking.
* Remove BaseChannel's reliance on MessageHandlerAutoCleanup
(still inherits from MessageHandler)
RtpTransceiver is the consumer of this event. That class is also the
class that sits between the PC classes and the channel object, holding
a pointer to the channel and managing calls that come in on the
signaling thread, such as SetChannel. The responsibility of delivering
the first packet received on the signaling thread is now with
RtpTransceiver:
* RtpTransceiver always requires a ChannelManager instance. Previously
this variable was sometimes set, but it's now required.
* Updated tests in rtp_transceiver_unittest.cc to include a
ChannelManager as well as fix them to include call expectations for
mock sender and receivers.
Bug: webrtc:11993, webrtc:11988
Change-Id: If49d6be157cd7599fa6fe3a42cd0a363464e3a74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215979
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33853}
There's a bit of copy/pasted code in the channel code, which is
making moving network traffic consistently over to the network thread
a bit trickier than it needs to be, so I'm also updating variable
names used in Set[Local|Remote]Content_w to be more explicitly the same
and make it clear that the code is copy/pasted (and future updates can
consolidate more of it).
Also removing some code from the video/voice media channels that's
effectively dead code (vector + registration methods that aren't needed)
Bug: webrtc:12705
Change-Id: I2e14e69fbc489a64fc1e8899aaf1cfc979fe840b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215978
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33847}
* Changing return value from bool to void since the operation as async
effects anyway.
* Removing the `enabled()` accessor due to potential threading issues
and potential TOCTOU issues. It was only used in one place anyway.
* Applying thread restrictions to member variables.
Bug: none
Change-Id: I51949f5594339952d7b717cfd82f99b532e86b23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216182
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33830}
* 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}
This reverts commit a743303211b89bbcf4cea438ee797bbbc7b59e80.
Reason for revert: Breaks downstream tests that attempt to call FindHeaderExtensionByUri with 2 arguments. Could you keep the old 2-argument method declaration and just forward the call to the new 3-argument method with a suitable no-op filter?
Original change's description:
> Fix RTP header extension encryption
>
> Previously, RTP header extensions with encryption had been filtered
> if the encryption had been activated (not the other way around) which
> was likely an unintended logic inversion.
>
> In addition, it ensures that encrypted RTP header extensions are only
> negotiated if RTP header extension encryption is turned on. Formerly,
> which extensions had been negotiated depended on the order in which
> they were inserted, regardless of whether or not header encryption was
> actually enabled, leading to no extensions being sent on the wire.
>
> Further changes:
>
> - If RTP header encryption enabled, prefer encrypted extensions over
> non-encrypted extensions
> - Add most extensions to list of extensions supported for encryption
> - Discard encrypted extensions in a session description in case encryption
> is not supported for that extension
>
> Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
> into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
> header extensions will prevent any RTP packets being sent/received.
>
> Bug: webrtc:11713
> Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33723}
TBR=deadbeef@webrtc.org,terelius@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com
Change-Id: I7df6b0fa611c6496dccdfb09a65ff33ae4a52b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33727}
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.
Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
Also updating SocketOptionsMergedOnSetTransport test code to make the
call to SetRtpTransport from the right context.
Bug: webrtc:12636
Change-Id: I343851bcf8ac663d7559128d12447a9a742786f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33633}
This is a reland of d48a2b14e7545d0a0778df753e062075c044e2a1
The diff of the reland (what caused the tsan error) can be seen
by diffing patch sets 2 and 3. Essentially I missed keeping the calls
to the transport controller on the worker thread. Note to self to add
thread/sequence checks to that code so that we won't have to rely on
tsan :)
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
Bug: webrtc:11993, webrtc:12430
Change-Id: I4fccaa418d22c2087a55bbb3ddbb25fac3b4dfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33153}
This reverts commit d48a2b14e7545d0a0778df753e062075c044e2a1.
Reason for revert: TSan tests started to fail constantly after this CL (it looks like it is flaky and the CQ was lucky to get green). See https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25042/overview.
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: Id87cf9cbcc8ed58e74d755a110f0ef9dd980e298
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205525
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33145}
This moves the thread hop for network events, from BaseChannel and
into Call. The reason for this is to move the control over those hops
(including DeliverPacket[Async]) into the same class where the state
is held that is affected by those hops. Once that's done, we can start
moving the relevant network state over to the network thread and
eventually remove the hops.
I'm also adding several TODOs for tracking future steps and give
developers a heads up.
Bug: webrtc:11993
Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33138}
By changing was_ever_writable_ to be guarded by the worker thread
instead of the network thread.
Gets rid of one network thread invoke per audio/video m= section per
round of negotiation.
NOTRY=True
Bug: webrtc:12266
Change-Id: Ie913a9f96db3fd8351559e916922c82d2d0337f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203881
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33130}
This reverts commit c1ad1ff178f0d0dfcde42843c51ae703005aaca1.
Reason for revert: This blocks the worker thread for a longer
contiguous period of time which can lead to delays in processing
packets. And due to other recent changes, the need to speed up
SetLocalDescription/SetRemoteDescription is reduced.
Still plan to reland some of the changes from the CL, just not the
part that groups the Invokes.
Original change's description:
> Do all BaseChannel operations within a single Thread::Invoke.
>
> Instead of doing a separate Invoke for each channel, this CL first
> gathers a list of operations to be performed on the signaling thread,
> then does a single Invoke on the worker thread (and nested Invoke
> on the network thread) to update all channels at once.
>
> This includes the methods:
> * Enable
> * SetLocalContent/SetRemoteContent
> * RegisterRtpDemuxerSink
> * UpdateRtpHeaderExtensionMap
>
> Also, removed the need for a network thread Invoke in
> IsReadyToSendMedia_w by moving ownership of was_ever_writable_ to the
> worker thread.
>
> Bug: webrtc:12266
> Change-Id: I31e61fe0758aeb053b09db84f234deb58dfb3d05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194181
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32817}
TBR=deadbeef@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12266
Change-Id: I40ec519a614dc740133219f775b5638a488529b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203860
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33111}
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}
all invokes, as well as BaseChannel constructor and destructor
should run on the same task queue which allow to use
simpler cancellation of pending task on BaseChannel destruction
Bug: webrtc:12339
Change-Id: I311b6de940cc24cf6bb5b49e1bbd132fea2439e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202032
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33009}
SetNegotiatedHeaderExtensions_w queued a task to update the list
of negotiated header extensions on the signal thread from the
worker thread, in belief that a later call to
GetNegotiatedHeaderExtensions() would happen on the WebRTC proxies,
leading to the update happening before the readout. In downstream
project, this is not always the case.
Fix this by synchronously updating the list of negotiated header
extensions.
Bug: chromium:1051821
Change-Id: I3266292e7508bb7a22a3f7d871e82c12f60cfc83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201728
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32977}
Instead of doing a separate Invoke for each channel, this CL first
gathers a list of operations to be performed on the signaling thread,
then does a single Invoke on the worker thread (and nested Invoke
on the network thread) to update all channels at once.
This includes the methods:
* Enable
* SetLocalContent/SetRemoteContent
* RegisterRtpDemuxerSink
* UpdateRtpHeaderExtensionMap
Also, removed the need for a network thread Invoke in
IsReadyToSendMedia_w by moving ownership of was_ever_writable_ to the
worker thread.
Bug: webrtc:12266
Change-Id: I31e61fe0758aeb053b09db84f234deb58dfb3d05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194181
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32817}
This makes it thread-safe to access, but not necessarily to use.
Bug: webrtc:12230
Change-Id: I6b48d86dff24b162d382135abeaf560971fdf614
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196524
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32785}
This CL also adds commentary to member variables that couldn't be guarded
because they're accessed from multiple threads.
Bug: webrtc:12230
Change-Id: I5193a7ef36ab25588c76ee6a1863de6a844be1dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195331
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32705}
This fixes regressions caused by:
https://webrtc-review.googlesource.com/c/src/+/183120
... which disabled payload type demuxing when multiple video tracks are
present, to avoid one channel creating a default track intended for
another channel.
However, this isn't an issue when not bundling, as each track will be
delivered on separate transport.
And it's also not an issue when each track uses a distinct set of
payload types (e.g., VP8 is mapped to PT 96 in one m= section, and PT 97
in another).
This CL addresses both of those cases; PT demuxing is only disabled
when two bundled m= sections have overlapping payload types.
Bug: chromium:1139052, webrtc:12029
Change-Id: Ied844bffac2a5fac29147c11b56a5f83a95ecb36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187560
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32419}
SignalDtlsSrtpSetupFailure is never fired, so the setup code for it,
is dead code. Also removing declarations for methods that have no
implementation.
For other public signals in BaseChannel I've added an accessor which
has revealed a threading problem due to the member variable being public.
Bug: webrtc:11994
Change-Id: Iec6046c6a598066b92c956002ba4160708ae7dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32211}
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 log line is causing test failures due to excessive logging (see
referenced bug); reducing log level.
Bug: chromium:984879
Change-Id: Ic94ba0a39b91b4253a58ad54de0cba1ca49882e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175913
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31803}
Logging of content name (mid) is valuable to debug issues
in scenarios with multiple m= line sections in SDP.
For example, video conferencing applications which
uses SFU and Unified Plan SDPs will likely to leverage
from more detailed logs when issues need to be debugged.
Bug: webrtc:10139
Change-Id: Id52ba3ad54af5caa0f8c03daaa51bdb0caf9fe67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175115
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31302}
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
